7485 lines
292 KiB
C++
7485 lines
292 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/peer_connection.h"
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#include <algorithm>
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#include <limits>
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#include <memory>
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#include <queue>
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#include <set>
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#include <utility>
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#include <vector>
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#include "absl/algorithm/container.h"
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#include "absl/strings/match.h"
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#include "api/jsep_ice_candidate.h"
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#include "api/jsep_session_description.h"
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#include "api/media_stream_proxy.h"
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#include "api/media_stream_track_proxy.h"
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#include "api/rtc_error.h"
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#include "api/rtc_event_log/rtc_event_log.h"
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#include "api/rtc_event_log_output_file.h"
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#include "api/rtp_parameters.h"
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#include "api/uma_metrics.h"
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#include "api/video/builtin_video_bitrate_allocator_factory.h"
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#include "call/call.h"
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#include "logging/rtc_event_log/ice_logger.h"
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#include "media/base/rid_description.h"
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#include "media/sctp/sctp_transport.h"
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#include "pc/audio_rtp_receiver.h"
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#include "pc/audio_track.h"
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#include "pc/channel.h"
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#include "pc/channel_manager.h"
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#include "pc/dtmf_sender.h"
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#include "pc/media_stream.h"
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#include "pc/media_stream_observer.h"
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#include "pc/remote_audio_source.h"
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#include "pc/rtp_media_utils.h"
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#include "pc/rtp_receiver.h"
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#include "pc/rtp_sender.h"
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#include "pc/sctp_transport.h"
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#include "pc/sctp_utils.h"
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#include "pc/sdp_utils.h"
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#include "pc/stream_collection.h"
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#include "pc/video_rtp_receiver.h"
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#include "pc/video_track.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/string_encode.h"
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#include "rtc_base/strings/string_builder.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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#include "system_wrappers/include/metrics.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::MediaProtocolType;
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using cricket::RidDescription;
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using cricket::RidDirection;
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using cricket::SessionDescription;
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using cricket::SimulcastDescription;
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using cricket::SimulcastLayer;
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using cricket::SimulcastLayerList;
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using cricket::StreamParams;
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using cricket::TransportInfo;
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using cricket::LOCAL_PORT_TYPE;
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using cricket::PRFLX_PORT_TYPE;
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using cricket::RELAY_PORT_TYPE;
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using cricket::STUN_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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namespace {
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// UMA metric names.
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const char kSimulcastVersionApplyLocalDescription[] =
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"WebRTC.PeerConnection.Simulcast.ApplyLocalDescription";
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const char kSimulcastVersionApplyRemoteDescription[] =
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"WebRTC.PeerConnection.Simulcast.ApplyRemoteDescription";
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const char kSimulcastNumberOfEncodings[] =
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"WebRTC.PeerConnection.Simulcast.NumberOfSendEncodings";
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const char kSimulcastDisabled[] = "WebRTC.PeerConnection.Simulcast.Disabled";
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static const char kDefaultStreamId[] = "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_REPORT_USAGE_PATTERN,
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};
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static const int REPORT_USAGE_PATTERN_DELAY_MS = 60000;
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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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rtc::scoped_refptr<webrtc::SetSessionDescriptionObserver> observer;
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RTCError 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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RTCError 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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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->id()) != nullptr) {
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RTC_LOG(LS_ERROR) << "MediaStream with ID " << new_stream->id()
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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 AddPlanBRtpSenderOptions(
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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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int num_sim_layers) {
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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(), {},
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SimulcastLayerList(), num_sim_layers);
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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<RtpDataChannel>>&
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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 RtpDataChannel* channel = kv.second;
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if (channel->state() == RtpDataChannel::kConnecting ||
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channel->state() == RtpDataChannel::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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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_hostname =
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!local.address().hostname().empty() && local.address().IsUnresolvedIP();
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bool remote_hostname = !remote.address().hostname().empty() &&
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remote.address().IsUnresolvedIP();
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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_hostname) {
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if (remote_hostname) {
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return kIceCandidatePairHostNameHostName;
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} else if (remote_private) {
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return kIceCandidatePairHostNameHostPrivate;
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} else {
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return kIceCandidatePairHostNameHostPublic;
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}
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} else if (local_private) {
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if (remote_hostname) {
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return kIceCandidatePairHostPrivateHostName;
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} else 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_hostname) {
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return kIceCandidatePairHostPublicHostName;
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} else 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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// Logic to decide if an m= section can be recycled. This means that the new
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// m= section is not rejected, but the old local or remote m= section is
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// rejected. |old_content_one| and |old_content_two| refer to the m= section
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// of the old remote and old local descriptions in no particular order.
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// We need to check both the old local and remote because either
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// could be the most current from the latest negotation.
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bool IsMediaSectionBeingRecycled(SdpType type,
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const ContentInfo& content,
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const ContentInfo* old_content_one,
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const ContentInfo* old_content_two) {
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return type == SdpType::kOffer && !content.rejected &&
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((old_content_one && old_content_one->rejected) ||
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(old_content_two && old_content_two->rejected));
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}
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// Verify that the order of media sections in |new_desc| matches
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// |current_desc|. The number of m= sections in |new_desc| should be no
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// less than |current_desc|. In the case of checking an answer's
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// |new_desc|, the |current_desc| is the last offer that was set as the
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// local or remote. In the case of checking an offer's |new_desc| we
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// check against the local and remote descriptions stored from the last
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// negotiation, because either of these could be the most up to date for
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// possible rejected m sections. These are the |current_desc| and
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// |secondary_current_desc|.
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bool MediaSectionsInSameOrder(const SessionDescription& current_desc,
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const SessionDescription* secondary_current_desc,
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const SessionDescription& new_desc,
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const SdpType type) {
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if (current_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 < current_desc.contents().size(); ++i) {
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const cricket::ContentInfo* secondary_content_info = nullptr;
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if (secondary_current_desc &&
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i < secondary_current_desc->contents().size()) {
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secondary_content_info = &secondary_current_desc->contents()[i];
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}
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if (IsMediaSectionBeingRecycled(type, new_desc.contents()[i],
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¤t_desc.contents()[i],
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secondary_content_info)) {
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// For new offer descriptions, if the media section can be recycled, it's
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// valid for the MID and media type to change.
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continue;
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}
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if (new_desc.contents()[i].name != current_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* current_desc_mdesc =
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current_desc.contents()[i].media_description();
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if (new_desc_mdesc->type() != current_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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return desc1.contents().size() == desc2.contents().size();
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}
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void NoteKeyProtocolAndMedia(KeyExchangeProtocolType protocol_type,
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cricket::MediaType media_type) {
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// Array of structs needed to map {KeyExchangeProtocolType,
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// cricket::MediaType} to KeyExchangeProtocolMedia without using std::map in
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// order to avoid -Wglobal-constructors and -Wexit-time-destructors.
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static constexpr struct {
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KeyExchangeProtocolType protocol_type;
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cricket::MediaType media_type;
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KeyExchangeProtocolMedia protocol_media;
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} kEnumCounterKeyProtocolMediaMap[] = {
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{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_AUDIO,
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kEnumCounterKeyProtocolMediaTypeDtlsAudio},
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{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_VIDEO,
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kEnumCounterKeyProtocolMediaTypeDtlsVideo},
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{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_DATA,
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kEnumCounterKeyProtocolMediaTypeDtlsData},
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{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_AUDIO,
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kEnumCounterKeyProtocolMediaTypeSdesAudio},
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{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_VIDEO,
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kEnumCounterKeyProtocolMediaTypeSdesVideo},
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{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_DATA,
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kEnumCounterKeyProtocolMediaTypeSdesData},
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};
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RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocol", protocol_type,
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kEnumCounterKeyProtocolMax);
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for (const auto& i : kEnumCounterKeyProtocolMediaMap) {
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if (i.protocol_type == protocol_type && i.media_type == media_type) {
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RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocolByMedia",
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i.protocol_media,
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kEnumCounterKeyProtocolMediaTypeMax);
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}
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}
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}
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void NoteAddIceCandidateResult(int result) {
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RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.AddIceCandidate", result,
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kAddIceCandidateMax);
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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 =
|
|
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
|
|
for (const cricket::ContentInfo& content_info : desc->contents()) {
|
|
if (content_info.rejected) {
|
|
continue;
|
|
}
|
|
// Note what media is used with each crypto protocol, for all sections.
|
|
NoteKeyProtocolAndMedia(dtls_enabled ? webrtc::kEnumCounterKeyProtocolDtls
|
|
: webrtc::kEnumCounterKeyProtocolSdes,
|
|
content_info.media_description()->type());
|
|
const std::string& mid = content_info.name;
|
|
if (bundle && bundle->HasContentName(mid) &&
|
|
mid != *(bundle->FirstContentName())) {
|
|
// This isn't the first media section in the BUNDLE group, so it's not
|
|
// required to have crypto attributes, since only the crypto attributes
|
|
// from the first section actually get used.
|
|
continue;
|
|
}
|
|
|
|
// If the content isn't rejected or bundled into another m= section, crypto
|
|
// must be present.
|
|
const MediaContentDescription* media = content_info.media_description();
|
|
const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
|
|
if (!media || !tinfo) {
|
|
// Something is not right.
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
|
|
}
|
|
if (dtls_enabled) {
|
|
if (!tinfo->description.identity_fingerprint) {
|
|
RTC_LOG(LS_WARNING)
|
|
<< "Session description must have DTLS fingerprint if "
|
|
"DTLS enabled.";
|
|
return RTCError(RTCErrorType::INVALID_PARAMETER,
|
|
kSdpWithoutDtlsFingerprint);
|
|
}
|
|
} else {
|
|
if (media->cryptos().empty()) {
|
|
RTC_LOG(LS_WARNING)
|
|
<< "Session description must have SDES when DTLS disabled.";
|
|
return RTCError(RTCErrorType::INVALID_PARAMETER, kSdpWithoutSdesCrypto);
|
|
}
|
|
}
|
|
}
|
|
return RTCError::OK();
|
|
}
|
|
|
|
// Checks that each non-rejected content has ice-ufrag and ice-pwd set, unless
|
|
// it's in a BUNDLE group, in which case only the BUNDLE-tag section (first
|
|
// media section/description in the BUNDLE group) needs a ufrag and pwd.
|
|
bool VerifyIceUfragPwdPresent(const SessionDescription* desc) {
|
|
const cricket::ContentGroup* bundle =
|
|
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
|
|
for (const cricket::ContentInfo& content_info : desc->contents()) {
|
|
if (content_info.rejected) {
|
|
continue;
|
|
}
|
|
const std::string& mid = content_info.name;
|
|
if (bundle && bundle->HasContentName(mid) &&
|
|
mid != *(bundle->FirstContentName())) {
|
|
// This isn't the first media section in the BUNDLE group, so it's not
|
|
// required to have ufrag/password, since only the ufrag/password from
|
|
// the first section actually get used.
|
|
continue;
|
|
}
|
|
|
|
// If the content isn't rejected or bundled into another m= section,
|
|
// ice-ufrag and ice-pwd must be present.
|
|
const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
|
|
if (!tinfo) {
|
|
// Something is not right.
|
|
RTC_LOG(LS_ERROR) << kInvalidSdp;
|
|
return false;
|
|
}
|
|
if (tinfo->description.ice_ufrag.empty() ||
|
|
tinfo->description.ice_pwd.empty()) {
|
|
RTC_LOG(LS_ERROR) << "Session description must have ice ufrag and pwd.";
|
|
return false;
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
|
|
// 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;
|
|
}
|
|
|
|
// Generates a string error message for SetLocalDescription/SetRemoteDescription
|
|
// from an RTCError.
|
|
std::string GetSetDescriptionErrorMessage(cricket::ContentSource source,
|
|
SdpType type,
|
|
const RTCError& error) {
|
|
rtc::StringBuilder oss;
|
|
oss << "Failed to set " << (source == cricket::CS_LOCAL ? "local" : "remote")
|
|
<< " " << SdpTypeToString(type) << " sdp: " << error.message();
|
|
return oss.Release();
|
|
}
|
|
|
|
std::string GetStreamIdsString(rtc::ArrayView<const std::string> stream_ids) {
|
|
std::string output = "streams=[";
|
|
const char* separator = "";
|
|
for (const auto& stream_id : stream_ids) {
|
|
output.append(separator).append(stream_id);
|
|
separator = ", ";
|
|
}
|
|
output.append("]");
|
|
return output;
|
|
}
|
|
|
|
absl::optional<int> RTCConfigurationToIceConfigOptionalInt(
|
|
int rtc_configuration_parameter) {
|
|
if (rtc_configuration_parameter ==
|
|
webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined) {
|
|
return absl::nullopt;
|
|
}
|
|
return rtc_configuration_parameter;
|
|
}
|
|
|
|
void ReportSimulcastApiVersion(const char* name,
|
|
const SessionDescription& session) {
|
|
bool has_legacy = false;
|
|
bool has_spec_compliant = false;
|
|
for (const ContentInfo& content : session.contents()) {
|
|
if (!content.media_description()) {
|
|
continue;
|
|
}
|
|
has_spec_compliant |= content.media_description()->HasSimulcast();
|
|
for (const StreamParams& sp : content.media_description()->streams()) {
|
|
has_legacy |= sp.has_ssrc_group(cricket::kSimSsrcGroupSemantics);
|
|
}
|
|
}
|
|
|
|
if (has_legacy) {
|
|
RTC_HISTOGRAM_ENUMERATION(name, kSimulcastApiVersionLegacy,
|
|
kSimulcastApiVersionMax);
|
|
}
|
|
if (has_spec_compliant) {
|
|
RTC_HISTOGRAM_ENUMERATION(name, kSimulcastApiVersionSpecCompliant,
|
|
kSimulcastApiVersionMax);
|
|
}
|
|
if (!has_legacy && !has_spec_compliant) {
|
|
RTC_HISTOGRAM_ENUMERATION(name, kSimulcastApiVersionNone,
|
|
kSimulcastApiVersionMax);
|
|
}
|
|
}
|
|
|
|
const ContentInfo* FindTransceiverMSection(
|
|
RtpTransceiverProxyWithInternal<RtpTransceiver>* transceiver,
|
|
const SessionDescriptionInterface* session_description) {
|
|
return transceiver->mid()
|
|
? session_description->description()->GetContentByName(
|
|
*transceiver->mid())
|
|
: nullptr;
|
|
}
|
|
|
|
// Wraps a CreateSessionDescriptionObserver and an OperationsChain operation
|
|
// complete callback. When the observer is invoked, the wrapped observer is
|
|
// invoked followed by invoking the completion callback.
|
|
class CreateSessionDescriptionObserverOperationWrapper
|
|
: public CreateSessionDescriptionObserver {
|
|
public:
|
|
CreateSessionDescriptionObserverOperationWrapper(
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserver> observer,
|
|
std::function<void()> operation_complete_callback)
|
|
: observer_(std::move(observer)),
|
|
operation_complete_callback_(std::move(operation_complete_callback)) {
|
|
RTC_DCHECK(observer_);
|
|
}
|
|
~CreateSessionDescriptionObserverOperationWrapper() override {
|
|
RTC_DCHECK(was_called_);
|
|
}
|
|
|
|
void OnSuccess(SessionDescriptionInterface* desc) override {
|
|
RTC_DCHECK(!was_called_);
|
|
#ifdef RTC_DCHECK_IS_ON
|
|
was_called_ = true;
|
|
#endif // RTC_DCHECK_IS_ON
|
|
// Completing the operation before invoking the observer allows the observer
|
|
// to execute SetLocalDescription() without delay.
|
|
operation_complete_callback_();
|
|
observer_->OnSuccess(desc);
|
|
}
|
|
|
|
void OnFailure(RTCError error) override {
|
|
RTC_DCHECK(!was_called_);
|
|
#ifdef RTC_DCHECK_IS_ON
|
|
was_called_ = true;
|
|
#endif // RTC_DCHECK_IS_ON
|
|
operation_complete_callback_();
|
|
observer_->OnFailure(std::move(error));
|
|
}
|
|
|
|
private:
|
|
#ifdef RTC_DCHECK_IS_ON
|
|
bool was_called_ = false;
|
|
#endif // RTC_DCHECK_IS_ON
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserver> observer_;
|
|
std::function<void()> operation_complete_callback_;
|
|
};
|
|
|
|
// Check if the changes of IceTransportsType motives an ice restart.
|
|
bool NeedIceRestart(bool surface_ice_candidates_on_ice_transport_type_changed,
|
|
PeerConnectionInterface::IceTransportsType current,
|
|
PeerConnectionInterface::IceTransportsType modified) {
|
|
if (current == modified) {
|
|
return false;
|
|
}
|
|
|
|
if (!surface_ice_candidates_on_ice_transport_type_changed) {
|
|
return true;
|
|
}
|
|
|
|
auto current_filter = ConvertIceTransportTypeToCandidateFilter(current);
|
|
auto modified_filter = ConvertIceTransportTypeToCandidateFilter(modified);
|
|
|
|
// If surface_ice_candidates_on_ice_transport_type_changed is true and we
|
|
// extend the filter, then no ice restart is needed.
|
|
return (current_filter & modified_filter) != current_filter;
|
|
}
|
|
|
|
} // namespace
|
|
|
|
// Used by parameterless SetLocalDescription() to create an offer or answer.
|
|
// Upon completion of creating the session description, SetLocalDescription() is
|
|
// invoked with the result.
|
|
// For consistency with DoSetLocalDescription(), if the PeerConnection is
|
|
// destroyed midst operation, we DO NOT inform the
|
|
// |set_local_description_observer| that the operation failed.
|
|
// TODO(hbos): If/when we process SLD messages in ~PeerConnection, the
|
|
// consistent thing would be to inform the observer here.
|
|
class PeerConnection::ImplicitCreateSessionDescriptionObserver
|
|
: public CreateSessionDescriptionObserver {
|
|
public:
|
|
ImplicitCreateSessionDescriptionObserver(
|
|
rtc::WeakPtr<PeerConnection> pc,
|
|
rtc::scoped_refptr<SetSessionDescriptionObserver>
|
|
set_local_description_observer)
|
|
: pc_(std::move(pc)),
|
|
set_local_description_observer_(
|
|
std::move(set_local_description_observer)) {}
|
|
~ImplicitCreateSessionDescriptionObserver() override {
|
|
RTC_DCHECK(was_called_);
|
|
}
|
|
|
|
void SetOperationCompleteCallback(
|
|
std::function<void()> operation_complete_callback) {
|
|
operation_complete_callback_ = std::move(operation_complete_callback);
|
|
}
|
|
|
|
bool was_called() const { return was_called_; }
|
|
|
|
void OnSuccess(SessionDescriptionInterface* desc_ptr) override {
|
|
RTC_DCHECK(!was_called_);
|
|
std::unique_ptr<SessionDescriptionInterface> desc(desc_ptr);
|
|
was_called_ = true;
|
|
|
|
// Abort early if |pc_| is no longer valid.
|
|
if (!pc_) {
|
|
operation_complete_callback_();
|
|
return;
|
|
}
|
|
// DoSetLocalDescription() is currently implemented as a synchronous
|
|
// operation but where the |set_local_description_observer_|'s callbacks are
|
|
// invoked asynchronously in a post to PeerConnection::OnMessage().
|
|
pc_->DoSetLocalDescription(std::move(desc),
|
|
std::move(set_local_description_observer_));
|
|
// For backwards-compatability reasons, we declare the operation as
|
|
// completed here (rather than in PeerConnection::OnMessage()). This ensures
|
|
// that subsequent offer/answer operations can start immediately (without
|
|
// waiting for OnMessage()).
|
|
operation_complete_callback_();
|
|
}
|
|
|
|
void OnFailure(RTCError error) override {
|
|
RTC_DCHECK(!was_called_);
|
|
was_called_ = true;
|
|
|
|
// Abort early if |pc_| is no longer valid.
|
|
if (!pc_) {
|
|
operation_complete_callback_();
|
|
return;
|
|
}
|
|
// DoSetLocalDescription() reports its failures in a post. We do the
|
|
// same thing here for consistency.
|
|
pc_->PostSetSessionDescriptionFailure(
|
|
set_local_description_observer_,
|
|
RTCError(error.type(),
|
|
std::string("SetLocalDescription failed to create "
|
|
"session description - ") +
|
|
error.message()));
|
|
operation_complete_callback_();
|
|
}
|
|
|
|
private:
|
|
bool was_called_ = false;
|
|
rtc::WeakPtr<PeerConnection> pc_;
|
|
rtc::scoped_refptr<SetSessionDescriptionObserver>
|
|
set_local_description_observer_;
|
|
std::function<void()> operation_complete_callback_;
|
|
};
|
|
|
|
class PeerConnection::LocalIceCredentialsToReplace {
|
|
public:
|
|
// Sets the ICE credentials that need restarting to the ICE credentials of
|
|
// the current and pending descriptions.
|
|
void SetIceCredentialsFromLocalDescriptions(
|
|
const SessionDescriptionInterface* current_local_description,
|
|
const SessionDescriptionInterface* pending_local_description) {
|
|
ice_credentials_.clear();
|
|
if (current_local_description) {
|
|
AppendIceCredentialsFromSessionDescription(*current_local_description);
|
|
}
|
|
if (pending_local_description) {
|
|
AppendIceCredentialsFromSessionDescription(*pending_local_description);
|
|
}
|
|
}
|
|
|
|
void ClearIceCredentials() { ice_credentials_.clear(); }
|
|
|
|
// Returns true if we have ICE credentials that need restarting.
|
|
bool HasIceCredentials() const { return !ice_credentials_.empty(); }
|
|
|
|
// Returns true if |local_description| shares no ICE credentials with the
|
|
// ICE credentials that need restarting.
|
|
bool SatisfiesIceRestart(
|
|
const SessionDescriptionInterface& local_description) const {
|
|
for (const auto& transport_info :
|
|
local_description.description()->transport_infos()) {
|
|
if (ice_credentials_.find(std::make_pair(
|
|
transport_info.description.ice_ufrag,
|
|
transport_info.description.ice_pwd)) != ice_credentials_.end()) {
|
|
return false;
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
|
|
private:
|
|
void AppendIceCredentialsFromSessionDescription(
|
|
const SessionDescriptionInterface& desc) {
|
|
for (const auto& transport_info : desc.description()->transport_infos()) {
|
|
ice_credentials_.insert(
|
|
std::make_pair(transport_info.description.ice_ufrag,
|
|
transport_info.description.ice_pwd));
|
|
}
|
|
}
|
|
|
|
std::set<std::pair<std::string, std::string>> ice_credentials_;
|
|
};
|
|
|
|
// 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_, std::move(error));
|
|
}
|
|
|
|
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 disable_link_local_networks;
|
|
bool enable_rtp_data_channel;
|
|
absl::optional<int> screencast_min_bitrate;
|
|
absl::optional<bool> combined_audio_video_bwe;
|
|
absl::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 audio_jitter_buffer_min_delay_ms;
|
|
bool audio_jitter_buffer_enable_rtx_handling;
|
|
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;
|
|
PortPrunePolicy turn_port_prune_policy;
|
|
bool presume_writable_when_fully_relayed;
|
|
bool enable_ice_renomination;
|
|
bool redetermine_role_on_ice_restart;
|
|
bool surface_ice_candidates_on_ice_transport_type_changed;
|
|
absl::optional<int> ice_check_interval_strong_connectivity;
|
|
absl::optional<int> ice_check_interval_weak_connectivity;
|
|
absl::optional<int> ice_check_min_interval;
|
|
absl::optional<int> ice_unwritable_timeout;
|
|
absl::optional<int> ice_unwritable_min_checks;
|
|
absl::optional<int> ice_inactive_timeout;
|
|
absl::optional<int> stun_candidate_keepalive_interval;
|
|
webrtc::TurnCustomizer* turn_customizer;
|
|
SdpSemantics sdp_semantics;
|
|
absl::optional<rtc::AdapterType> network_preference;
|
|
bool active_reset_srtp_params;
|
|
absl::optional<CryptoOptions> crypto_options;
|
|
bool offer_extmap_allow_mixed;
|
|
std::string turn_logging_id;
|
|
bool enable_implicit_rollback;
|
|
absl::optional<bool> allow_codec_switching;
|
|
};
|
|
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 &&
|
|
audio_jitter_buffer_min_delay_ms ==
|
|
o.audio_jitter_buffer_min_delay_ms &&
|
|
audio_jitter_buffer_enable_rtx_handling ==
|
|
o.audio_jitter_buffer_enable_rtx_handling &&
|
|
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 &&
|
|
disable_link_local_networks == o.disable_link_local_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 &&
|
|
turn_port_prune_policy == o.turn_port_prune_policy &&
|
|
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 &&
|
|
surface_ice_candidates_on_ice_transport_type_changed ==
|
|
o.surface_ice_candidates_on_ice_transport_type_changed &&
|
|
ice_check_interval_strong_connectivity ==
|
|
o.ice_check_interval_strong_connectivity &&
|
|
ice_check_interval_weak_connectivity ==
|
|
o.ice_check_interval_weak_connectivity &&
|
|
ice_check_min_interval == o.ice_check_min_interval &&
|
|
ice_unwritable_timeout == o.ice_unwritable_timeout &&
|
|
ice_unwritable_min_checks == o.ice_unwritable_min_checks &&
|
|
ice_inactive_timeout == o.ice_inactive_timeout &&
|
|
stun_candidate_keepalive_interval ==
|
|
o.stun_candidate_keepalive_interval &&
|
|
turn_customizer == o.turn_customizer &&
|
|
sdp_semantics == o.sdp_semantics &&
|
|
network_preference == o.network_preference &&
|
|
active_reset_srtp_params == o.active_reset_srtp_params &&
|
|
crypto_options == o.crypto_options &&
|
|
offer_extmap_allow_mixed == o.offer_extmap_allow_mixed &&
|
|
turn_logging_id == o.turn_logging_id &&
|
|
enable_implicit_rollback == o.enable_implicit_rollback &&
|
|
allow_codec_switching == o.allow_codec_switching;
|
|
}
|
|
|
|
bool PeerConnectionInterface::RTCConfiguration::operator!=(
|
|
const PeerConnectionInterface::RTCConfiguration& o) const {
|
|
return !(*this == o);
|
|
}
|
|
|
|
void PeerConnection::TransceiverStableState::set_newly_created() {
|
|
RTC_DCHECK(!has_m_section_);
|
|
newly_created_ = true;
|
|
}
|
|
|
|
void PeerConnection::TransceiverStableState::SetMSectionIfUnset(
|
|
absl::optional<std::string> mid,
|
|
absl::optional<size_t> mline_index) {
|
|
if (!has_m_section_) {
|
|
mid_ = mid;
|
|
mline_index_ = mline_index;
|
|
has_m_section_ = true;
|
|
}
|
|
}
|
|
|
|
void PeerConnection::TransceiverStableState::SetRemoteStreamIdsIfUnset(
|
|
const std::vector<std::string>& ids) {
|
|
if (!remote_stream_ids_.has_value()) {
|
|
remote_stream_ids_ = ids;
|
|
}
|
|
}
|
|
|
|
// 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;
|
|
session_options->raw_packetization_for_video =
|
|
rtc_options.raw_packetization_for_video;
|
|
}
|
|
|
|
PeerConnection::PeerConnection(PeerConnectionFactory* factory,
|
|
std::unique_ptr<RtcEventLog> event_log,
|
|
std::unique_ptr<Call> call)
|
|
: factory_(factory),
|
|
event_log_(std::move(event_log)),
|
|
event_log_ptr_(event_log_.get()),
|
|
operations_chain_(rtc::OperationsChain::Create()),
|
|
rtcp_cname_(GenerateRtcpCname()),
|
|
local_streams_(StreamCollection::Create()),
|
|
remote_streams_(StreamCollection::Create()),
|
|
call_(std::move(call)),
|
|
call_ptr_(call_.get()),
|
|
local_ice_credentials_to_replace_(new LocalIceCredentialsToReplace()),
|
|
data_channel_controller_(this),
|
|
weak_ptr_factory_(this) {}
|
|
|
|
PeerConnection::~PeerConnection() {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
|
|
weak_ptr_factory_.InvalidateWeakPtrs();
|
|
|
|
// Need to stop transceivers before destroying the stats collector because
|
|
// AudioRtpSender has a reference to the StatsCollector it will update when
|
|
// stopping.
|
|
for (const 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_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] {
|
|
RTC_DCHECK_RUN_ON(network_thread());
|
|
port_allocator_.reset();
|
|
});
|
|
// call_ and event_log_ must be destroyed on the worker thread.
|
|
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
call_.reset();
|
|
// The event log must outlive call (and any other object that uses it).
|
|
event_log_.reset();
|
|
});
|
|
|
|
// Process all pending notifications in the message queue. If we don't do
|
|
// this, requests will linger and not know they succeeded or failed.
|
|
rtc::MessageList list;
|
|
signaling_thread()->Clear(this, rtc::MQID_ANY, &list);
|
|
for (auto& msg : list) {
|
|
if (msg.message_id == MSG_CREATE_SESSIONDESCRIPTION_FAILED) {
|
|
// Processing CreateOffer() and CreateAnswer() messages ensures their
|
|
// observers are invoked even if the PeerConnection is destroyed early.
|
|
OnMessage(&msg);
|
|
} else {
|
|
// TODO(hbos): Consider processing all pending messages. This would mean
|
|
// that SetLocalDescription() and SetRemoteDescription() observers are
|
|
// informed of successes and failures; this is currently NOT the case.
|
|
delete msg.pdata;
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::DestroyAllChannels() {
|
|
// Destroy video channels first since they may have a pointer to a voice
|
|
// channel.
|
|
for (const auto& transceiver : transceivers_) {
|
|
if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
|
|
DestroyTransceiverChannel(transceiver);
|
|
}
|
|
}
|
|
for (const auto& transceiver : transceivers_) {
|
|
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
|
DestroyTransceiverChannel(transceiver);
|
|
}
|
|
}
|
|
DestroyDataChannelTransport();
|
|
}
|
|
|
|
bool PeerConnection::Initialize(
|
|
const PeerConnectionInterface::RTCConfiguration& configuration,
|
|
PeerConnectionDependencies dependencies) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
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 (!dependencies.allocator) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "PeerConnection initialized without a PortAllocator? "
|
|
"This shouldn't happen if using PeerConnectionFactory.";
|
|
return false;
|
|
}
|
|
|
|
if (!dependencies.observer) {
|
|
// TODO(deadbeef): Why do we do this?
|
|
RTC_LOG(LS_ERROR) << "PeerConnection initialized without a "
|
|
"PeerConnectionObserver";
|
|
return false;
|
|
}
|
|
|
|
observer_ = dependencies.observer;
|
|
async_resolver_factory_ = std::move(dependencies.async_resolver_factory);
|
|
port_allocator_ = std::move(dependencies.allocator);
|
|
packet_socket_factory_ = std::move(dependencies.packet_socket_factory);
|
|
ice_transport_factory_ = std::move(dependencies.ice_transport_factory);
|
|
tls_cert_verifier_ = std::move(dependencies.tls_cert_verifier);
|
|
|
|
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 false;
|
|
}
|
|
|
|
// Add the turn logging id to all turn servers
|
|
for (cricket::RelayServerConfig& turn_server : turn_servers) {
|
|
turn_server.turn_logging_id = configuration.turn_logging_id;
|
|
}
|
|
|
|
// The port allocator lives on the network thread and should be initialized
|
|
// there.
|
|
const auto pa_result =
|
|
network_thread()->Invoke<InitializePortAllocatorResult>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&PeerConnection::InitializePortAllocator_n, this,
|
|
stun_servers, turn_servers, configuration));
|
|
|
|
// If initialization was successful, note if STUN or TURN servers
|
|
// were supplied.
|
|
if (!stun_servers.empty()) {
|
|
NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
|
|
}
|
|
if (!turn_servers.empty()) {
|
|
NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
|
|
}
|
|
|
|
// Send information about IPv4/IPv6 status.
|
|
PeerConnectionAddressFamilyCounter address_family;
|
|
if (pa_result.enable_ipv6) {
|
|
address_family = kPeerConnection_IPv6;
|
|
} else {
|
|
address_family = kPeerConnection_IPv4;
|
|
}
|
|
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics", address_family,
|
|
kPeerConnectionAddressFamilyCounter_Max);
|
|
|
|
const PeerConnectionFactoryInterface::Options& options = factory_->options();
|
|
|
|
// 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);
|
|
JsepTransportController::Config config;
|
|
config.redetermine_role_on_ice_restart =
|
|
configuration.redetermine_role_on_ice_restart;
|
|
config.ssl_max_version = factory_->options().ssl_max_version;
|
|
config.disable_encryption = options.disable_encryption;
|
|
config.bundle_policy = configuration.bundle_policy;
|
|
config.rtcp_mux_policy = configuration.rtcp_mux_policy;
|
|
// TODO(bugs.webrtc.org/9891) - Remove options.crypto_options then remove this
|
|
// stub.
|
|
config.crypto_options = configuration.crypto_options.has_value()
|
|
? *configuration.crypto_options
|
|
: options.crypto_options;
|
|
config.transport_observer = this;
|
|
// It's safe to pass |this| and using |rtcp_invoker_| and the |call_| pointer
|
|
// since the JsepTransportController instance is owned by this PeerConnection
|
|
// instance and is destroyed before both |rtcp_invoker_| and the |call_|
|
|
// pointer.
|
|
config.rtcp_handler = [this](const rtc::CopyOnWriteBuffer& packet,
|
|
int64_t packet_time_us) {
|
|
RTC_DCHECK_RUN_ON(network_thread());
|
|
rtcp_invoker_.AsyncInvoke<void>(
|
|
RTC_FROM_HERE, worker_thread(), [this, packet, packet_time_us] {
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
// |call_| is reset on the worker thread in the PeerConnection
|
|
// destructor, so we check that it's still valid before propagating
|
|
// the packet.
|
|
if (call_) {
|
|
call_->Receiver()->DeliverPacket(MediaType::ANY, packet,
|
|
packet_time_us);
|
|
}
|
|
});
|
|
};
|
|
config.event_log = event_log_ptr_;
|
|
#if defined(ENABLE_EXTERNAL_AUTH)
|
|
config.enable_external_auth = true;
|
|
#endif
|
|
config.active_reset_srtp_params = configuration.active_reset_srtp_params;
|
|
|
|
// 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];
|
|
}
|
|
|
|
if (options.disable_encryption) {
|
|
dtls_enabled_ = false;
|
|
} else {
|
|
// Enable DTLS by default if we have an identity store or a certificate.
|
|
dtls_enabled_ = (dependencies.cert_generator || certificate);
|
|
// |configuration| can override the default |dtls_enabled_| value.
|
|
if (configuration.enable_dtls_srtp) {
|
|
dtls_enabled_ = *(configuration.enable_dtls_srtp);
|
|
}
|
|
}
|
|
|
|
sctp_factory_ = factory_->CreateSctpTransportInternalFactory();
|
|
|
|
if (configuration.enable_rtp_data_channel) {
|
|
// Enable creation of RTP data channels if the kEnableRtpDataChannels is
|
|
// set. It takes precendence over the disable_sctp_data_channels
|
|
// PeerConnectionFactoryInterface::Options.
|
|
data_channel_controller_.set_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_controller_.set_data_channel_type(cricket::DCT_SCTP);
|
|
config.sctp_factory = sctp_factory_.get();
|
|
}
|
|
}
|
|
|
|
config.ice_transport_factory = ice_transport_factory_.get();
|
|
|
|
transport_controller_.reset(new JsepTransportController(
|
|
signaling_thread(), network_thread(), port_allocator_.get(),
|
|
async_resolver_factory_.get(), config));
|
|
transport_controller_->SignalIceConnectionState.connect(
|
|
this, &PeerConnection::OnTransportControllerConnectionState);
|
|
transport_controller_->SignalStandardizedIceConnectionState.connect(
|
|
this, &PeerConnection::SetStandardizedIceConnectionState);
|
|
transport_controller_->SignalConnectionState.connect(
|
|
this, &PeerConnection::SetConnectionState);
|
|
transport_controller_->SignalIceGatheringState.connect(
|
|
this, &PeerConnection::OnTransportControllerGatheringState);
|
|
transport_controller_->SignalIceCandidatesGathered.connect(
|
|
this, &PeerConnection::OnTransportControllerCandidatesGathered);
|
|
transport_controller_->SignalIceCandidateError.connect(
|
|
this, &PeerConnection::OnTransportControllerCandidateError);
|
|
transport_controller_->SignalIceCandidatesRemoved.connect(
|
|
this, &PeerConnection::OnTransportControllerCandidatesRemoved);
|
|
transport_controller_->SignalDtlsHandshakeError.connect(
|
|
this, &PeerConnection::OnTransportControllerDtlsHandshakeError);
|
|
transport_controller_->SignalIceCandidatePairChanged.connect(
|
|
this, &PeerConnection::OnTransportControllerCandidateChanged);
|
|
|
|
stats_.reset(new StatsCollector(this));
|
|
stats_collector_ = RTCStatsCollector::Create(this);
|
|
|
|
configuration_ = configuration;
|
|
|
|
transport_controller_->SetIceConfig(ParseIceConfig(configuration));
|
|
|
|
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;
|
|
|
|
audio_options_.audio_jitter_buffer_min_delay_ms =
|
|
configuration.audio_jitter_buffer_min_delay_ms;
|
|
|
|
audio_options_.audio_jitter_buffer_enable_rtx_handling =
|
|
configuration.audio_jitter_buffer_enable_rtx_handling;
|
|
|
|
// 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_) {
|
|
dependencies.cert_generator.reset();
|
|
certificate = nullptr;
|
|
} else if (certificate) {
|
|
// Favor generated certificate over the certificate generator.
|
|
dependencies.cert_generator.reset();
|
|
}
|
|
|
|
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
|
|
signaling_thread(), channel_manager(), this, session_id(),
|
|
std::move(dependencies.cert_generator), certificate, &ssrc_generator_));
|
|
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(
|
|
GetCryptoOptions().srtp.enable_encrypted_rtp_header_extensions);
|
|
webrtc_session_desc_factory_->set_is_unified_plan(IsUnifiedPlan());
|
|
|
|
// 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)));
|
|
}
|
|
int delay_ms =
|
|
return_histogram_very_quickly_ ? 0 : REPORT_USAGE_PATTERN_DELAY_MS;
|
|
signaling_thread()->PostDelayed(RTC_FROM_HERE, delay_ms, this,
|
|
MSG_REPORT_USAGE_PATTERN, nullptr);
|
|
|
|
if (dependencies.video_bitrate_allocator_factory) {
|
|
video_bitrate_allocator_factory_ =
|
|
std::move(dependencies.video_bitrate_allocator_factory);
|
|
} else {
|
|
video_bitrate_allocator_factory_ =
|
|
CreateBuiltinVideoBitrateAllocatorFactory();
|
|
}
|
|
return true;
|
|
}
|
|
|
|
RTCError PeerConnection::ValidateConfiguration(
|
|
const RTCConfiguration& config) const {
|
|
return cricket::P2PTransportChannel::ValidateIceConfig(
|
|
ParseIceConfig(config));
|
|
}
|
|
|
|
rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::local_streams() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(!IsUnifiedPlan()) << "local_streams is not available with Unified "
|
|
"Plan SdpSemantics. Please use GetSenders "
|
|
"instead.";
|
|
return local_streams_;
|
|
}
|
|
|
|
rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::remote_streams() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(!IsUnifiedPlan()) << "remote_streams is not available with Unified "
|
|
"Plan SdpSemantics. Please use GetReceivers "
|
|
"instead.";
|
|
return remote_streams_;
|
|
}
|
|
|
|
bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(!IsUnifiedPlan()) << "AddStream is not available with Unified Plan "
|
|
"SdpSemantics. Please use AddTrack instead.";
|
|
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);
|
|
UpdateNegotiationNeeded();
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(!IsUnifiedPlan()) << "RemoveStream is not available with Unified "
|
|
"Plan SdpSemantics. Please use RemoveTrack "
|
|
"instead.";
|
|
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()->id().compare(local_stream->id()) == 0;
|
|
}),
|
|
stream_observers_.end());
|
|
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const std::vector<std::string>& stream_ids) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "PeerConnection::AddTrack");
|
|
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());
|
|
}
|
|
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() + ".");
|
|
}
|
|
auto sender_or_error =
|
|
(IsUnifiedPlan() ? AddTrackUnifiedPlan(track, stream_ids)
|
|
: AddTrackPlanB(track, stream_ids));
|
|
if (sender_or_error.ok()) {
|
|
UpdateNegotiationNeeded();
|
|
stats_->AddTrack(track);
|
|
}
|
|
return sender_or_error;
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
|
|
PeerConnection::AddTrackPlanB(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const std::vector<std::string>& stream_ids) {
|
|
if (stream_ids.size() > 1u) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
|
|
"AddTrack with more than one stream is not "
|
|
"supported with Plan B semantics.");
|
|
}
|
|
std::vector<std::string> adjusted_stream_ids = stream_ids;
|
|
if (adjusted_stream_ids.empty()) {
|
|
adjusted_stream_ids.push_back(rtc::CreateRandomUuid());
|
|
}
|
|
cricket::MediaType media_type =
|
|
(track->kind() == MediaStreamTrackInterface::kAudioKind
|
|
? cricket::MEDIA_TYPE_AUDIO
|
|
: cricket::MEDIA_TYPE_VIDEO);
|
|
auto new_sender =
|
|
CreateSender(media_type, track->id(), track, adjusted_stream_ids, {});
|
|
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
|
|
new_sender->internal()->SetMediaChannel(voice_media_channel());
|
|
GetAudioTransceiver()->internal()->AddSender(new_sender);
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(local_audio_sender_infos_,
|
|
new_sender->internal()->stream_ids()[0], track->id());
|
|
if (sender_info) {
|
|
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
|
|
}
|
|
} else {
|
|
RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
|
|
new_sender->internal()->SetMediaChannel(video_media_channel());
|
|
GetVideoTransceiver()->internal()->AddSender(new_sender);
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(local_video_sender_infos_,
|
|
new_sender->internal()->stream_ids()[0], 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_ids) {
|
|
auto transceiver = FindFirstTransceiverForAddedTrack(track);
|
|
if (transceiver) {
|
|
RTC_LOG(LS_INFO) << "Reusing an existing "
|
|
<< cricket::MediaTypeToString(transceiver->media_type())
|
|
<< " transceiver for AddTrack.";
|
|
if (transceiver->direction() == RtpTransceiverDirection::kRecvOnly) {
|
|
transceiver->internal()->set_direction(
|
|
RtpTransceiverDirection::kSendRecv);
|
|
} else if (transceiver->direction() == RtpTransceiverDirection::kInactive) {
|
|
transceiver->internal()->set_direction(
|
|
RtpTransceiverDirection::kSendOnly);
|
|
}
|
|
transceiver->sender()->SetTrack(track);
|
|
transceiver->internal()->sender_internal()->set_stream_ids(stream_ids);
|
|
transceiver->internal()->set_reused_for_addtrack(true);
|
|
} else {
|
|
cricket::MediaType media_type =
|
|
(track->kind() == MediaStreamTrackInterface::kAudioKind
|
|
? cricket::MEDIA_TYPE_AUDIO
|
|
: cricket::MEDIA_TYPE_VIDEO);
|
|
RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
|
|
<< " transceiver in response to a call to AddTrack.";
|
|
std::string sender_id = track->id();
|
|
// Avoid creating a sender with an existing ID by generating a random ID.
|
|
// This can happen if this is the second time AddTrack has created a sender
|
|
// for this track.
|
|
if (FindSenderById(sender_id)) {
|
|
sender_id = rtc::CreateRandomUuid();
|
|
}
|
|
auto sender = CreateSender(media_type, sender_id, track, stream_ids, {});
|
|
auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());
|
|
transceiver = CreateAndAddTransceiver(sender, receiver);
|
|
transceiver->internal()->set_created_by_addtrack(true);
|
|
transceiver->internal()->set_direction(RtpTransceiverDirection::kSendRecv);
|
|
}
|
|
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->media_type()) ==
|
|
track->kind() &&
|
|
!transceiver->internal()->has_ever_been_used_to_send() &&
|
|
!transceiver->stopped()) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
bool PeerConnection::RemoveTrack(RtpSenderInterface* sender) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::RemoveTrack");
|
|
return RemoveTrackNew(sender).ok();
|
|
}
|
|
|
|
RTCError PeerConnection::RemoveTrackNew(
|
|
rtc::scoped_refptr<RtpSenderInterface> sender) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
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()->set_direction(
|
|
RtpTransceiverDirection::kRecvOnly);
|
|
} else if (transceiver->direction() == RtpTransceiverDirection::kSendOnly) {
|
|
transceiver->internal()->set_direction(
|
|
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.");
|
|
}
|
|
}
|
|
UpdateNegotiationNeeded();
|
|
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) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(IsUnifiedPlan())
|
|
<< "AddTransceiver is only available with Unified Plan SdpSemantics";
|
|
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) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(IsUnifiedPlan())
|
|
<< "AddTransceiver is only available with Unified Plan SdpSemantics";
|
|
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,
|
|
bool update_negotiation_needed) {
|
|
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));
|
|
}
|
|
|
|
RTC_HISTOGRAM_COUNTS_LINEAR(kSimulcastNumberOfEncodings,
|
|
init.send_encodings.size(), 0, 7, 8);
|
|
|
|
size_t num_rids = absl::c_count_if(init.send_encodings,
|
|
[](const RtpEncodingParameters& encoding) {
|
|
return !encoding.rid.empty();
|
|
});
|
|
if (num_rids > 0 && num_rids != init.send_encodings.size()) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::INVALID_PARAMETER,
|
|
"RIDs must be provided for either all or none of the send encodings.");
|
|
}
|
|
|
|
if (num_rids > 0 && absl::c_any_of(init.send_encodings,
|
|
[](const RtpEncodingParameters& encoding) {
|
|
return !IsLegalRsidName(encoding.rid);
|
|
})) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"Invalid RID value provided.");
|
|
}
|
|
|
|
if (absl::c_any_of(init.send_encodings,
|
|
[](const RtpEncodingParameters& encoding) {
|
|
return encoding.ssrc.has_value();
|
|
})) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::UNSUPPORTED_PARAMETER,
|
|
"Attempted to set an unimplemented parameter of RtpParameters.");
|
|
}
|
|
|
|
RtpParameters parameters;
|
|
parameters.encodings = init.send_encodings;
|
|
|
|
// Encodings are dropped from the tail if too many are provided.
|
|
if (parameters.encodings.size() > kMaxSimulcastStreams) {
|
|
parameters.encodings.erase(
|
|
parameters.encodings.begin() + kMaxSimulcastStreams,
|
|
parameters.encodings.end());
|
|
}
|
|
|
|
// Single RID should be removed.
|
|
if (parameters.encodings.size() == 1 &&
|
|
!parameters.encodings[0].rid.empty()) {
|
|
RTC_LOG(LS_INFO) << "Removing RID: " << parameters.encodings[0].rid << ".";
|
|
parameters.encodings[0].rid.clear();
|
|
}
|
|
|
|
// If RIDs were not provided, they are generated for simulcast scenario.
|
|
if (parameters.encodings.size() > 1 && num_rids == 0) {
|
|
rtc::UniqueStringGenerator rid_generator;
|
|
for (RtpEncodingParameters& encoding : parameters.encodings) {
|
|
encoding.rid = rid_generator();
|
|
}
|
|
}
|
|
|
|
if (UnimplementedRtpParameterHasValue(parameters)) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::UNSUPPORTED_PARAMETER,
|
|
"Attempted to set an unimplemented parameter of RtpParameters.");
|
|
}
|
|
|
|
auto result = cricket::CheckRtpParametersValues(parameters);
|
|
if (!result.ok()) {
|
|
LOG_AND_RETURN_ERROR(result.type(), result.message());
|
|
}
|
|
|
|
RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
|
|
<< " transceiver in response to a call to AddTransceiver.";
|
|
// Set the sender ID equal to the track ID if the track is specified unless
|
|
// that sender ID is already in use.
|
|
std::string sender_id =
|
|
(track && !FindSenderById(track->id()) ? track->id()
|
|
: rtc::CreateRandomUuid());
|
|
auto sender = CreateSender(media_type, sender_id, track, init.stream_ids,
|
|
parameters.encodings);
|
|
auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());
|
|
auto transceiver = CreateAndAddTransceiver(sender, receiver);
|
|
transceiver->internal()->set_direction(init.direction);
|
|
|
|
if (update_negotiation_needed) {
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
return rtc::scoped_refptr<RtpTransceiverInterface>(transceiver);
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
|
|
PeerConnection::CreateSender(
|
|
cricket::MediaType media_type,
|
|
const std::string& id,
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const std::vector<std::string>& stream_ids,
|
|
const std::vector<RtpEncodingParameters>& send_encodings) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender;
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
|
RTC_DCHECK(!track ||
|
|
(track->kind() == MediaStreamTrackInterface::kAudioKind));
|
|
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(),
|
|
AudioRtpSender::Create(worker_thread(), id, stats_.get(), this));
|
|
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
|
|
} else {
|
|
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
|
|
RTC_DCHECK(!track ||
|
|
(track->kind() == MediaStreamTrackInterface::kVideoKind));
|
|
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(), VideoRtpSender::Create(worker_thread(), id, this));
|
|
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
|
|
}
|
|
bool set_track_succeeded = sender->SetTrack(track);
|
|
RTC_DCHECK(set_track_succeeded);
|
|
sender->internal()->set_stream_ids(stream_ids);
|
|
sender->internal()->set_init_send_encodings(send_encodings);
|
|
return sender;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
PeerConnection::CreateReceiver(cricket::MediaType media_type,
|
|
const std::string& receiver_id) {
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
receiver;
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
|
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
|
signaling_thread(), new AudioRtpReceiver(worker_thread(), receiver_id,
|
|
std::vector<std::string>({})));
|
|
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
|
|
} else {
|
|
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
|
|
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
|
signaling_thread(), new VideoRtpReceiver(worker_thread(), receiver_id,
|
|
std::vector<std::string>({})));
|
|
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
|
|
}
|
|
return receiver;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::CreateAndAddTransceiver(
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender,
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
receiver) {
|
|
// Ensure that the new sender does not have an ID that is already in use by
|
|
// another sender.
|
|
// Allow receiver IDs to conflict since those come from remote SDP (which
|
|
// could be invalid, but should not cause a crash).
|
|
RTC_DCHECK(!FindSenderById(sender->id()));
|
|
auto transceiver = RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
|
|
signaling_thread(),
|
|
new RtpTransceiver(
|
|
sender, receiver, channel_manager(),
|
|
sender->media_type() == cricket::MEDIA_TYPE_AUDIO
|
|
? channel_manager()->GetSupportedAudioRtpHeaderExtensions()
|
|
: channel_manager()->GetSupportedVideoRtpHeaderExtensions()));
|
|
transceivers_.push_back(transceiver);
|
|
transceiver->internal()->SignalNegotiationNeeded.connect(
|
|
this, &PeerConnection::OnNegotiationNeeded);
|
|
return transceiver;
|
|
}
|
|
|
|
void PeerConnection::OnNegotiationNeeded() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(!IsClosed());
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
|
|
const std::string& kind,
|
|
const std::string& stream_id) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(!IsUnifiedPlan()) << "CreateSender is not available with Unified "
|
|
"Plan SdpSemantics. Please use AddTransceiver "
|
|
"instead.";
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
|
|
if (IsClosed()) {
|
|
return nullptr;
|
|
}
|
|
|
|
// Internally we need to have one stream with Plan B semantics, so we
|
|
// generate a random stream ID if not specified.
|
|
std::vector<std::string> stream_ids;
|
|
if (stream_id.empty()) {
|
|
stream_ids.push_back(rtc::CreateRandomUuid());
|
|
RTC_LOG(LS_INFO)
|
|
<< "No stream_id specified for sender. Generated stream ID: "
|
|
<< stream_ids[0];
|
|
} else {
|
|
stream_ids.push_back(stream_id);
|
|
}
|
|
|
|
// TODO(steveanton): Move construction of the RtpSenders to RtpTransceiver.
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
|
|
if (kind == MediaStreamTrackInterface::kAudioKind) {
|
|
auto audio_sender = AudioRtpSender::Create(
|
|
worker_thread(), rtc::CreateRandomUuid(), stats_.get(), this);
|
|
audio_sender->SetMediaChannel(voice_media_channel());
|
|
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(), audio_sender);
|
|
GetAudioTransceiver()->internal()->AddSender(new_sender);
|
|
} else if (kind == MediaStreamTrackInterface::kVideoKind) {
|
|
auto video_sender =
|
|
VideoRtpSender::Create(worker_thread(), rtc::CreateRandomUuid(), this);
|
|
video_sender->SetMediaChannel(video_media_channel());
|
|
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(), video_sender);
|
|
GetVideoTransceiver()->internal()->AddSender(new_sender);
|
|
} else {
|
|
RTC_LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
|
|
return nullptr;
|
|
}
|
|
new_sender->internal()->set_stream_ids(stream_ids);
|
|
|
|
return new_sender;
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
std::vector<rtc::scoped_refptr<RtpSenderInterface>> ret;
|
|
for (const 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 (const 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 {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
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 (const 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_RUN_ON(signaling_thread());
|
|
RTC_CHECK(IsUnifiedPlan())
|
|
<< "GetTransceivers is only supported with Unified Plan SdpSemantics.";
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers;
|
|
for (const 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_RUN_ON(signaling_thread());
|
|
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) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(stats_collector_);
|
|
RTC_DCHECK(callback);
|
|
stats_collector_->GetStatsReport(callback);
|
|
}
|
|
|
|
void PeerConnection::GetStats(
|
|
rtc::scoped_refptr<RtpSenderInterface> selector,
|
|
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(callback);
|
|
RTC_DCHECK(stats_collector_);
|
|
rtc::scoped_refptr<RtpSenderInternal> internal_sender;
|
|
if (selector) {
|
|
for (const auto& proxy_transceiver : transceivers_) {
|
|
for (const auto& proxy_sender :
|
|
proxy_transceiver->internal()->senders()) {
|
|
if (proxy_sender == selector) {
|
|
internal_sender = proxy_sender->internal();
|
|
break;
|
|
}
|
|
}
|
|
if (internal_sender)
|
|
break;
|
|
}
|
|
}
|
|
// If there is no |internal_sender| then |selector| is either null or does not
|
|
// belong to the PeerConnection (in Plan B, senders can be removed from the
|
|
// PeerConnection). This means that "all the stats objects representing the
|
|
// selector" is an empty set. Invoking GetStatsReport() with a null selector
|
|
// produces an empty stats report.
|
|
stats_collector_->GetStatsReport(internal_sender, callback);
|
|
}
|
|
|
|
void PeerConnection::GetStats(
|
|
rtc::scoped_refptr<RtpReceiverInterface> selector,
|
|
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(callback);
|
|
RTC_DCHECK(stats_collector_);
|
|
rtc::scoped_refptr<RtpReceiverInternal> internal_receiver;
|
|
if (selector) {
|
|
for (const auto& proxy_transceiver : transceivers_) {
|
|
for (const auto& proxy_receiver :
|
|
proxy_transceiver->internal()->receivers()) {
|
|
if (proxy_receiver == selector) {
|
|
internal_receiver = proxy_receiver->internal();
|
|
break;
|
|
}
|
|
}
|
|
if (internal_receiver)
|
|
break;
|
|
}
|
|
}
|
|
// If there is no |internal_receiver| then |selector| is either null or does
|
|
// not belong to the PeerConnection (in Plan B, receivers can be removed from
|
|
// the PeerConnection). This means that "all the stats objects representing
|
|
// the selector" is an empty set. Invoking GetStatsReport() with a null
|
|
// selector produces an empty stats report.
|
|
stats_collector_->GetStatsReport(internal_receiver, callback);
|
|
}
|
|
|
|
PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return signaling_state_;
|
|
}
|
|
|
|
PeerConnectionInterface::IceConnectionState
|
|
PeerConnection::ice_connection_state() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return ice_connection_state_;
|
|
}
|
|
|
|
PeerConnectionInterface::IceConnectionState
|
|
PeerConnection::standardized_ice_connection_state() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return standardized_ice_connection_state_;
|
|
}
|
|
|
|
PeerConnectionInterface::PeerConnectionState
|
|
PeerConnection::peer_connection_state() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return connection_state_;
|
|
}
|
|
|
|
PeerConnectionInterface::IceGatheringState
|
|
PeerConnection::ice_gathering_state() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return ice_gathering_state_;
|
|
}
|
|
|
|
absl::optional<bool> PeerConnection::can_trickle_ice_candidates() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
SessionDescriptionInterface* description = current_remote_description_.get();
|
|
if (!description) {
|
|
description = pending_remote_description_.get();
|
|
}
|
|
if (!description) {
|
|
return absl::nullopt;
|
|
}
|
|
// TODO(bugs.webrtc.org/7443): Change to retrieve from session-level option.
|
|
if (description->description()->transport_infos().size() < 1) {
|
|
return absl::nullopt;
|
|
}
|
|
return description->description()->transport_infos()[0].description.HasOption(
|
|
"trickle");
|
|
}
|
|
|
|
rtc::scoped_refptr<DataChannelInterface> PeerConnection::CreateDataChannel(
|
|
const std::string& label,
|
|
const DataChannelInit* config) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
|
|
|
|
bool first_datachannel = !data_channel_controller_.HasDataChannels();
|
|
|
|
std::unique_ptr<InternalDataChannelInit> internal_config;
|
|
if (config) {
|
|
internal_config.reset(new InternalDataChannelInit(*config));
|
|
}
|
|
rtc::scoped_refptr<DataChannelInterface> channel(
|
|
data_channel_controller_.InternalCreateDataChannelWithProxy(
|
|
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) {
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
NoteUsageEvent(UsageEvent::DATA_ADDED);
|
|
return channel;
|
|
}
|
|
|
|
void PeerConnection::RestartIce() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
local_ice_credentials_to_replace_->SetIceCredentialsFromLocalDescriptions(
|
|
current_local_description_.get(), pending_local_description_.get());
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
|
|
const RTCOfferAnswerOptions& options) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
|
|
observer_refptr =
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserver>(observer),
|
|
options](std::function<void()> operations_chain_callback) {
|
|
// Abort early if |this_weak_ptr| is no longer valid.
|
|
if (!this_weak_ptr) {
|
|
observer_refptr->OnFailure(
|
|
RTCError(RTCErrorType::INTERNAL_ERROR,
|
|
"CreateOffer failed because the session was shut down"));
|
|
operations_chain_callback();
|
|
return;
|
|
}
|
|
// The operation completes asynchronously when the wrapper is invoked.
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserverOperationWrapper>
|
|
observer_wrapper(new rtc::RefCountedObject<
|
|
CreateSessionDescriptionObserverOperationWrapper>(
|
|
std::move(observer_refptr),
|
|
std::move(operations_chain_callback)));
|
|
this_weak_ptr->DoCreateOffer(options, observer_wrapper);
|
|
});
|
|
}
|
|
|
|
void PeerConnection::DoCreateOffer(
|
|
const RTCOfferAnswerOptions& options,
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserver> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "PeerConnection::DoCreateOffer");
|
|
|
|
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, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
|
|
return;
|
|
}
|
|
|
|
// If a session error has occurred the PeerConnection is in a possibly
|
|
// inconsistent state so fail right away.
|
|
if (session_error() != SessionError::kNone) {
|
|
std::string error_message = GetSessionErrorMsg();
|
|
RTC_LOG(LS_ERROR) << "CreateOffer: " << error_message;
|
|
PostCreateSessionDescriptionFailure(
|
|
observer,
|
|
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
|
return;
|
|
}
|
|
|
|
if (!ValidateOfferAnswerOptions(options)) {
|
|
std::string error = "CreateOffer called with invalid options.";
|
|
RTC_LOG(LS_ERROR) << error;
|
|
PostCreateSessionDescriptionFailure(
|
|
observer, RTCError(RTCErrorType::INVALID_PARAMETER, std::move(error)));
|
|
return;
|
|
}
|
|
|
|
// Legacy handling for offer_to_receive_audio and offer_to_receive_video.
|
|
// Specified in WebRTC section 4.4.3.2 "Legacy configuration extensions".
|
|
if (IsUnifiedPlan()) {
|
|
RTCError error = HandleLegacyOfferOptions(options);
|
|
if (!error.ok()) {
|
|
PostCreateSessionDescriptionFailure(observer, std::move(error));
|
|
return;
|
|
}
|
|
}
|
|
|
|
cricket::MediaSessionOptions session_options;
|
|
GetOptionsForOffer(options, &session_options);
|
|
webrtc_session_desc_factory_->CreateOffer(observer, options, session_options);
|
|
}
|
|
|
|
RTCError PeerConnection::HandleLegacyOfferOptions(
|
|
const RTCOfferAnswerOptions& options) {
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
|
|
if (options.offer_to_receive_audio == 0) {
|
|
RemoveRecvDirectionFromReceivingTransceiversOfType(
|
|
cricket::MEDIA_TYPE_AUDIO);
|
|
} else if (options.offer_to_receive_audio == 1) {
|
|
AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_AUDIO);
|
|
} else if (options.offer_to_receive_audio > 1) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
|
|
"offer_to_receive_audio > 1 is not supported.");
|
|
}
|
|
|
|
if (options.offer_to_receive_video == 0) {
|
|
RemoveRecvDirectionFromReceivingTransceiversOfType(
|
|
cricket::MEDIA_TYPE_VIDEO);
|
|
} else if (options.offer_to_receive_video == 1) {
|
|
AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
|
|
} else if (options.offer_to_receive_video > 1) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
|
|
"offer_to_receive_video > 1 is not supported.");
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
void PeerConnection::RemoveRecvDirectionFromReceivingTransceiversOfType(
|
|
cricket::MediaType media_type) {
|
|
for (const auto& transceiver : GetReceivingTransceiversOfType(media_type)) {
|
|
RtpTransceiverDirection new_direction =
|
|
RtpTransceiverDirectionWithRecvSet(transceiver->direction(), false);
|
|
if (new_direction != transceiver->direction()) {
|
|
RTC_LOG(LS_INFO) << "Changing " << cricket::MediaTypeToString(media_type)
|
|
<< " transceiver (MID="
|
|
<< transceiver->mid().value_or("<not set>") << ") from "
|
|
<< RtpTransceiverDirectionToString(
|
|
transceiver->direction())
|
|
<< " to "
|
|
<< RtpTransceiverDirectionToString(new_direction)
|
|
<< " since CreateOffer specified offer_to_receive=0";
|
|
transceiver->internal()->set_direction(new_direction);
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::AddUpToOneReceivingTransceiverOfType(
|
|
cricket::MediaType media_type) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (GetReceivingTransceiversOfType(media_type).empty()) {
|
|
RTC_LOG(LS_INFO)
|
|
<< "Adding one recvonly " << cricket::MediaTypeToString(media_type)
|
|
<< " transceiver since CreateOffer specified offer_to_receive=1";
|
|
RtpTransceiverInit init;
|
|
init.direction = RtpTransceiverDirection::kRecvOnly;
|
|
AddTransceiver(media_type, nullptr, init,
|
|
/*update_negotiation_needed=*/false);
|
|
}
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
|
|
PeerConnection::GetReceivingTransceiversOfType(cricket::MediaType media_type) {
|
|
std::vector<
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
|
|
receiving_transceivers;
|
|
for (const auto& transceiver : transceivers_) {
|
|
if (!transceiver->stopped() && transceiver->media_type() == media_type &&
|
|
RtpTransceiverDirectionHasRecv(transceiver->direction())) {
|
|
receiving_transceivers.push_back(transceiver);
|
|
}
|
|
}
|
|
return receiving_transceivers;
|
|
}
|
|
|
|
void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
|
|
const RTCOfferAnswerOptions& options) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
|
|
observer_refptr =
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserver>(observer),
|
|
options](std::function<void()> operations_chain_callback) {
|
|
// Abort early if |this_weak_ptr| is no longer valid.
|
|
if (!this_weak_ptr) {
|
|
observer_refptr->OnFailure(RTCError(
|
|
RTCErrorType::INTERNAL_ERROR,
|
|
"CreateAnswer failed because the session was shut down"));
|
|
operations_chain_callback();
|
|
return;
|
|
}
|
|
// The operation completes asynchronously when the wrapper is invoked.
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserverOperationWrapper>
|
|
observer_wrapper(new rtc::RefCountedObject<
|
|
CreateSessionDescriptionObserverOperationWrapper>(
|
|
std::move(observer_refptr),
|
|
std::move(operations_chain_callback)));
|
|
this_weak_ptr->DoCreateAnswer(options, observer_wrapper);
|
|
});
|
|
}
|
|
|
|
void PeerConnection::DoCreateAnswer(
|
|
const RTCOfferAnswerOptions& options,
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserver> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "PeerConnection::DoCreateAnswer");
|
|
if (!observer) {
|
|
RTC_LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
// If a session error has occurred the PeerConnection is in a possibly
|
|
// inconsistent state so fail right away.
|
|
if (session_error() != SessionError::kNone) {
|
|
std::string error_message = GetSessionErrorMsg();
|
|
RTC_LOG(LS_ERROR) << "CreateAnswer: " << error_message;
|
|
PostCreateSessionDescriptionFailure(
|
|
observer,
|
|
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
|
return;
|
|
}
|
|
|
|
if (!(signaling_state_ == kHaveRemoteOffer ||
|
|
signaling_state_ == kHaveLocalPrAnswer)) {
|
|
std::string error =
|
|
"PeerConnection cannot create an answer in a state other than "
|
|
"have-remote-offer or have-local-pranswer.";
|
|
RTC_LOG(LS_ERROR) << error;
|
|
PostCreateSessionDescriptionFailure(
|
|
observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
|
|
return;
|
|
}
|
|
|
|
// The remote description should be set if we're in the right state.
|
|
RTC_DCHECK(remote_description());
|
|
|
|
if (IsUnifiedPlan()) {
|
|
if (options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
|
|
RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_audio is not "
|
|
"supported with Unified Plan semantics. Use the "
|
|
"RtpTransceiver API instead.";
|
|
}
|
|
if (options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
|
|
RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_video is not "
|
|
"supported with Unified Plan semantics. Use the "
|
|
"RtpTransceiver API instead.";
|
|
}
|
|
}
|
|
|
|
cricket::MediaSessionOptions session_options;
|
|
GetOptionsForAnswer(options, &session_options);
|
|
|
|
webrtc_session_desc_factory_->CreateAnswer(observer, session_options);
|
|
}
|
|
|
|
void PeerConnection::SetLocalDescription(
|
|
SetSessionDescriptionObserver* observer,
|
|
SessionDescriptionInterface* desc_ptr) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
|
|
observer_refptr =
|
|
rtc::scoped_refptr<SetSessionDescriptionObserver>(observer),
|
|
desc = std::unique_ptr<SessionDescriptionInterface>(desc_ptr)](
|
|
std::function<void()> operations_chain_callback) mutable {
|
|
// Abort early if |this_weak_ptr| is no longer valid.
|
|
if (!this_weak_ptr) {
|
|
// For consistency with DoSetLocalDescription(), we DO NOT inform the
|
|
// |observer_refptr| that the operation failed in this case.
|
|
// TODO(hbos): If/when we process SLD messages in ~PeerConnection,
|
|
// the consistent thing would be to inform the observer here.
|
|
operations_chain_callback();
|
|
return;
|
|
}
|
|
this_weak_ptr->DoSetLocalDescription(std::move(desc),
|
|
std::move(observer_refptr));
|
|
// DoSetLocalDescription() is currently implemented as a synchronous
|
|
// operation but where the |observer|'s callbacks are invoked
|
|
// asynchronously in a post to OnMessage().
|
|
// For backwards-compatability reasons, we declare the operation as
|
|
// completed here (rather than in OnMessage()). This ensures that
|
|
// subsequent offer/answer operations can start immediately (without
|
|
// waiting for OnMessage()).
|
|
operations_chain_callback();
|
|
});
|
|
}
|
|
|
|
void PeerConnection::SetLocalDescription(
|
|
SetSessionDescriptionObserver* observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// The |create_sdp_observer| handles performing DoSetLocalDescription() with
|
|
// the resulting description as well as completing the operation.
|
|
rtc::scoped_refptr<ImplicitCreateSessionDescriptionObserver>
|
|
create_sdp_observer(
|
|
new rtc::RefCountedObject<ImplicitCreateSessionDescriptionObserver>(
|
|
weak_ptr_factory_.GetWeakPtr(),
|
|
rtc::scoped_refptr<SetSessionDescriptionObserver>(observer)));
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
|
|
create_sdp_observer](std::function<void()> operations_chain_callback) {
|
|
// The |create_sdp_observer| is responsible for completing the
|
|
// operation.
|
|
create_sdp_observer->SetOperationCompleteCallback(
|
|
std::move(operations_chain_callback));
|
|
// Abort early if |this_weak_ptr| is no longer valid. This triggers the
|
|
// same code path as if DoCreateOffer() or DoCreateAnswer() failed.
|
|
if (!this_weak_ptr) {
|
|
create_sdp_observer->OnFailure(RTCError(
|
|
RTCErrorType::INTERNAL_ERROR,
|
|
"SetLocalDescription failed because the session was shut down"));
|
|
return;
|
|
}
|
|
switch (this_weak_ptr->signaling_state()) {
|
|
case PeerConnectionInterface::kStable:
|
|
case PeerConnectionInterface::kHaveLocalOffer:
|
|
case PeerConnectionInterface::kHaveRemotePrAnswer:
|
|
// TODO(hbos): If [LastCreatedOffer] exists and still represents the
|
|
// current state of the system, use that instead of creating another
|
|
// offer.
|
|
this_weak_ptr->DoCreateOffer(RTCOfferAnswerOptions(),
|
|
create_sdp_observer);
|
|
break;
|
|
case PeerConnectionInterface::kHaveLocalPrAnswer:
|
|
case PeerConnectionInterface::kHaveRemoteOffer:
|
|
// TODO(hbos): If [LastCreatedAnswer] exists and still represents
|
|
// the current state of the system, use that instead of creating
|
|
// another answer.
|
|
this_weak_ptr->DoCreateAnswer(RTCOfferAnswerOptions(),
|
|
create_sdp_observer);
|
|
break;
|
|
case PeerConnectionInterface::kClosed:
|
|
create_sdp_observer->OnFailure(RTCError(
|
|
RTCErrorType::INVALID_STATE,
|
|
"SetLocalDescription called when PeerConnection is closed."));
|
|
break;
|
|
}
|
|
});
|
|
}
|
|
|
|
void PeerConnection::DoSetLocalDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc,
|
|
rtc::scoped_refptr<SetSessionDescriptionObserver> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "PeerConnection::DoSetLocalDescription");
|
|
|
|
if (!observer) {
|
|
RTC_LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
if (!desc) {
|
|
PostSetSessionDescriptionFailure(
|
|
observer,
|
|
RTCError(RTCErrorType::INTERNAL_ERROR, "SessionDescription is NULL."));
|
|
return;
|
|
}
|
|
|
|
// If a session error has occurred the PeerConnection is in a possibly
|
|
// inconsistent state so fail right away.
|
|
if (session_error() != SessionError::kNone) {
|
|
std::string error_message = GetSessionErrorMsg();
|
|
RTC_LOG(LS_ERROR) << "SetLocalDescription: " << error_message;
|
|
PostSetSessionDescriptionFailure(
|
|
observer,
|
|
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
|
return;
|
|
}
|
|
|
|
// For SLD we support only explicit rollback.
|
|
if (desc->GetType() == SdpType::kRollback) {
|
|
if (IsUnifiedPlan()) {
|
|
RTCError error = Rollback(desc->GetType());
|
|
if (error.ok()) {
|
|
PostSetSessionDescriptionSuccess(observer);
|
|
} else {
|
|
PostSetSessionDescriptionFailure(observer, std::move(error));
|
|
}
|
|
} else {
|
|
PostSetSessionDescriptionFailure(
|
|
observer, RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
|
|
"Rollback not supported in Plan B"));
|
|
}
|
|
return;
|
|
}
|
|
|
|
RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_LOCAL);
|
|
if (!error.ok()) {
|
|
std::string error_message = GetSetDescriptionErrorMessage(
|
|
cricket::CS_LOCAL, desc->GetType(), error);
|
|
RTC_LOG(LS_ERROR) << error_message;
|
|
PostSetSessionDescriptionFailure(
|
|
observer,
|
|
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
|
return;
|
|
}
|
|
|
|
// Grab the description type before moving ownership to ApplyLocalDescription,
|
|
// which may destroy it before returning.
|
|
const SdpType type = desc->GetType();
|
|
|
|
error = ApplyLocalDescription(std::move(desc));
|
|
// |desc| may be destroyed at this point.
|
|
|
|
if (!error.ok()) {
|
|
// If ApplyLocalDescription fails, the PeerConnection could be in an
|
|
// inconsistent state, so act conservatively here and set the session error
|
|
// so that future calls to SetLocalDescription/SetRemoteDescription fail.
|
|
SetSessionError(SessionError::kContent, error.message());
|
|
std::string error_message =
|
|
GetSetDescriptionErrorMessage(cricket::CS_LOCAL, type, error);
|
|
RTC_LOG(LS_ERROR) << error_message;
|
|
PostSetSessionDescriptionFailure(
|
|
observer,
|
|
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
|
return;
|
|
}
|
|
RTC_DCHECK(local_description());
|
|
|
|
PostSetSessionDescriptionSuccess(observer);
|
|
|
|
// 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()));
|
|
// Make UMA notes about what was agreed to.
|
|
ReportNegotiatedSdpSemantics(*local_description());
|
|
}
|
|
|
|
if (IsUnifiedPlan()) {
|
|
bool was_negotiation_needed = is_negotiation_needed_;
|
|
UpdateNegotiationNeeded();
|
|
if (signaling_state() == kStable && was_negotiation_needed &&
|
|
is_negotiation_needed_) {
|
|
Observer()->OnRenegotiationNeeded();
|
|
}
|
|
}
|
|
|
|
NoteUsageEvent(UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED);
|
|
}
|
|
|
|
RTCError PeerConnection::ApplyLocalDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(desc);
|
|
|
|
// 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);
|
|
|
|
// Take a reference to the old local description since it's used below to
|
|
// compare against the new local description. When setting the new local
|
|
// description, grab ownership of the replaced session description in case it
|
|
// is the same as |old_local_description|, to keep it alive for the duration
|
|
// of the method.
|
|
const SessionDescriptionInterface* old_local_description =
|
|
local_description();
|
|
std::unique_ptr<SessionDescriptionInterface> replaced_local_description;
|
|
SdpType type = desc->GetType();
|
|
if (type == SdpType::kAnswer) {
|
|
replaced_local_description = pending_local_description_
|
|
? std::move(pending_local_description_)
|
|
: std::move(current_local_description_);
|
|
current_local_description_ = std::move(desc);
|
|
pending_local_description_ = nullptr;
|
|
current_remote_description_ = std::move(pending_remote_description_);
|
|
} else {
|
|
replaced_local_description = std::move(pending_local_description_);
|
|
pending_local_description_ = std::move(desc);
|
|
}
|
|
// The session description to apply now must be accessed by
|
|
// |local_description()|.
|
|
RTC_DCHECK(local_description());
|
|
|
|
// Report statistics about any use of simulcast.
|
|
ReportSimulcastApiVersion(kSimulcastVersionApplyLocalDescription,
|
|
*local_description()->description());
|
|
|
|
if (!is_caller_) {
|
|
if (remote_description()) {
|
|
// Remote description was applied first, so this PC is the callee.
|
|
is_caller_ = false;
|
|
} else {
|
|
// Local description is applied first, so this PC is the caller.
|
|
is_caller_ = true;
|
|
}
|
|
}
|
|
|
|
RTCError error = PushdownTransportDescription(cricket::CS_LOCAL, type);
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
|
|
if (IsUnifiedPlan()) {
|
|
RTCError error = UpdateTransceiversAndDataChannels(
|
|
cricket::CS_LOCAL, *local_description(), old_local_description,
|
|
remote_description());
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
|
|
for (const auto& transceiver : transceivers_) {
|
|
// 2.2.7.1.1.(6-9): Set sender and receiver's transport slots.
|
|
// Note that code paths that don't set MID won't be able to use
|
|
// information about DTLS transports.
|
|
if (transceiver->mid()) {
|
|
auto dtls_transport =
|
|
LookupDtlsTransportByMidInternal(*transceiver->mid());
|
|
transceiver->internal()->sender_internal()->set_transport(
|
|
dtls_transport);
|
|
transceiver->internal()->receiver_internal()->set_transport(
|
|
dtls_transport);
|
|
}
|
|
|
|
const ContentInfo* content =
|
|
FindMediaSectionForTransceiver(transceiver, local_description());
|
|
if (!content) {
|
|
continue;
|
|
}
|
|
const MediaContentDescription* media_desc = content->media_description();
|
|
// 2.2.7.1.6: If description is of type "answer" or "pranswer", then run
|
|
// the following steps:
|
|
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
|
|
// 2.2.7.1.6.1: If direction is "sendonly" or "inactive", and
|
|
// transceiver's [[FiredDirection]] slot is either "sendrecv" or
|
|
// "recvonly", process the removal of a remote track for the media
|
|
// description, given transceiver, removeList, and muteTracks.
|
|
if (!RtpTransceiverDirectionHasRecv(media_desc->direction()) &&
|
|
(transceiver->internal()->fired_direction() &&
|
|
RtpTransceiverDirectionHasRecv(
|
|
*transceiver->internal()->fired_direction()))) {
|
|
ProcessRemovalOfRemoteTrack(transceiver, &remove_list,
|
|
&removed_streams);
|
|
}
|
|
// 2.2.7.1.6.2: Set transceiver's [[CurrentDirection]] and
|
|
// [[FiredDirection]] slots to direction.
|
|
transceiver->internal()->set_current_direction(media_desc->direction());
|
|
transceiver->internal()->set_fired_direction(media_desc->direction());
|
|
}
|
|
}
|
|
auto observer = Observer();
|
|
for (const auto& transceiver : remove_list) {
|
|
observer->OnRemoveTrack(transceiver->receiver());
|
|
}
|
|
for (const auto& stream : removed_streams) {
|
|
observer->OnRemoveStream(stream);
|
|
}
|
|
} else {
|
|
// Media channels will be created only when offer is set. These may use new
|
|
// transports just created by PushdownTransportDescription.
|
|
if (type == SdpType::kOffer) {
|
|
// TODO(bugs.webrtc.org/4676) - 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,
|
|
local_description()->description());
|
|
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 (IsSctpLike(data_channel_type()) && GetSctpSslRole(&role)) {
|
|
data_channel_controller_.AllocateSctpSids(role);
|
|
}
|
|
|
|
if (IsUnifiedPlan()) {
|
|
for (const auto& transceiver : transceivers_) {
|
|
const ContentInfo* content =
|
|
FindMediaSectionForTransceiver(transceiver, local_description());
|
|
if (!content) {
|
|
continue;
|
|
}
|
|
cricket::ChannelInterface* channel = transceiver->internal()->channel();
|
|
if (content->rejected || !channel || channel->local_streams().empty()) {
|
|
// 0 is a special value meaning "this sender has no associated send
|
|
// stream". Need to call this so the sender won't attempt to configure
|
|
// a no longer existing stream and run into DCHECKs in the lower
|
|
// layers.
|
|
transceiver->internal()->sender_internal()->SetSsrc(0);
|
|
} else {
|
|
// Get the StreamParams from the channel which could generate SSRCs.
|
|
const std::vector<StreamParams>& streams = channel->local_streams();
|
|
transceiver->internal()->sender_internal()->set_stream_ids(
|
|
streams[0].stream_ids());
|
|
transceiver->internal()->sender_internal()->SetSsrc(
|
|
streams[0].first_ssrc());
|
|
}
|
|
}
|
|
} else {
|
|
// Plan B semantics.
|
|
|
|
// 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::RtpDataContentDescription* rtp_data_desc =
|
|
data_content->media_description()->as_rtp_data();
|
|
// rtp_data_desc will be null if this is an SCTP description.
|
|
if (rtp_data_desc) {
|
|
data_channel_controller_.UpdateLocalRtpDataChannels(
|
|
rtp_data_desc->streams());
|
|
}
|
|
}
|
|
|
|
if (type == SdpType::kAnswer &&
|
|
local_ice_credentials_to_replace_->SatisfiesIceRestart(
|
|
*current_local_description_)) {
|
|
local_ice_credentials_to_replace_->ClearIceCredentials();
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
// The SDP parser used to populate these values by default for the 'content
|
|
// name' if an a=mid line was absent.
|
|
static absl::string_view GetDefaultMidForPlanB(cricket::MediaType media_type) {
|
|
switch (media_type) {
|
|
case cricket::MEDIA_TYPE_AUDIO:
|
|
return cricket::CN_AUDIO;
|
|
case cricket::MEDIA_TYPE_VIDEO:
|
|
return cricket::CN_VIDEO;
|
|
case cricket::MEDIA_TYPE_DATA:
|
|
return cricket::CN_DATA;
|
|
}
|
|
RTC_NOTREACHED();
|
|
return "";
|
|
}
|
|
|
|
void PeerConnection::FillInMissingRemoteMids(
|
|
cricket::SessionDescription* new_remote_description) {
|
|
RTC_DCHECK(new_remote_description);
|
|
const cricket::ContentInfos no_infos;
|
|
const cricket::ContentInfos& local_contents =
|
|
(local_description() ? local_description()->description()->contents()
|
|
: no_infos);
|
|
const cricket::ContentInfos& remote_contents =
|
|
(remote_description() ? remote_description()->description()->contents()
|
|
: no_infos);
|
|
for (size_t i = 0; i < new_remote_description->contents().size(); ++i) {
|
|
cricket::ContentInfo& content = new_remote_description->contents()[i];
|
|
if (!content.name.empty()) {
|
|
continue;
|
|
}
|
|
std::string new_mid;
|
|
absl::string_view source_explanation;
|
|
if (IsUnifiedPlan()) {
|
|
if (i < local_contents.size()) {
|
|
new_mid = local_contents[i].name;
|
|
source_explanation = "from the matching local media section";
|
|
} else if (i < remote_contents.size()) {
|
|
new_mid = remote_contents[i].name;
|
|
source_explanation = "from the matching previous remote media section";
|
|
} else {
|
|
new_mid = mid_generator_();
|
|
source_explanation = "generated just now";
|
|
}
|
|
} else {
|
|
new_mid = std::string(
|
|
GetDefaultMidForPlanB(content.media_description()->type()));
|
|
source_explanation = "to match pre-existing behavior";
|
|
}
|
|
RTC_DCHECK(!new_mid.empty());
|
|
content.name = new_mid;
|
|
new_remote_description->transport_infos()[i].content_name = new_mid;
|
|
RTC_LOG(LS_INFO) << "SetRemoteDescription: Remote media section at i=" << i
|
|
<< " is missing an a=mid line. Filling in the value '"
|
|
<< new_mid << "' " << source_explanation << ".";
|
|
}
|
|
}
|
|
|
|
void PeerConnection::SetRemoteDescription(
|
|
SetSessionDescriptionObserver* observer,
|
|
SessionDescriptionInterface* desc_ptr) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
|
|
observer_refptr =
|
|
rtc::scoped_refptr<SetSessionDescriptionObserver>(observer),
|
|
desc = std::unique_ptr<SessionDescriptionInterface>(desc_ptr)](
|
|
std::function<void()> operations_chain_callback) mutable {
|
|
// Abort early if |this_weak_ptr| is no longer valid.
|
|
if (!this_weak_ptr) {
|
|
// For consistency with SetRemoteDescriptionObserverAdapter, we DO NOT
|
|
// inform the |observer_refptr| that the operation failed in this
|
|
// case.
|
|
// TODO(hbos): If/when we process SRD messages in ~PeerConnection,
|
|
// the consistent thing would be to inform the observer here.
|
|
operations_chain_callback();
|
|
return;
|
|
}
|
|
this_weak_ptr->DoSetRemoteDescription(
|
|
std::move(desc),
|
|
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>(
|
|
new SetRemoteDescriptionObserverAdapter(
|
|
this_weak_ptr.get(), std::move(observer_refptr))));
|
|
// DoSetRemoteDescription() is currently implemented as a synchronous
|
|
// operation but where SetRemoteDescriptionObserverAdapter ensures that
|
|
// the |observer|'s callbacks are invoked asynchronously in a post to
|
|
// OnMessage().
|
|
// For backwards-compatability reasons, we declare the operation as
|
|
// completed here (rather than in OnMessage()). This ensures that
|
|
// subsequent offer/answer operations can start immediately (without
|
|
// waiting for OnMessage()).
|
|
operations_chain_callback();
|
|
});
|
|
}
|
|
|
|
void PeerConnection::SetRemoteDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc,
|
|
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(), observer,
|
|
desc = std::move(desc)](
|
|
std::function<void()> operations_chain_callback) mutable {
|
|
// Abort early if |this_weak_ptr| is no longer valid.
|
|
if (!this_weak_ptr) {
|
|
// For consistency with DoSetRemoteDescription(), we DO inform the
|
|
// |observer| that the operation failed in this case.
|
|
observer->OnSetRemoteDescriptionComplete(RTCError(
|
|
RTCErrorType::INVALID_STATE,
|
|
"Failed to set remote offer sdp: failed because the session was "
|
|
"shut down"));
|
|
operations_chain_callback();
|
|
return;
|
|
}
|
|
this_weak_ptr->DoSetRemoteDescription(std::move(desc),
|
|
std::move(observer));
|
|
// DoSetRemoteDescription() is currently implemented as a synchronous
|
|
// operation. The |observer| will already have been informed that it
|
|
// completed, and we can mark this operation as complete without any
|
|
// loose ends.
|
|
operations_chain_callback();
|
|
});
|
|
}
|
|
|
|
void PeerConnection::DoSetRemoteDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc,
|
|
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "PeerConnection::DoSetRemoteDescription");
|
|
|
|
if (!observer) {
|
|
RTC_LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
if (!desc) {
|
|
observer->OnSetRemoteDescriptionComplete(RTCError(
|
|
RTCErrorType::INVALID_PARAMETER, "SessionDescription is NULL."));
|
|
return;
|
|
}
|
|
|
|
// If a session error has occurred the PeerConnection is in a possibly
|
|
// inconsistent state so fail right away.
|
|
if (session_error() != SessionError::kNone) {
|
|
std::string error_message = GetSessionErrorMsg();
|
|
RTC_LOG(LS_ERROR) << "SetRemoteDescription: " << error_message;
|
|
observer->OnSetRemoteDescriptionComplete(
|
|
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
|
return;
|
|
}
|
|
if (IsUnifiedPlan()) {
|
|
if (configuration_.enable_implicit_rollback) {
|
|
if (desc->GetType() == SdpType::kOffer &&
|
|
signaling_state() == kHaveLocalOffer) {
|
|
Rollback(desc->GetType());
|
|
}
|
|
}
|
|
// Explicit rollback.
|
|
if (desc->GetType() == SdpType::kRollback) {
|
|
observer->OnSetRemoteDescriptionComplete(Rollback(desc->GetType()));
|
|
return;
|
|
}
|
|
} else if (desc->GetType() == SdpType::kRollback) {
|
|
observer->OnSetRemoteDescriptionComplete(
|
|
RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
|
|
"Rollback not supported in Plan B"));
|
|
return;
|
|
}
|
|
if (desc->GetType() == SdpType::kOffer) {
|
|
// Report to UMA the format of the received offer.
|
|
ReportSdpFormatReceived(*desc);
|
|
}
|
|
|
|
// Handle remote descriptions missing a=mid lines for interop with legacy end
|
|
// points.
|
|
FillInMissingRemoteMids(desc->description());
|
|
|
|
RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_REMOTE);
|
|
if (!error.ok()) {
|
|
std::string error_message = GetSetDescriptionErrorMessage(
|
|
cricket::CS_REMOTE, desc->GetType(), error);
|
|
RTC_LOG(LS_ERROR) << error_message;
|
|
observer->OnSetRemoteDescriptionComplete(
|
|
RTCError(error.type(), std::move(error_message)));
|
|
return;
|
|
}
|
|
|
|
// Grab the description type before moving ownership to
|
|
// ApplyRemoteDescription, which may destroy it before returning.
|
|
const SdpType type = desc->GetType();
|
|
|
|
error = ApplyRemoteDescription(std::move(desc));
|
|
// |desc| may be destroyed at this point.
|
|
|
|
if (!error.ok()) {
|
|
// If ApplyRemoteDescription fails, the PeerConnection could be in an
|
|
// inconsistent state, so act conservatively here and set the session error
|
|
// so that future calls to SetLocalDescription/SetRemoteDescription fail.
|
|
SetSessionError(SessionError::kContent, error.message());
|
|
std::string error_message =
|
|
GetSetDescriptionErrorMessage(cricket::CS_REMOTE, type, error);
|
|
RTC_LOG(LS_ERROR) << error_message;
|
|
observer->OnSetRemoteDescriptionComplete(
|
|
RTCError(error.type(), std::move(error_message)));
|
|
return;
|
|
}
|
|
RTC_DCHECK(remote_description());
|
|
|
|
if (type == 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()));
|
|
// Make UMA notes about what was agreed to.
|
|
ReportNegotiatedSdpSemantics(*remote_description());
|
|
}
|
|
|
|
if (IsUnifiedPlan()) {
|
|
bool was_negotiation_needed = is_negotiation_needed_;
|
|
UpdateNegotiationNeeded();
|
|
if (signaling_state() == kStable && was_negotiation_needed &&
|
|
is_negotiation_needed_) {
|
|
Observer()->OnRenegotiationNeeded();
|
|
}
|
|
}
|
|
|
|
observer->OnSetRemoteDescriptionComplete(RTCError::OK());
|
|
NoteUsageEvent(UsageEvent::SET_REMOTE_DESCRIPTION_SUCCEEDED);
|
|
}
|
|
|
|
RTCError PeerConnection::ApplyRemoteDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(desc);
|
|
|
|
// 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);
|
|
|
|
// Take a reference to the old remote description since it's used below to
|
|
// compare against the new remote description. When setting the new remote
|
|
// description, grab ownership of the replaced session description in case it
|
|
// is the same as |old_remote_description|, to keep it alive for the duration
|
|
// of the method.
|
|
const SessionDescriptionInterface* old_remote_description =
|
|
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());
|
|
|
|
// Report statistics about any use of simulcast.
|
|
ReportSimulcastApiVersion(kSimulcastVersionApplyRemoteDescription,
|
|
*remote_description()->description());
|
|
|
|
RTCError error = PushdownTransportDescription(cricket::CS_REMOTE, type);
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
// Transport and Media channels will be created only when offer is set.
|
|
if (IsUnifiedPlan()) {
|
|
RTCError error = UpdateTransceiversAndDataChannels(
|
|
cricket::CS_REMOTE, *remote_description(), local_description(),
|
|
old_remote_description);
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
} else {
|
|
// Media channels will be created only when offer is set. These may use new
|
|
// transports just created by PushdownTransportDescription.
|
|
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,
|
|
remote_description()->description());
|
|
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 &&
|
|
remote_description()->number_of_mediasections() > 0u &&
|
|
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 (IsSctpLike(data_channel_type()) && GetSctpSslRole(&role)) {
|
|
data_channel_controller_.AllocateSctpSids(role);
|
|
}
|
|
|
|
if (IsUnifiedPlan()) {
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
|
|
now_receiving_transceivers;
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> added_streams;
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
|
|
for (const auto& transceiver : transceivers_) {
|
|
const ContentInfo* content =
|
|
FindMediaSectionForTransceiver(transceiver, remote_description());
|
|
if (!content) {
|
|
continue;
|
|
}
|
|
const MediaContentDescription* media_desc = content->media_description();
|
|
RtpTransceiverDirection local_direction =
|
|
RtpTransceiverDirectionReversed(media_desc->direction());
|
|
// Roughly the same as steps 2.2.8.6 of section 4.4.1.6 "Set the
|
|
// RTCSessionDescription: Set the associated remote streams given
|
|
// transceiver.[[Receiver]], msids, addList, and removeList".
|
|
// https://w3c.github.io/webrtc-pc/#set-the-rtcsessiondescription
|
|
if (RtpTransceiverDirectionHasRecv(local_direction)) {
|
|
std::vector<std::string> stream_ids;
|
|
if (!media_desc->streams().empty()) {
|
|
// The remote description has signaled the stream IDs.
|
|
stream_ids = media_desc->streams()[0].stream_ids();
|
|
}
|
|
transceiver_stable_states_by_transceivers_[transceiver]
|
|
.SetRemoteStreamIdsIfUnset(transceiver->receiver()->stream_ids());
|
|
|
|
RTC_LOG(LS_INFO) << "Processing the MSIDs for MID=" << content->name
|
|
<< " (" << GetStreamIdsString(stream_ids) << ").";
|
|
SetAssociatedRemoteStreams(transceiver->internal()->receiver_internal(),
|
|
stream_ids, &added_streams,
|
|
&removed_streams);
|
|
// From the WebRTC specification, steps 2.2.8.5/6 of section 4.4.1.6
|
|
// "Set the RTCSessionDescription: If direction is sendrecv or recvonly,
|
|
// and transceiver's current direction is neither sendrecv nor recvonly,
|
|
// process the addition of a remote track for the media description.
|
|
if (!transceiver->fired_direction() ||
|
|
!RtpTransceiverDirectionHasRecv(*transceiver->fired_direction())) {
|
|
RTC_LOG(LS_INFO)
|
|
<< "Processing the addition of a remote track for MID="
|
|
<< content->name << ".";
|
|
now_receiving_transceivers.push_back(transceiver);
|
|
}
|
|
}
|
|
// 2.2.8.1.9: If direction is "sendonly" or "inactive", and transceiver's
|
|
// [[FiredDirection]] slot is either "sendrecv" or "recvonly", process the
|
|
// removal of a remote track for the media description, given transceiver,
|
|
// removeList, and muteTracks.
|
|
if (!RtpTransceiverDirectionHasRecv(local_direction) &&
|
|
(transceiver->fired_direction() &&
|
|
RtpTransceiverDirectionHasRecv(*transceiver->fired_direction()))) {
|
|
ProcessRemovalOfRemoteTrack(transceiver, &remove_list,
|
|
&removed_streams);
|
|
}
|
|
// 2.2.8.1.10: Set transceiver's [[FiredDirection]] slot to direction.
|
|
transceiver->internal()->set_fired_direction(local_direction);
|
|
// 2.2.8.1.11: If description is of type "answer" or "pranswer", then run
|
|
// the following steps:
|
|
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
|
|
// 2.2.8.1.11.1: Set transceiver's [[CurrentDirection]] slot to
|
|
// direction.
|
|
transceiver->internal()->set_current_direction(local_direction);
|
|
// 2.2.8.1.11.[3-6]: Set the transport internal slots.
|
|
if (transceiver->mid()) {
|
|
auto dtls_transport =
|
|
LookupDtlsTransportByMidInternal(*transceiver->mid());
|
|
transceiver->internal()->sender_internal()->set_transport(
|
|
dtls_transport);
|
|
transceiver->internal()->receiver_internal()->set_transport(
|
|
dtls_transport);
|
|
}
|
|
}
|
|
// 2.2.8.1.12: If the media description is rejected, and transceiver is
|
|
// not already stopped, stop the RTCRtpTransceiver transceiver.
|
|
if (content->rejected && !transceiver->stopped()) {
|
|
RTC_LOG(LS_INFO) << "Stopping transceiver for MID=" << content->name
|
|
<< " since the media section was rejected.";
|
|
transceiver->Stop();
|
|
}
|
|
if (!content->rejected &&
|
|
RtpTransceiverDirectionHasRecv(local_direction)) {
|
|
if (!media_desc->streams().empty() &&
|
|
media_desc->streams()[0].has_ssrcs()) {
|
|
uint32_t ssrc = media_desc->streams()[0].first_ssrc();
|
|
transceiver->internal()->receiver_internal()->SetupMediaChannel(ssrc);
|
|
} else {
|
|
transceiver->internal()
|
|
->receiver_internal()
|
|
->SetupUnsignaledMediaChannel();
|
|
}
|
|
}
|
|
}
|
|
// Once all processing has finished, fire off callbacks.
|
|
auto observer = Observer();
|
|
for (const auto& transceiver : now_receiving_transceivers) {
|
|
stats_->AddTrack(transceiver->receiver()->track());
|
|
observer->OnTrack(transceiver);
|
|
observer->OnAddTrack(transceiver->receiver(),
|
|
transceiver->receiver()->streams());
|
|
}
|
|
for (const auto& stream : added_streams) {
|
|
observer->OnAddStream(stream);
|
|
}
|
|
for (const auto& transceiver : remove_list) {
|
|
observer->OnRemoveTrack(transceiver->receiver());
|
|
}
|
|
for (const auto& stream : removed_streams) {
|
|
observer->OnRemoveStream(stream);
|
|
}
|
|
}
|
|
|
|
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::RtpDataContentDescription* rtp_data_desc =
|
|
GetFirstRtpDataContentDescription(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());
|
|
|
|
if (!IsUnifiedPlan()) {
|
|
// 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);
|
|
}
|
|
}
|
|
|
|
// If this is an RTP data transport, update the DataChannels with the
|
|
// information from the remote peer.
|
|
if (rtp_data_desc) {
|
|
data_channel_controller_.UpdateRemoteRtpDataChannels(
|
|
GetActiveStreams(rtp_data_desc));
|
|
}
|
|
|
|
// Iterate new_streams and notify the observer about new MediaStreams.
|
|
auto observer = Observer();
|
|
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();
|
|
}
|
|
|
|
if (type == SdpType::kAnswer &&
|
|
local_ice_credentials_to_replace_->SatisfiesIceRestart(
|
|
*current_local_description_)) {
|
|
local_ice_credentials_to_replace_->ClearIceCredentials();
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
void PeerConnection::SetAssociatedRemoteStreams(
|
|
rtc::scoped_refptr<RtpReceiverInternal> receiver,
|
|
const std::vector<std::string>& stream_ids,
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* added_streams,
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) {
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> media_streams;
|
|
for (const std::string& stream_id : stream_ids) {
|
|
rtc::scoped_refptr<MediaStreamInterface> stream =
|
|
remote_streams_->find(stream_id);
|
|
if (!stream) {
|
|
stream = MediaStreamProxy::Create(rtc::Thread::Current(),
|
|
MediaStream::Create(stream_id));
|
|
remote_streams_->AddStream(stream);
|
|
added_streams->push_back(stream);
|
|
}
|
|
media_streams.push_back(stream);
|
|
}
|
|
// Special case: "a=msid" missing, use random stream ID.
|
|
if (media_streams.empty() &&
|
|
!(remote_description()->description()->msid_signaling() &
|
|
cricket::kMsidSignalingMediaSection)) {
|
|
if (!missing_msid_default_stream_) {
|
|
missing_msid_default_stream_ = MediaStreamProxy::Create(
|
|
rtc::Thread::Current(), MediaStream::Create(rtc::CreateRandomUuid()));
|
|
added_streams->push_back(missing_msid_default_stream_);
|
|
}
|
|
media_streams.push_back(missing_msid_default_stream_);
|
|
}
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> previous_streams =
|
|
receiver->streams();
|
|
// SetStreams() will add/remove the receiver's track to/from the streams. This
|
|
// differs from the spec - the spec uses an "addList" and "removeList" to
|
|
// update the stream-track relationships in a later step. We do this earlier,
|
|
// changing the order of things, but the end-result is the same.
|
|
// TODO(hbos): When we remove remote_streams(), use set_stream_ids()
|
|
// instead. https://crbug.com/webrtc/9480
|
|
receiver->SetStreams(media_streams);
|
|
RemoveRemoteStreamsIfEmpty(previous_streams, removed_streams);
|
|
}
|
|
|
|
void PeerConnection::ProcessRemovalOfRemoteTrack(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver,
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>* remove_list,
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) {
|
|
RTC_DCHECK(transceiver->mid());
|
|
RTC_LOG(LS_INFO) << "Processing the removal of a track for MID="
|
|
<< *transceiver->mid();
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> previous_streams =
|
|
transceiver->internal()->receiver_internal()->streams();
|
|
// This will remove the remote track from the streams.
|
|
transceiver->internal()->receiver_internal()->set_stream_ids({});
|
|
remove_list->push_back(transceiver);
|
|
RemoveRemoteStreamsIfEmpty(previous_streams, removed_streams);
|
|
}
|
|
|
|
void PeerConnection::RemoveRemoteStreamsIfEmpty(
|
|
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& remote_streams,
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) {
|
|
// TODO(https://crbug.com/webrtc/9480): When we use stream IDs instead of
|
|
// streams, see if the stream was removed by checking if this was the last
|
|
// receiver with that stream ID.
|
|
for (const auto& remote_stream : remote_streams) {
|
|
if (remote_stream->GetAudioTracks().empty() &&
|
|
remote_stream->GetVideoTracks().empty()) {
|
|
remote_streams_->RemoveStream(remote_stream);
|
|
removed_streams->push_back(remote_stream);
|
|
}
|
|
}
|
|
}
|
|
|
|
RTCError PeerConnection::UpdateTransceiversAndDataChannels(
|
|
cricket::ContentSource source,
|
|
const SessionDescriptionInterface& new_session,
|
|
const SessionDescriptionInterface* old_local_description,
|
|
const SessionDescriptionInterface* old_remote_description) {
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
|
|
const cricket::ContentGroup* bundle_group = nullptr;
|
|
if (new_session.GetType() == SdpType::kOffer) {
|
|
auto bundle_group_or_error =
|
|
GetEarlyBundleGroup(*new_session.description());
|
|
if (!bundle_group_or_error.ok()) {
|
|
return bundle_group_or_error.MoveError();
|
|
}
|
|
bundle_group = bundle_group_or_error.MoveValue();
|
|
}
|
|
|
|
const ContentInfos& new_contents = new_session.description()->contents();
|
|
for (size_t i = 0; i < new_contents.size(); ++i) {
|
|
const cricket::ContentInfo& new_content = new_contents[i];
|
|
cricket::MediaType media_type = new_content.media_description()->type();
|
|
mid_generator_.AddKnownId(new_content.name);
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO) {
|
|
const cricket::ContentInfo* old_local_content = nullptr;
|
|
if (old_local_description &&
|
|
i < old_local_description->description()->contents().size()) {
|
|
old_local_content =
|
|
&old_local_description->description()->contents()[i];
|
|
}
|
|
const cricket::ContentInfo* old_remote_content = nullptr;
|
|
if (old_remote_description &&
|
|
i < old_remote_description->description()->contents().size()) {
|
|
old_remote_content =
|
|
&old_remote_description->description()->contents()[i];
|
|
}
|
|
auto transceiver_or_error =
|
|
AssociateTransceiver(source, new_session.GetType(), i, new_content,
|
|
old_local_content, old_remote_content);
|
|
if (!transceiver_or_error.ok()) {
|
|
return transceiver_or_error.MoveError();
|
|
}
|
|
auto transceiver = transceiver_or_error.MoveValue();
|
|
RTCError error =
|
|
UpdateTransceiverChannel(transceiver, new_content, bundle_group);
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
} else if (media_type == cricket::MEDIA_TYPE_DATA) {
|
|
if (GetDataMid() && new_content.name != *GetDataMid()) {
|
|
// Ignore all but the first data section.
|
|
RTC_LOG(LS_INFO) << "Ignoring data media section with MID="
|
|
<< new_content.name;
|
|
continue;
|
|
}
|
|
RTCError error = UpdateDataChannel(source, new_content, bundle_group);
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
} else {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Unknown section type.");
|
|
}
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
RTCError PeerConnection::UpdateTransceiverChannel(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver,
|
|
const cricket::ContentInfo& content,
|
|
const cricket::ContentGroup* bundle_group) {
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
RTC_DCHECK(transceiver);
|
|
cricket::ChannelInterface* channel = transceiver->internal()->channel();
|
|
if (content.rejected) {
|
|
if (channel) {
|
|
transceiver->internal()->SetChannel(nullptr);
|
|
DestroyChannelInterface(channel);
|
|
}
|
|
} else {
|
|
if (!channel) {
|
|
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
|
channel = CreateVoiceChannel(content.name);
|
|
} else {
|
|
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->media_type());
|
|
channel = CreateVideoChannel(content.name);
|
|
}
|
|
if (!channel) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::INTERNAL_ERROR,
|
|
"Failed to create channel for mid=" + content.name);
|
|
}
|
|
transceiver->internal()->SetChannel(channel);
|
|
}
|
|
}
|
|
return RTCError::OK();
|
|
}
|
|
|
|
RTCError PeerConnection::UpdateDataChannel(
|
|
cricket::ContentSource source,
|
|
const cricket::ContentInfo& content,
|
|
const cricket::ContentGroup* bundle_group) {
|
|
if (data_channel_type() == cricket::DCT_NONE) {
|
|
// If data channels are disabled, ignore this media section. CreateAnswer
|
|
// will take care of rejecting it.
|
|
return RTCError::OK();
|
|
}
|
|
if (content.rejected) {
|
|
RTC_LOG(LS_INFO) << "Rejected data channel, mid=" << content.mid();
|
|
DestroyDataChannelTransport();
|
|
} else {
|
|
if (!data_channel_controller_.rtp_data_channel() &&
|
|
!data_channel_controller_.data_channel_transport()) {
|
|
RTC_LOG(LS_INFO) << "Creating data channel, mid=" << content.mid();
|
|
if (!CreateDataChannel(content.name)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Failed to create data channel.");
|
|
}
|
|
}
|
|
if (source == cricket::CS_REMOTE) {
|
|
const MediaContentDescription* data_desc = content.media_description();
|
|
if (data_desc && cricket::IsRtpProtocol(data_desc->protocol())) {
|
|
data_channel_controller_.UpdateRemoteRtpDataChannels(
|
|
GetActiveStreams(data_desc));
|
|
}
|
|
}
|
|
}
|
|
return RTCError::OK();
|
|
}
|
|
|
|
// This method will extract any send encodings that were sent by the remote
|
|
// connection. This is currently only relevant for Simulcast scenario (where
|
|
// the number of layers may be communicated by the server).
|
|
static std::vector<RtpEncodingParameters> GetSendEncodingsFromRemoteDescription(
|
|
const MediaContentDescription& desc) {
|
|
if (!desc.HasSimulcast()) {
|
|
return {};
|
|
}
|
|
std::vector<RtpEncodingParameters> result;
|
|
const SimulcastDescription& simulcast = desc.simulcast_description();
|
|
|
|
// This is a remote description, the parameters we are after should appear
|
|
// as receive streams.
|
|
for (const auto& alternatives : simulcast.receive_layers()) {
|
|
RTC_DCHECK(!alternatives.empty());
|
|
// There is currently no way to specify or choose from alternatives.
|
|
// We will always use the first alternative, which is the most preferred.
|
|
const SimulcastLayer& layer = alternatives[0];
|
|
RtpEncodingParameters parameters;
|
|
parameters.rid = layer.rid;
|
|
parameters.active = !layer.is_paused;
|
|
result.push_back(parameters);
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
static RTCError UpdateSimulcastLayerStatusInSender(
|
|
const std::vector<SimulcastLayer>& layers,
|
|
rtc::scoped_refptr<RtpSenderInternal> sender) {
|
|
RTC_DCHECK(sender);
|
|
RtpParameters parameters = sender->GetParametersInternal();
|
|
std::vector<std::string> disabled_layers;
|
|
|
|
// The simulcast envelope cannot be changed, only the status of the streams.
|
|
// So we will iterate over the send encodings rather than the layers.
|
|
for (RtpEncodingParameters& encoding : parameters.encodings) {
|
|
auto iter = std::find_if(layers.begin(), layers.end(),
|
|
[&encoding](const SimulcastLayer& layer) {
|
|
return layer.rid == encoding.rid;
|
|
});
|
|
// A layer that cannot be found may have been removed by the remote party.
|
|
if (iter == layers.end()) {
|
|
disabled_layers.push_back(encoding.rid);
|
|
continue;
|
|
}
|
|
|
|
encoding.active = !iter->is_paused;
|
|
}
|
|
|
|
RTCError result = sender->SetParametersInternal(parameters);
|
|
if (result.ok()) {
|
|
result = sender->DisableEncodingLayers(disabled_layers);
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
static bool SimulcastIsRejected(
|
|
const ContentInfo* local_content,
|
|
const MediaContentDescription& answer_media_desc) {
|
|
bool simulcast_offered = local_content &&
|
|
local_content->media_description() &&
|
|
local_content->media_description()->HasSimulcast();
|
|
bool simulcast_answered = answer_media_desc.HasSimulcast();
|
|
bool rids_supported = RtpExtension::FindHeaderExtensionByUri(
|
|
answer_media_desc.rtp_header_extensions(), RtpExtension::kRidUri);
|
|
return simulcast_offered && (!simulcast_answered || !rids_supported);
|
|
}
|
|
|
|
static RTCError DisableSimulcastInSender(
|
|
rtc::scoped_refptr<RtpSenderInternal> sender) {
|
|
RTC_DCHECK(sender);
|
|
RtpParameters parameters = sender->GetParametersInternal();
|
|
if (parameters.encodings.size() <= 1) {
|
|
return RTCError::OK();
|
|
}
|
|
|
|
std::vector<std::string> disabled_layers;
|
|
std::transform(
|
|
parameters.encodings.begin() + 1, parameters.encodings.end(),
|
|
std::back_inserter(disabled_layers),
|
|
[](const RtpEncodingParameters& encoding) { return encoding.rid; });
|
|
return sender->DisableEncodingLayers(disabled_layers);
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
|
|
PeerConnection::AssociateTransceiver(cricket::ContentSource source,
|
|
SdpType type,
|
|
size_t mline_index,
|
|
const ContentInfo& content,
|
|
const ContentInfo* old_local_content,
|
|
const ContentInfo* old_remote_content) {
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
// If this is an offer then the m= section might be recycled. If the m=
|
|
// section is being recycled (defined as: rejected in the current local or
|
|
// remote description and not rejected in new description), dissociate the
|
|
// currently associated RtpTransceiver by setting its mid property to null,
|
|
// and discard the mapping between the transceiver and its m= section index.
|
|
if (IsMediaSectionBeingRecycled(type, content, old_local_content,
|
|
old_remote_content)) {
|
|
// We want to dissociate the transceiver that has the rejected mid.
|
|
const std::string& old_mid =
|
|
(old_local_content && old_local_content->rejected)
|
|
? old_local_content->name
|
|
: old_remote_content->name;
|
|
auto old_transceiver = GetAssociatedTransceiver(old_mid);
|
|
if (old_transceiver) {
|
|
RTC_LOG(LS_INFO) << "Dissociating transceiver for MID=" << old_mid
|
|
<< " since the media section is being recycled.";
|
|
old_transceiver->internal()->set_mid(absl::nullopt);
|
|
old_transceiver->internal()->set_mline_index(absl::nullopt);
|
|
}
|
|
}
|
|
const MediaContentDescription* media_desc = content.media_description();
|
|
auto transceiver = GetAssociatedTransceiver(content.name);
|
|
if (source == cricket::CS_LOCAL) {
|
|
// Find the RtpTransceiver that corresponds to this m= section, using the
|
|
// mapping between transceivers and m= section indices established when
|
|
// creating the offer.
|
|
if (!transceiver) {
|
|
transceiver = GetTransceiverByMLineIndex(mline_index);
|
|
}
|
|
if (!transceiver) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"Unknown transceiver");
|
|
}
|
|
} else {
|
|
RTC_DCHECK_EQ(source, cricket::CS_REMOTE);
|
|
// If the m= section is sendrecv or recvonly, and there are RtpTransceivers
|
|
// of the same type...
|
|
// When simulcast is requested, a transceiver cannot be associated because
|
|
// AddTrack cannot be called to initialize it.
|
|
if (!transceiver &&
|
|
RtpTransceiverDirectionHasRecv(media_desc->direction()) &&
|
|
!media_desc->HasSimulcast()) {
|
|
transceiver = FindAvailableTransceiverToReceive(media_desc->type());
|
|
}
|
|
// If no RtpTransceiver was found in the previous step, create one with a
|
|
// recvonly direction.
|
|
if (!transceiver) {
|
|
RTC_LOG(LS_INFO) << "Adding "
|
|
<< cricket::MediaTypeToString(media_desc->type())
|
|
<< " transceiver for MID=" << content.name
|
|
<< " at i=" << mline_index
|
|
<< " in response to the remote description.";
|
|
std::string sender_id = rtc::CreateRandomUuid();
|
|
std::vector<RtpEncodingParameters> send_encodings =
|
|
GetSendEncodingsFromRemoteDescription(*media_desc);
|
|
auto sender = CreateSender(media_desc->type(), sender_id, nullptr, {},
|
|
send_encodings);
|
|
std::string receiver_id;
|
|
if (!media_desc->streams().empty()) {
|
|
receiver_id = media_desc->streams()[0].id;
|
|
} else {
|
|
receiver_id = rtc::CreateRandomUuid();
|
|
}
|
|
auto receiver = CreateReceiver(media_desc->type(), receiver_id);
|
|
transceiver = CreateAndAddTransceiver(sender, receiver);
|
|
transceiver->internal()->set_direction(
|
|
RtpTransceiverDirection::kRecvOnly);
|
|
if (type == SdpType::kOffer) {
|
|
transceiver_stable_states_by_transceivers_[transceiver]
|
|
.set_newly_created();
|
|
}
|
|
}
|
|
// Check if the offer indicated simulcast but the answer rejected it.
|
|
// This can happen when simulcast is not supported on the remote party.
|
|
if (SimulcastIsRejected(old_local_content, *media_desc)) {
|
|
RTC_HISTOGRAM_BOOLEAN(kSimulcastDisabled, true);
|
|
RTCError error =
|
|
DisableSimulcastInSender(transceiver->internal()->sender_internal());
|
|
if (!error.ok()) {
|
|
RTC_LOG(LS_ERROR) << "Failed to remove rejected simulcast.";
|
|
return std::move(error);
|
|
}
|
|
}
|
|
}
|
|
RTC_DCHECK(transceiver);
|
|
if (transceiver->media_type() != media_desc->type()) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::INVALID_PARAMETER,
|
|
"Transceiver type does not match media description type.");
|
|
}
|
|
if (media_desc->HasSimulcast()) {
|
|
std::vector<SimulcastLayer> layers =
|
|
source == cricket::CS_LOCAL
|
|
? media_desc->simulcast_description().send_layers().GetAllLayers()
|
|
: media_desc->simulcast_description()
|
|
.receive_layers()
|
|
.GetAllLayers();
|
|
RTCError error = UpdateSimulcastLayerStatusInSender(
|
|
layers, transceiver->internal()->sender_internal());
|
|
if (!error.ok()) {
|
|
RTC_LOG(LS_ERROR) << "Failed updating status for simulcast layers.";
|
|
return std::move(error);
|
|
}
|
|
}
|
|
if (type == SdpType::kOffer) {
|
|
bool state_changes = transceiver->internal()->mid() != content.name ||
|
|
transceiver->internal()->mline_index() != mline_index;
|
|
if (state_changes) {
|
|
transceiver_stable_states_by_transceivers_[transceiver]
|
|
.SetMSectionIfUnset(transceiver->internal()->mid(),
|
|
transceiver->internal()->mline_index());
|
|
}
|
|
}
|
|
// Associate the found or created RtpTransceiver with the m= section by
|
|
// setting the value of the RtpTransceiver's mid property to the MID of the m=
|
|
// section, and establish a mapping between the transceiver and the index of
|
|
// the m= section.
|
|
transceiver->internal()->set_mid(content.name);
|
|
transceiver->internal()->set_mline_index(mline_index);
|
|
return std::move(transceiver);
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::GetAssociatedTransceiver(const std::string& mid) const {
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
for (auto transceiver : transceivers_) {
|
|
if (transceiver->mid() == mid) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::GetTransceiverByMLineIndex(size_t mline_index) const {
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
for (auto transceiver : transceivers_) {
|
|
if (transceiver->internal()->mline_index() == mline_index) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::FindAvailableTransceiverToReceive(
|
|
cricket::MediaType media_type) const {
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
// From JSEP section 5.10 (Applying a Remote Description):
|
|
// If the m= section is sendrecv or recvonly, and there are RtpTransceivers of
|
|
// the same type that were added to the PeerConnection by addTrack and are not
|
|
// associated with any m= section and are not stopped, find the first such
|
|
// RtpTransceiver.
|
|
for (auto transceiver : transceivers_) {
|
|
if (transceiver->media_type() == media_type &&
|
|
transceiver->internal()->created_by_addtrack() && !transceiver->mid() &&
|
|
!transceiver->stopped()) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
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->media_type());
|
|
}
|
|
}
|
|
|
|
PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return configuration_;
|
|
}
|
|
|
|
RTCError PeerConnection::SetConfiguration(
|
|
const RTCConfiguration& configuration) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
|
|
if (IsClosed()) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
|
|
"SetConfiguration: PeerConnection is closed.");
|
|
}
|
|
|
|
// 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.
|
|
if (local_description() && configuration.ice_candidate_pool_size !=
|
|
configuration_.ice_candidate_pool_size) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
|
|
"Can't change candidate pool size after calling "
|
|
"SetLocalDescription.");
|
|
}
|
|
|
|
if (local_description() &&
|
|
configuration.crypto_options != configuration_.crypto_options) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
|
|
"Can't change crypto_options after calling "
|
|
"SetLocalDescription.");
|
|
}
|
|
|
|
// 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.turn_port_prune_policy = configuration.turn_port_prune_policy;
|
|
modified_config.surface_ice_candidates_on_ice_transport_type_changed =
|
|
configuration.surface_ice_candidates_on_ice_transport_type_changed;
|
|
modified_config.ice_check_min_interval = configuration.ice_check_min_interval;
|
|
modified_config.ice_check_interval_strong_connectivity =
|
|
configuration.ice_check_interval_strong_connectivity;
|
|
modified_config.ice_check_interval_weak_connectivity =
|
|
configuration.ice_check_interval_weak_connectivity;
|
|
modified_config.ice_unwritable_timeout = configuration.ice_unwritable_timeout;
|
|
modified_config.ice_unwritable_min_checks =
|
|
configuration.ice_unwritable_min_checks;
|
|
modified_config.ice_inactive_timeout = configuration.ice_inactive_timeout;
|
|
modified_config.stun_candidate_keepalive_interval =
|
|
configuration.stun_candidate_keepalive_interval;
|
|
modified_config.turn_customizer = configuration.turn_customizer;
|
|
modified_config.network_preference = configuration.network_preference;
|
|
modified_config.active_reset_srtp_params =
|
|
configuration.active_reset_srtp_params;
|
|
modified_config.turn_logging_id = configuration.turn_logging_id;
|
|
modified_config.allow_codec_switching = configuration.allow_codec_switching;
|
|
if (configuration != modified_config) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
|
|
"Modifying the configuration in an unsupported way.");
|
|
}
|
|
|
|
// Validate the modified configuration.
|
|
RTCError validate_error = ValidateConfiguration(modified_config);
|
|
if (!validate_error.ok()) {
|
|
return validate_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 > static_cast<int>(UINT16_MAX)) {
|
|
return RTCError(RTCErrorType::INVALID_RANGE);
|
|
}
|
|
|
|
// 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 RTCError(parse_error);
|
|
}
|
|
// Add the turn logging id to all turn servers
|
|
for (cricket::RelayServerConfig& turn_server : turn_servers) {
|
|
turn_server.turn_logging_id = configuration.turn_logging_id;
|
|
}
|
|
|
|
// Note if STUN or TURN servers were supplied.
|
|
if (!stun_servers.empty()) {
|
|
NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
|
|
}
|
|
if (!turn_servers.empty()) {
|
|
NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
|
|
}
|
|
|
|
// 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.GetTurnPortPrunePolicy(),
|
|
modified_config.turn_customizer,
|
|
modified_config.stun_candidate_keepalive_interval,
|
|
static_cast<bool>(local_description())))) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Failed to apply configuration to PortAllocator.");
|
|
}
|
|
|
|
// 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 ||
|
|
NeedIceRestart(
|
|
configuration_.surface_ice_candidates_on_ice_transport_type_changed,
|
|
configuration_.type, modified_config.type) ||
|
|
modified_config.GetTurnPortPrunePolicy() !=
|
|
configuration_.GetTurnPortPrunePolicy()) {
|
|
transport_controller_->SetNeedsIceRestartFlag();
|
|
}
|
|
|
|
transport_controller_->SetIceConfig(ParseIceConfig(modified_config));
|
|
|
|
if (configuration_.active_reset_srtp_params !=
|
|
modified_config.active_reset_srtp_params) {
|
|
transport_controller_->SetActiveResetSrtpParams(
|
|
modified_config.active_reset_srtp_params);
|
|
}
|
|
|
|
if (modified_config.allow_codec_switching.has_value()) {
|
|
cricket::VideoMediaChannel* video_channel = video_media_channel();
|
|
if (video_channel) {
|
|
video_channel->SetVideoCodecSwitchingEnabled(
|
|
*modified_config.allow_codec_switching);
|
|
}
|
|
}
|
|
|
|
configuration_ = modified_config;
|
|
return RTCError::OK();
|
|
}
|
|
|
|
bool PeerConnection::AddIceCandidate(
|
|
const IceCandidateInterface* ice_candidate) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
|
|
if (IsClosed()) {
|
|
RTC_LOG(LS_ERROR) << "AddIceCandidate: PeerConnection is closed.";
|
|
NoteAddIceCandidateResult(kAddIceCandidateFailClosed);
|
|
return false;
|
|
}
|
|
|
|
if (!remote_description()) {
|
|
RTC_LOG(LS_ERROR) << "AddIceCandidate: ICE candidates can't be added "
|
|
"without any remote session description.";
|
|
NoteAddIceCandidateResult(kAddIceCandidateFailNoRemoteDescription);
|
|
return false;
|
|
}
|
|
|
|
if (!ice_candidate) {
|
|
RTC_LOG(LS_ERROR) << "AddIceCandidate: Candidate is null.";
|
|
NoteAddIceCandidateResult(kAddIceCandidateFailNullCandidate);
|
|
return false;
|
|
}
|
|
|
|
bool valid = false;
|
|
bool ready = ReadyToUseRemoteCandidate(ice_candidate, nullptr, &valid);
|
|
if (!valid) {
|
|
NoteAddIceCandidateResult(kAddIceCandidateFailNotValid);
|
|
return false;
|
|
}
|
|
|
|
// Add this candidate to the remote session description.
|
|
if (!mutable_remote_description()->AddCandidate(ice_candidate)) {
|
|
RTC_LOG(LS_ERROR) << "AddIceCandidate: Candidate cannot be used.";
|
|
NoteAddIceCandidateResult(kAddIceCandidateFailInAddition);
|
|
return false;
|
|
}
|
|
|
|
if (ready) {
|
|
bool result = UseCandidate(ice_candidate);
|
|
if (result) {
|
|
NoteUsageEvent(UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED);
|
|
NoteAddIceCandidateResult(kAddIceCandidateSuccess);
|
|
} else {
|
|
NoteAddIceCandidateResult(kAddIceCandidateFailNotUsable);
|
|
}
|
|
return result;
|
|
} else {
|
|
RTC_LOG(LS_INFO) << "AddIceCandidate: Not ready to use candidate.";
|
|
NoteAddIceCandidateResult(kAddIceCandidateFailNotReady);
|
|
return true;
|
|
}
|
|
}
|
|
|
|
void PeerConnection::AddIceCandidate(
|
|
std::unique_ptr<IceCandidateInterface> candidate,
|
|
std::function<void(RTCError)> callback) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
|
|
candidate = std::move(candidate), callback = std::move(callback)](
|
|
std::function<void()> operations_chain_callback) {
|
|
if (!this_weak_ptr) {
|
|
operations_chain_callback();
|
|
callback(RTCError(
|
|
RTCErrorType::INVALID_STATE,
|
|
"AddIceCandidate failed because the session was shut down"));
|
|
return;
|
|
}
|
|
if (!this_weak_ptr->AddIceCandidate(candidate.get())) {
|
|
operations_chain_callback();
|
|
// Fail with an error type and message consistent with Chromium.
|
|
// TODO(hbos): Fail with error types according to spec.
|
|
callback(RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
|
|
"Error processing ICE candidate"));
|
|
return;
|
|
}
|
|
operations_chain_callback();
|
|
callback(RTCError::OK());
|
|
});
|
|
}
|
|
|
|
bool PeerConnection::RemoveIceCandidates(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (IsClosed()) {
|
|
RTC_LOG(LS_ERROR) << "RemoveIceCandidates: PeerConnection is closed.";
|
|
return false;
|
|
}
|
|
|
|
if (!remote_description()) {
|
|
RTC_LOG(LS_ERROR) << "RemoveIceCandidates: ICE candidates can't be removed "
|
|
"without any remote session description.";
|
|
return false;
|
|
}
|
|
|
|
if (candidates.empty()) {
|
|
RTC_LOG(LS_ERROR) << "RemoveIceCandidates: candidates are empty.";
|
|
return false;
|
|
}
|
|
|
|
size_t number_removed =
|
|
mutable_remote_description()->RemoveCandidates(candidates);
|
|
if (number_removed != candidates.size()) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "RemoveIceCandidates: Failed to remove candidates. Requested "
|
|
<< candidates.size() << " but only " << number_removed
|
|
<< " are removed.";
|
|
}
|
|
|
|
// Remove the candidates from the transport controller.
|
|
RTCError error = transport_controller_->RemoveRemoteCandidates(candidates);
|
|
if (!error.ok()) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "RemoveIceCandidates: Error when removing remote candidates: "
|
|
<< error.message();
|
|
}
|
|
return true;
|
|
}
|
|
|
|
RTCError PeerConnection::SetBitrate(const BitrateSettings& bitrate) {
|
|
if (!worker_thread()->IsCurrent()) {
|
|
return worker_thread()->Invoke<RTCError>(
|
|
RTC_FROM_HERE, [&]() { return SetBitrate(bitrate); });
|
|
}
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
|
|
const bool has_min = bitrate.min_bitrate_bps.has_value();
|
|
const bool has_start = bitrate.start_bitrate_bps.has_value();
|
|
const bool has_max = bitrate.max_bitrate_bps.has_value();
|
|
if (has_min && *bitrate.min_bitrate_bps < 0) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"min_bitrate_bps <= 0");
|
|
}
|
|
if (has_start) {
|
|
if (has_min && *bitrate.start_bitrate_bps < *bitrate.min_bitrate_bps) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"start_bitrate_bps < min_bitrate_bps");
|
|
} else if (*bitrate.start_bitrate_bps < 0) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"curent_bitrate_bps < 0");
|
|
}
|
|
}
|
|
if (has_max) {
|
|
if (has_start && *bitrate.max_bitrate_bps < *bitrate.start_bitrate_bps) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"max_bitrate_bps < start_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");
|
|
}
|
|
}
|
|
|
|
RTC_DCHECK(call_.get());
|
|
call_->SetClientBitratePreferences(bitrate);
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
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()->voice().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()->voice().GetAudioState();
|
|
audio_state->SetRecording(recording);
|
|
}
|
|
|
|
std::unique_ptr<rtc::SSLCertificate>
|
|
PeerConnection::GetRemoteAudioSSLCertificate() {
|
|
std::unique_ptr<rtc::SSLCertChain> chain = GetRemoteAudioSSLCertChain();
|
|
if (!chain || !chain->GetSize()) {
|
|
return nullptr;
|
|
}
|
|
return chain->Get(0).Clone();
|
|
}
|
|
|
|
std::unique_ptr<rtc::SSLCertChain>
|
|
PeerConnection::GetRemoteAudioSSLCertChain() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
auto audio_transceiver = GetFirstAudioTransceiver();
|
|
if (!audio_transceiver || !audio_transceiver->internal()->channel()) {
|
|
return nullptr;
|
|
}
|
|
return transport_controller_->GetRemoteSSLCertChain(
|
|
audio_transceiver->internal()->channel()->transport_name());
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::GetFirstAudioTransceiver() const {
|
|
for (auto transceiver : transceivers_) {
|
|
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
void PeerConnection::AddAdaptationResource(
|
|
rtc::scoped_refptr<Resource> resource) {
|
|
if (!worker_thread()->IsCurrent()) {
|
|
return worker_thread()->Invoke<void>(RTC_FROM_HERE, [this, resource]() {
|
|
return AddAdaptationResource(resource);
|
|
});
|
|
}
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
if (!call_) {
|
|
// The PeerConnection has been closed.
|
|
return;
|
|
}
|
|
call_->AddAdaptationResource(resource);
|
|
}
|
|
|
|
bool PeerConnection::StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
|
|
int64_t output_period_ms) {
|
|
return worker_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE,
|
|
[this, output = std::move(output), output_period_ms]() mutable {
|
|
return StartRtcEventLog_w(std::move(output), output_period_ms);
|
|
});
|
|
}
|
|
|
|
bool PeerConnection::StartRtcEventLog(
|
|
std::unique_ptr<RtcEventLogOutput> output) {
|
|
int64_t output_period_ms = webrtc::RtcEventLog::kImmediateOutput;
|
|
if (field_trial::IsEnabled("WebRTC-RtcEventLogNewFormat")) {
|
|
output_period_ms = 5000;
|
|
}
|
|
return StartRtcEventLog(std::move(output), output_period_ms);
|
|
}
|
|
|
|
void PeerConnection::StopRtcEventLog() {
|
|
worker_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&PeerConnection::StopRtcEventLog_w, this));
|
|
}
|
|
|
|
rtc::scoped_refptr<DtlsTransportInterface>
|
|
PeerConnection::LookupDtlsTransportByMid(const std::string& mid) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return transport_controller_->LookupDtlsTransportByMid(mid);
|
|
}
|
|
|
|
rtc::scoped_refptr<DtlsTransport>
|
|
PeerConnection::LookupDtlsTransportByMidInternal(const std::string& mid) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return transport_controller_->LookupDtlsTransportByMid(mid);
|
|
}
|
|
|
|
rtc::scoped_refptr<SctpTransportInterface> PeerConnection::GetSctpTransport()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (!sctp_mid_s_) {
|
|
return nullptr;
|
|
}
|
|
return transport_controller_->GetSctpTransport(*sctp_mid_s_);
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::local_description() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return pending_local_description_ ? pending_local_description_.get()
|
|
: current_local_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::remote_description() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return pending_remote_description_ ? pending_remote_description_.get()
|
|
: current_remote_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::current_local_description()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return current_local_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::current_remote_description()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return current_remote_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::pending_local_description()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return pending_local_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::pending_remote_description()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return pending_remote_description_.get();
|
|
}
|
|
|
|
void PeerConnection::Close() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
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);
|
|
NoteUsageEvent(UsageEvent::CLOSE_CALLED);
|
|
|
|
for (const auto& transceiver : transceivers_) {
|
|
transceiver->Stop();
|
|
}
|
|
|
|
// Ensure that all asynchronous stats requests are completed before destroying
|
|
// the transport controller below.
|
|
if (stats_collector_) {
|
|
stats_collector_->WaitForPendingRequest();
|
|
}
|
|
|
|
// Don't destroy BaseChannels until after stats has been cleaned up so that
|
|
// the last stats request can still read from the channels.
|
|
DestroyAllChannels();
|
|
|
|
// The event log is used in the transport controller, which must be outlived
|
|
// by the former. CreateOffer by the peer connection is implemented
|
|
// asynchronously and if the peer connection is closed without resetting the
|
|
// WebRTC session description factory, the session description factory would
|
|
// call the transport controller.
|
|
webrtc_session_desc_factory_.reset();
|
|
transport_controller_.reset();
|
|
|
|
network_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
|
|
port_allocator_.get()));
|
|
|
|
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
call_.reset();
|
|
// The event log must outlive call (and any other object that uses it).
|
|
event_log_.reset();
|
|
});
|
|
ReportUsagePattern();
|
|
// The .h file says that observer can be discarded after close() returns.
|
|
// Make sure this is true.
|
|
observer_ = nullptr;
|
|
}
|
|
|
|
void PeerConnection::OnMessage(rtc::Message* msg) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
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(std::move(param->error));
|
|
delete param;
|
|
break;
|
|
}
|
|
case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
|
|
CreateSessionDescriptionMsg* param =
|
|
static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
|
|
param->observer->OnFailure(std::move(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_REPORT_USAGE_PATTERN: {
|
|
ReportUsagePattern();
|
|
break;
|
|
}
|
|
default:
|
|
RTC_NOTREACHED() << "Not implemented";
|
|
break;
|
|
}
|
|
}
|
|
|
|
cricket::VoiceMediaChannel* PeerConnection::voice_media_channel() const {
|
|
RTC_DCHECK(!IsUnifiedPlan());
|
|
auto* voice_channel = static_cast<cricket::VoiceChannel*>(
|
|
GetAudioTransceiver()->internal()->channel());
|
|
if (voice_channel) {
|
|
return voice_channel->media_channel();
|
|
} else {
|
|
return nullptr;
|
|
}
|
|
}
|
|
|
|
cricket::VideoMediaChannel* PeerConnection::video_media_channel() const {
|
|
RTC_DCHECK(!IsUnifiedPlan());
|
|
auto* video_channel = static_cast<cricket::VideoChannel*>(
|
|
GetVideoTransceiver()->internal()->channel());
|
|
if (video_channel) {
|
|
return video_channel->media_channel();
|
|
} else {
|
|
return nullptr;
|
|
}
|
|
}
|
|
|
|
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));
|
|
// TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
|
|
// the constructor taking stream IDs instead.
|
|
auto* audio_receiver = new AudioRtpReceiver(
|
|
worker_thread(), remote_sender_info.sender_id, streams);
|
|
audio_receiver->SetMediaChannel(voice_media_channel());
|
|
if (remote_sender_info.sender_id == kDefaultAudioSenderId) {
|
|
audio_receiver->SetupUnsignaledMediaChannel();
|
|
} else {
|
|
audio_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
|
|
}
|
|
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
|
signaling_thread(), audio_receiver);
|
|
GetAudioTransceiver()->internal()->AddReceiver(receiver);
|
|
Observer()->OnAddTrack(receiver, streams);
|
|
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
|
|
}
|
|
|
|
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));
|
|
// TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
|
|
// the constructor taking stream IDs instead.
|
|
auto* video_receiver = new VideoRtpReceiver(
|
|
worker_thread(), remote_sender_info.sender_id, streams);
|
|
video_receiver->SetMediaChannel(video_media_channel());
|
|
if (remote_sender_info.sender_id == kDefaultVideoSenderId) {
|
|
video_receiver->SetupUnsignaledMediaChannel();
|
|
} else {
|
|
video_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
|
|
}
|
|
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
|
signaling_thread(), video_receiver);
|
|
GetVideoTransceiver()->internal()->AddReceiver(receiver);
|
|
Observer()->OnAddTrack(receiver, streams);
|
|
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
|
|
}
|
|
|
|
// 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());
|
|
RTC_DCHECK(track);
|
|
RTC_DCHECK(stream);
|
|
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_ids({stream->id()});
|
|
return;
|
|
}
|
|
|
|
// Normal case; we've never seen this track before.
|
|
auto new_sender = CreateSender(cricket::MEDIA_TYPE_AUDIO, track->id(), track,
|
|
{stream->id()}, {});
|
|
new_sender->internal()->SetMediaChannel(voice_media_channel());
|
|
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->id(), 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());
|
|
RTC_DCHECK(track);
|
|
RTC_DCHECK(stream);
|
|
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_ids({stream->id()});
|
|
return;
|
|
}
|
|
|
|
// Normal case; we've never seen this track before.
|
|
auto new_sender = CreateSender(cricket::MEDIA_TYPE_VIDEO, track->id(), track,
|
|
{stream->id()}, {});
|
|
new_sender->internal()->SetMediaChannel(video_media_channel());
|
|
GetVideoTransceiver()->internal()->AddSender(new_sender);
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(local_video_sender_infos_, stream->id(), 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) {
|
|
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::SetStandardizedIceConnectionState(
|
|
PeerConnectionInterface::IceConnectionState new_state) {
|
|
if (standardized_ice_connection_state_ == new_state) {
|
|
return;
|
|
}
|
|
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
|
|
RTC_LOG(LS_INFO) << "Changing standardized IceConnectionState "
|
|
<< standardized_ice_connection_state_ << " => " << new_state;
|
|
|
|
standardized_ice_connection_state_ = new_state;
|
|
Observer()->OnStandardizedIceConnectionChange(new_state);
|
|
}
|
|
|
|
void PeerConnection::SetConnectionState(
|
|
PeerConnectionInterface::PeerConnectionState new_state) {
|
|
if (connection_state_ == new_state)
|
|
return;
|
|
if (IsClosed())
|
|
return;
|
|
connection_state_ = new_state;
|
|
Observer()->OnConnectionChange(new_state);
|
|
}
|
|
|
|
void PeerConnection::OnIceGatheringChange(
|
|
PeerConnectionInterface::IceGatheringState new_state) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
ice_gathering_state_ = new_state;
|
|
Observer()->OnIceGatheringChange(ice_gathering_state_);
|
|
}
|
|
|
|
void PeerConnection::OnIceCandidate(
|
|
std::unique_ptr<IceCandidateInterface> candidate) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
ReportIceCandidateCollected(candidate->candidate());
|
|
Observer()->OnIceCandidate(candidate.get());
|
|
}
|
|
|
|
void PeerConnection::OnIceCandidateError(const std::string& address,
|
|
int port,
|
|
const std::string& url,
|
|
int error_code,
|
|
const std::string& error_text) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
Observer()->OnIceCandidateError(address, port, url, error_code, error_text);
|
|
// Leftover not to break wpt test during migration to the new API.
|
|
Observer()->OnIceCandidateError(address + ":", url, error_code, error_text);
|
|
}
|
|
|
|
void PeerConnection::OnIceCandidatesRemoved(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
Observer()->OnIceCandidatesRemoved(candidates);
|
|
}
|
|
|
|
void PeerConnection::OnSelectedCandidatePairChanged(
|
|
const cricket::CandidatePairChangeEvent& event) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
|
|
if (event.selected_candidate_pair.local_candidate().type() ==
|
|
LOCAL_PORT_TYPE &&
|
|
event.selected_candidate_pair.remote_candidate().type() ==
|
|
LOCAL_PORT_TYPE) {
|
|
NoteUsageEvent(UsageEvent::DIRECT_CONNECTION_SELECTED);
|
|
}
|
|
|
|
Observer()->OnIceSelectedCandidatePairChanged(event);
|
|
}
|
|
|
|
void PeerConnection::ChangeSignalingState(
|
|
PeerConnectionInterface::SignalingState signaling_state) {
|
|
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_);
|
|
standardized_ice_connection_state_ =
|
|
PeerConnectionInterface::IceConnectionState::kIceConnectionClosed;
|
|
connection_state_ = PeerConnectionInterface::PeerConnectionState::kClosed;
|
|
Observer()->OnConnectionChange(connection_state_);
|
|
}
|
|
Observer()->OnSignalingChange(signaling_state_);
|
|
}
|
|
|
|
void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
AddAudioTrack(track, stream);
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
RemoveAudioTrack(track, stream);
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
AddVideoTrack(track, stream);
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
RemoveVideoTrack(track, stream);
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
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,
|
|
RTCError&& error) {
|
|
RTC_DCHECK(!error.ok());
|
|
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
|
|
msg->error = std::move(error);
|
|
signaling_thread()->Post(RTC_FROM_HERE, this,
|
|
MSG_SET_SESSIONDESCRIPTION_FAILED, msg);
|
|
}
|
|
|
|
void PeerConnection::PostCreateSessionDescriptionFailure(
|
|
CreateSessionDescriptionObserver* observer,
|
|
RTCError error) {
|
|
RTC_DCHECK(!error.ok());
|
|
CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
|
|
msg->error = std::move(error);
|
|
signaling_thread()->Post(RTC_FROM_HERE, this,
|
|
MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
|
|
}
|
|
|
|
void PeerConnection::GetOptionsForOffer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
ExtractSharedMediaSessionOptions(offer_answer_options, session_options);
|
|
|
|
if (IsUnifiedPlan()) {
|
|
GetOptionsForUnifiedPlanOffer(offer_answer_options, session_options);
|
|
} else {
|
|
GetOptionsForPlanBOffer(offer_answer_options, session_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 (data_channel_controller_.HasRtpDataChannels() ||
|
|
data_channel_type() != cricket::DCT_RTP) {
|
|
session_options->data_channel_type = data_channel_type();
|
|
}
|
|
|
|
// Apply ICE restart flag and renomination flag.
|
|
bool ice_restart = offer_answer_options.ice_restart ||
|
|
local_ice_credentials_to_replace_->HasIceCredentials();
|
|
for (auto& options : session_options->media_description_options) {
|
|
options.transport_options.ice_restart = ice_restart;
|
|
options.transport_options.enable_ice_renomination =
|
|
configuration_.enable_ice_renomination;
|
|
}
|
|
|
|
session_options->rtcp_cname = rtcp_cname_;
|
|
session_options->crypto_options = GetCryptoOptions();
|
|
session_options->pooled_ice_credentials =
|
|
network_thread()->Invoke<std::vector<cricket::IceParameters>>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&cricket::PortAllocator::GetPooledIceCredentials,
|
|
port_allocator_.get()));
|
|
session_options->offer_extmap_allow_mixed =
|
|
configuration_.offer_extmap_allow_mixed;
|
|
|
|
// Allow fallback for using obsolete SCTP syntax.
|
|
// Note that the default in |session_options| is true, while
|
|
// the default in |options| is false.
|
|
session_options->use_obsolete_sctp_sdp =
|
|
offer_answer_options.use_obsolete_sctp_sdp;
|
|
}
|
|
|
|
void PeerConnection::GetOptionsForPlanBOffer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
// Figure out transceiver directional preferences.
|
|
bool send_audio = !GetAudioTransceiver()->internal()->senders().empty();
|
|
bool send_video = !GetVideoTransceiver()->internal()->senders().empty();
|
|
|
|
// 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 = data_channel_controller_.HasDataChannels();
|
|
|
|
// The "offer_to_receive_X" options allow those defaults to be overridden.
|
|
if (offer_answer_options.offer_to_receive_audio !=
|
|
RTCOfferAnswerOptions::kUndefined) {
|
|
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
|
|
offer_new_audio_description =
|
|
offer_new_audio_description ||
|
|
(offer_answer_options.offer_to_receive_audio > 0);
|
|
}
|
|
if (offer_answer_options.offer_to_receive_video !=
|
|
RTCOfferAnswerOptions::kUndefined) {
|
|
recv_video = (offer_answer_options.offer_to_receive_video > 0);
|
|
offer_new_video_description =
|
|
offer_new_video_description ||
|
|
(offer_answer_options.offer_to_receive_video > 0);
|
|
}
|
|
|
|
absl::optional<size_t> audio_index;
|
|
absl::optional<size_t> video_index;
|
|
absl::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) {
|
|
cricket::MediaDescriptionOptions options(
|
|
cricket::MEDIA_TYPE_AUDIO, cricket::CN_AUDIO,
|
|
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio), false);
|
|
options.header_extensions =
|
|
channel_manager()->GetSupportedAudioRtpHeaderExtensions();
|
|
session_options->media_description_options.push_back(options);
|
|
audio_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
if (!video_index && offer_new_video_description) {
|
|
cricket::MediaDescriptionOptions options(
|
|
cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO,
|
|
RtpTransceiverDirectionFromSendRecv(send_video, recv_video), false);
|
|
options.header_extensions =
|
|
channel_manager()->GetSupportedVideoRtpHeaderExtensions();
|
|
session_options->media_description_options.push_back(options);
|
|
video_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
if (!data_index && offer_new_data_description) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForActiveData(cricket::CN_DATA));
|
|
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];
|
|
|
|
AddPlanBRtpSenderOptions(GetSendersInternal(),
|
|
audio_media_description_options,
|
|
video_media_description_options,
|
|
offer_answer_options.num_simulcast_layers);
|
|
}
|
|
|
|
static cricket::MediaDescriptionOptions
|
|
GetMediaDescriptionOptionsForTransceiver(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver,
|
|
const std::string& mid) {
|
|
cricket::MediaDescriptionOptions media_description_options(
|
|
transceiver->media_type(), mid, transceiver->direction(),
|
|
transceiver->stopped());
|
|
media_description_options.codec_preferences =
|
|
transceiver->codec_preferences();
|
|
media_description_options.header_extensions =
|
|
transceiver->HeaderExtensionsToOffer();
|
|
// This behavior is specified in JSEP. The gist is that:
|
|
// 1. The MSID is included if the RtpTransceiver's direction is sendonly or
|
|
// sendrecv.
|
|
// 2. If the MSID is included, then it must be included in any subsequent
|
|
// offer/answer exactly the same until the RtpTransceiver is stopped.
|
|
if (transceiver->stopped() ||
|
|
(!RtpTransceiverDirectionHasSend(transceiver->direction()) &&
|
|
!transceiver->internal()->has_ever_been_used_to_send())) {
|
|
return media_description_options;
|
|
}
|
|
|
|
cricket::SenderOptions sender_options;
|
|
sender_options.track_id = transceiver->sender()->id();
|
|
sender_options.stream_ids = transceiver->sender()->stream_ids();
|
|
|
|
// The following sets up RIDs and Simulcast.
|
|
// RIDs are included if Simulcast is requested or if any RID was specified.
|
|
RtpParameters send_parameters =
|
|
transceiver->internal()->sender_internal()->GetParametersInternal();
|
|
bool has_rids = std::any_of(send_parameters.encodings.begin(),
|
|
send_parameters.encodings.end(),
|
|
[](const RtpEncodingParameters& encoding) {
|
|
return !encoding.rid.empty();
|
|
});
|
|
|
|
std::vector<RidDescription> send_rids;
|
|
SimulcastLayerList send_layers;
|
|
for (const RtpEncodingParameters& encoding : send_parameters.encodings) {
|
|
if (encoding.rid.empty()) {
|
|
continue;
|
|
}
|
|
send_rids.push_back(RidDescription(encoding.rid, RidDirection::kSend));
|
|
send_layers.AddLayer(SimulcastLayer(encoding.rid, !encoding.active));
|
|
}
|
|
|
|
if (has_rids) {
|
|
sender_options.rids = send_rids;
|
|
}
|
|
|
|
sender_options.simulcast_layers = send_layers;
|
|
// When RIDs are configured, we must set num_sim_layers to 0 to.
|
|
// Otherwise, num_sim_layers must be 1 because either there is no
|
|
// simulcast, or simulcast is acheived by munging the SDP.
|
|
sender_options.num_sim_layers = has_rids ? 0 : 1;
|
|
media_description_options.sender_options.push_back(sender_options);
|
|
|
|
return media_description_options;
|
|
}
|
|
|
|
// Returns the ContentInfo at mline index |i|, or null if none exists.
|
|
static const ContentInfo* GetContentByIndex(
|
|
const SessionDescriptionInterface* sdesc,
|
|
size_t i) {
|
|
if (!sdesc) {
|
|
return nullptr;
|
|
}
|
|
const ContentInfos& contents = sdesc->description()->contents();
|
|
return (i < contents.size() ? &contents[i] : nullptr);
|
|
}
|
|
|
|
void PeerConnection::GetOptionsForUnifiedPlanOffer(
|
|
const RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
// Rules for generating an offer are dictated by JSEP sections 5.2.1 (Initial
|
|
// Offers) and 5.2.2 (Subsequent Offers).
|
|
RTC_DCHECK_EQ(session_options->media_description_options.size(), 0);
|
|
const ContentInfos no_infos;
|
|
const ContentInfos& local_contents =
|
|
(local_description() ? local_description()->description()->contents()
|
|
: no_infos);
|
|
const ContentInfos& remote_contents =
|
|
(remote_description() ? remote_description()->description()->contents()
|
|
: no_infos);
|
|
// The mline indices that can be recycled. New transceivers should reuse these
|
|
// slots first.
|
|
std::queue<size_t> recycleable_mline_indices;
|
|
// First, go through each media section that exists in either the local or
|
|
// remote description and generate a media section in this offer for the
|
|
// associated transceiver. If a media section can be recycled, generate a
|
|
// default, rejected media section here that can be later overwritten.
|
|
for (size_t i = 0;
|
|
i < std::max(local_contents.size(), remote_contents.size()); ++i) {
|
|
// Either |local_content| or |remote_content| is non-null.
|
|
const ContentInfo* local_content =
|
|
(i < local_contents.size() ? &local_contents[i] : nullptr);
|
|
const ContentInfo* current_local_content =
|
|
GetContentByIndex(current_local_description(), i);
|
|
const ContentInfo* remote_content =
|
|
(i < remote_contents.size() ? &remote_contents[i] : nullptr);
|
|
const ContentInfo* current_remote_content =
|
|
GetContentByIndex(current_remote_description(), i);
|
|
bool had_been_rejected =
|
|
(current_local_content && current_local_content->rejected) ||
|
|
(current_remote_content && current_remote_content->rejected);
|
|
const std::string& mid =
|
|
(local_content ? local_content->name : remote_content->name);
|
|
cricket::MediaType media_type =
|
|
(local_content ? local_content->media_description()->type()
|
|
: remote_content->media_description()->type());
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO) {
|
|
auto transceiver = GetAssociatedTransceiver(mid);
|
|
RTC_CHECK(transceiver);
|
|
// A media section is considered eligible for recycling if it is marked as
|
|
// rejected in either the current local or current remote description.
|
|
if (had_been_rejected && transceiver->stopped()) {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(transceiver->media_type(), mid,
|
|
RtpTransceiverDirection::kInactive,
|
|
/*stopped=*/true));
|
|
recycleable_mline_indices.push(i);
|
|
} else {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForTransceiver(transceiver, mid));
|
|
// CreateOffer shouldn't really cause any state changes in
|
|
// PeerConnection, but we need a way to match new transceivers to new
|
|
// media sections in SetLocalDescription and JSEP specifies this is done
|
|
// by recording the index of the media section generated for the
|
|
// transceiver in the offer.
|
|
transceiver->internal()->set_mline_index(i);
|
|
}
|
|
} else {
|
|
RTC_CHECK_EQ(cricket::MEDIA_TYPE_DATA, media_type);
|
|
if (had_been_rejected) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForRejectedData(mid));
|
|
} else {
|
|
RTC_CHECK(GetDataMid());
|
|
if (mid == *GetDataMid()) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForActiveData(mid));
|
|
} else {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForRejectedData(mid));
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// Next, look for transceivers that are newly added (that is, are not stopped
|
|
// and not associated). Reuse media sections marked as recyclable first,
|
|
// otherwise append to the end of the offer. New media sections should be
|
|
// added in the order they were added to the PeerConnection.
|
|
for (const auto& transceiver : transceivers_) {
|
|
if (transceiver->mid() || transceiver->stopped()) {
|
|
continue;
|
|
}
|
|
size_t mline_index;
|
|
if (!recycleable_mline_indices.empty()) {
|
|
mline_index = recycleable_mline_indices.front();
|
|
recycleable_mline_indices.pop();
|
|
session_options->media_description_options[mline_index] =
|
|
GetMediaDescriptionOptionsForTransceiver(transceiver,
|
|
mid_generator_());
|
|
} else {
|
|
mline_index = session_options->media_description_options.size();
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForTransceiver(transceiver,
|
|
mid_generator_()));
|
|
}
|
|
// See comment above for why CreateOffer changes the transceiver's state.
|
|
transceiver->internal()->set_mline_index(mline_index);
|
|
}
|
|
// Lastly, add a m-section if we have local data channels and an m section
|
|
// does not already exist.
|
|
if (!GetDataMid() && data_channel_controller_.HasDataChannels()) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForActiveData(mid_generator_()));
|
|
}
|
|
}
|
|
|
|
void PeerConnection::GetOptionsForAnswer(
|
|
const RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
ExtractSharedMediaSessionOptions(offer_answer_options, session_options);
|
|
|
|
if (IsUnifiedPlan()) {
|
|
GetOptionsForUnifiedPlanAnswer(offer_answer_options, session_options);
|
|
} else {
|
|
GetOptionsForPlanBAnswer(offer_answer_options, session_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 (data_channel_controller_.HasRtpDataChannels() ||
|
|
data_channel_type() != cricket::DCT_RTP) {
|
|
session_options->data_channel_type = data_channel_type();
|
|
}
|
|
|
|
// Apply ICE renomination flag.
|
|
for (auto& options : session_options->media_description_options) {
|
|
options.transport_options.enable_ice_renomination =
|
|
configuration_.enable_ice_renomination;
|
|
}
|
|
|
|
session_options->rtcp_cname = rtcp_cname_;
|
|
session_options->crypto_options = GetCryptoOptions();
|
|
session_options->pooled_ice_credentials =
|
|
network_thread()->Invoke<std::vector<cricket::IceParameters>>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&cricket::PortAllocator::GetPooledIceCredentials,
|
|
port_allocator_.get()));
|
|
}
|
|
|
|
void PeerConnection::GetOptionsForPlanBAnswer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
// Figure out transceiver directional preferences.
|
|
bool send_audio = !GetAudioTransceiver()->internal()->senders().empty();
|
|
bool send_video = !GetVideoTransceiver()->internal()->senders().empty();
|
|
|
|
// 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 (offer_answer_options.offer_to_receive_audio !=
|
|
RTCOfferAnswerOptions::kUndefined) {
|
|
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
|
|
}
|
|
if (offer_answer_options.offer_to_receive_video !=
|
|
RTCOfferAnswerOptions::kUndefined) {
|
|
recv_video = (offer_answer_options.offer_to_receive_video > 0);
|
|
}
|
|
|
|
absl::optional<size_t> audio_index;
|
|
absl::optional<size_t> video_index;
|
|
absl::optional<size_t> data_index;
|
|
|
|
// 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];
|
|
|
|
AddPlanBRtpSenderOptions(GetSendersInternal(),
|
|
audio_media_description_options,
|
|
video_media_description_options,
|
|
offer_answer_options.num_simulcast_layers);
|
|
}
|
|
|
|
void PeerConnection::GetOptionsForUnifiedPlanAnswer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
// Rules for generating an answer are dictated by JSEP sections 5.3.1 (Initial
|
|
// Answers) and 5.3.2 (Subsequent Answers).
|
|
RTC_DCHECK(remote_description());
|
|
RTC_DCHECK(remote_description()->GetType() == SdpType::kOffer);
|
|
for (const ContentInfo& content :
|
|
remote_description()->description()->contents()) {
|
|
cricket::MediaType media_type = content.media_description()->type();
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO) {
|
|
auto transceiver = GetAssociatedTransceiver(content.name);
|
|
RTC_CHECK(transceiver);
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForTransceiver(transceiver, content.name));
|
|
} else {
|
|
RTC_CHECK_EQ(cricket::MEDIA_TYPE_DATA, media_type);
|
|
// Reject all data sections if data channels are disabled.
|
|
// Reject a data section if it has already been rejected.
|
|
// Reject all data sections except for the first one.
|
|
if (data_channel_type() == cricket::DCT_NONE || content.rejected ||
|
|
content.name != *GetDataMid()) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForRejectedData(content.name));
|
|
} else {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForActiveData(content.name));
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::GenerateMediaDescriptionOptions(
|
|
const SessionDescriptionInterface* session_desc,
|
|
RtpTransceiverDirection audio_direction,
|
|
RtpTransceiverDirection video_direction,
|
|
absl::optional<size_t>* audio_index,
|
|
absl::optional<size_t>* video_index,
|
|
absl::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, /*stopped=*/true));
|
|
} else {
|
|
bool stopped = (audio_direction == RtpTransceiverDirection::kInactive);
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO,
|
|
content.name, audio_direction,
|
|
stopped));
|
|
*audio_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
session_options->media_description_options.back().header_extensions =
|
|
channel_manager()->GetSupportedAudioRtpHeaderExtensions();
|
|
} 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, /*stopped=*/true));
|
|
} else {
|
|
bool stopped = (video_direction == RtpTransceiverDirection::kInactive);
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO,
|
|
content.name, video_direction,
|
|
stopped));
|
|
*video_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
session_options->media_description_options.back().header_extensions =
|
|
channel_manager()->GetSupportedVideoRtpHeaderExtensions();
|
|
} 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(
|
|
GetMediaDescriptionOptionsForRejectedData(content.name));
|
|
} else {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForActiveData(content.name));
|
|
*data_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
cricket::MediaDescriptionOptions
|
|
PeerConnection::GetMediaDescriptionOptionsForActiveData(
|
|
const std::string& mid) const {
|
|
// Direction for data sections is meaningless, but legacy endpoints might
|
|
// expect sendrecv.
|
|
cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
|
|
RtpTransceiverDirection::kSendRecv,
|
|
/*stopped=*/false);
|
|
AddRtpDataChannelOptions(*data_channel_controller_.rtp_data_channels(),
|
|
&options);
|
|
return options;
|
|
}
|
|
|
|
cricket::MediaDescriptionOptions
|
|
PeerConnection::GetMediaDescriptionOptionsForRejectedData(
|
|
const std::string& mid) const {
|
|
cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
|
|
RtpTransceiverDirection::kInactive,
|
|
/*stopped=*/true);
|
|
AddRtpDataChannelOptions(*data_channel_controller_.rtp_data_channels(),
|
|
&options);
|
|
return options;
|
|
}
|
|
|
|
absl::optional<std::string> PeerConnection::GetDataMid() const {
|
|
switch (data_channel_type()) {
|
|
case cricket::DCT_RTP:
|
|
if (!data_channel_controller_.rtp_data_channel()) {
|
|
return absl::nullopt;
|
|
}
|
|
return data_channel_controller_.rtp_data_channel()->content_name();
|
|
case cricket::DCT_SCTP:
|
|
return sctp_mid_s_;
|
|
default:
|
|
return absl::nullopt;
|
|
}
|
|
}
|
|
|
|
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) {
|
|
RTC_DCHECK(!IsUnifiedPlan());
|
|
|
|
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);
|
|
std::string params_stream_id;
|
|
if (params) {
|
|
params_stream_id =
|
|
(!params->first_stream_id().empty() ? params->first_stream_id()
|
|
: kDefaultStreamId);
|
|
}
|
|
bool sender_exists = params && params->id == info.sender_id &&
|
|
params_stream_id == info.stream_id;
|
|
// If this is a default track, and we still need it, don't remove it.
|
|
if ((info.stream_id == kDefaultStreamId && 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) {
|
|
if (!params.has_ssrcs()) {
|
|
// The remote endpoint has streams, but didn't signal ssrcs. For an active
|
|
// sender, this means it is coming from a Unified Plan endpoint,so we just
|
|
// create a default.
|
|
default_sender_needed = true;
|
|
break;
|
|
}
|
|
|
|
// |params.id| is the sender id and the stream id uses the first of
|
|
// |params.stream_ids|. The remote description could come from a Unified
|
|
// Plan endpoint, with multiple or no stream_ids() signaled. Since this is
|
|
// not supported in Plan B, we just take the first here and create the
|
|
// default stream ID if none is specified.
|
|
const std::string& stream_id =
|
|
(!params.first_stream_id().empty() ? params.first_stream_id()
|
|
: kDefaultStreamId);
|
|
const std::string& sender_id = params.id;
|
|
uint32_t ssrc = params.first_ssrc();
|
|
|
|
rtc::scoped_refptr<MediaStreamInterface> stream =
|
|
remote_streams_->find(stream_id);
|
|
if (!stream) {
|
|
// This is a new MediaStream. Create a new remote MediaStream.
|
|
stream = MediaStreamProxy::Create(rtc::Thread::Current(),
|
|
MediaStream::Create(stream_id));
|
|
remote_streams_->AddStream(stream);
|
|
new_streams->AddStream(stream);
|
|
}
|
|
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(*current_senders, stream_id, sender_id);
|
|
if (!sender_info) {
|
|
current_senders->push_back(RtpSenderInfo(stream_id, 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(kDefaultStreamId);
|
|
if (!default_stream) {
|
|
// Create the new default MediaStream.
|
|
default_stream = MediaStreamProxy::Create(
|
|
rtc::Thread::Current(), MediaStream::Create(kDefaultStreamId));
|
|
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, kDefaultStreamId, default_sender_id);
|
|
if (!default_sender_info) {
|
|
current_senders->push_back(
|
|
RtpSenderInfo(kDefaultStreamId, default_sender_id, /*ssrc=*/0));
|
|
OnRemoteSenderAdded(current_senders->back(), media_type);
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnRemoteSenderAdded(const RtpSenderInfo& sender_info,
|
|
cricket::MediaType media_type) {
|
|
RTC_LOG(LS_INFO) << "Creating " << cricket::MediaTypeToString(media_type)
|
|
<< " receiver for track_id=" << sender_info.sender_id
|
|
<< " and stream_id=" << sender_info.stream_id;
|
|
|
|
MediaStreamInterface* stream = remote_streams_->find(sender_info.stream_id);
|
|
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) {
|
|
RTC_LOG(LS_INFO) << "Removing " << cricket::MediaTypeToString(media_type)
|
|
<< " receiver for track_id=" << sender_info.sender_id
|
|
<< " and stream_id=" << sender_info.stream_id;
|
|
|
|
MediaStreamInterface* stream = remote_streams_->find(sender_info.stream_id);
|
|
|
|
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 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);
|
|
if (!params || params->id != info.sender_id ||
|
|
params->first_stream_id() != info.stream_id) {
|
|
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_id = params.first_stream_id();
|
|
const std::string& sender_id = params.id;
|
|
uint32_t ssrc = params.first_ssrc();
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(*current_senders, stream_id, sender_id);
|
|
if (!sender_info) {
|
|
current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
|
|
OnLocalSenderAdded(current_senders->back(), media_type);
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnLocalSenderAdded(const RtpSenderInfo& sender_info,
|
|
cricket::MediaType media_type) {
|
|
RTC_DCHECK(!IsUnifiedPlan());
|
|
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_ids({sender_info.stream_id});
|
|
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::OnSctpDataChannelClosed(DataChannelInterface* channel) {
|
|
// Since data_channel_controller doesn't do signals, this
|
|
// signal is relayed here.
|
|
data_channel_controller_.OnSctpDataChannelClosed(
|
|
static_cast<SctpDataChannel*>(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->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->media_type() == cricket::MEDIA_TYPE_VIDEO) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
RTC_NOTREACHED();
|
|
return nullptr;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
|
|
PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) const {
|
|
for (const 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 (const 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 (const 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_id,
|
|
const std::string sender_id) const {
|
|
for (const RtpSenderInfo& sender_info : infos) {
|
|
if (sender_info.stream_id == stream_id &&
|
|
sender_info.sender_id == sender_id) {
|
|
return &sender_info;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
SctpDataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
|
|
return data_channel_controller_.FindDataChannelBySid(sid);
|
|
}
|
|
|
|
PeerConnection::InitializePortAllocatorResult
|
|
PeerConnection::InitializePortAllocator_n(
|
|
const cricket::ServerAddresses& stun_servers,
|
|
const std::vector<cricket::RelayServerConfig>& turn_servers,
|
|
const RTCConfiguration& configuration) {
|
|
RTC_DCHECK_RUN_ON(network_thread());
|
|
|
|
port_allocator_->Initialize();
|
|
// To handle both internal and externally created port allocator, we will
|
|
// enable BUNDLE here.
|
|
int port_allocator_flags = port_allocator_->flags();
|
|
port_allocator_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) {
|
|
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
|
|
} else if (absl::StartsWith(
|
|
webrtc::field_trial::FindFullName("WebRTC-IPv6Default"),
|
|
"Disabled")) {
|
|
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
|
|
}
|
|
|
|
if (configuration.disable_ipv6_on_wifi) {
|
|
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
|
|
RTC_LOG(LS_INFO) << "IPv6 candidates on Wi-Fi are disabled.";
|
|
}
|
|
|
|
if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
|
|
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
|
|
RTC_LOG(LS_INFO) << "TCP candidates are disabled.";
|
|
}
|
|
|
|
if (configuration.candidate_network_policy ==
|
|
kCandidateNetworkPolicyLowCost) {
|
|
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS;
|
|
RTC_LOG(LS_INFO) << "Do not gather candidates on high-cost networks";
|
|
}
|
|
|
|
if (configuration.disable_link_local_networks) {
|
|
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS;
|
|
RTC_LOG(LS_INFO) << "Disable candidates on link-local network interfaces.";
|
|
}
|
|
|
|
port_allocator_->set_flags(port_allocator_flags);
|
|
// No step delay is used while allocating ports.
|
|
port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
|
|
port_allocator_->SetCandidateFilter(
|
|
ConvertIceTransportTypeToCandidateFilter(configuration.type));
|
|
port_allocator_->set_max_ipv6_networks(configuration.max_ipv6_networks);
|
|
|
|
auto turn_servers_copy = turn_servers;
|
|
for (auto& turn_server : turn_servers_copy) {
|
|
turn_server.tls_cert_verifier = tls_cert_verifier_.get();
|
|
}
|
|
// Call this last since it may create pooled allocator sessions using the
|
|
// properties set above.
|
|
port_allocator_->SetConfiguration(
|
|
stun_servers, std::move(turn_servers_copy),
|
|
configuration.ice_candidate_pool_size,
|
|
configuration.GetTurnPortPrunePolicy(), configuration.turn_customizer,
|
|
configuration.stun_candidate_keepalive_interval);
|
|
|
|
InitializePortAllocatorResult res;
|
|
res.enable_ipv6 = port_allocator_flags & cricket::PORTALLOCATOR_ENABLE_IPV6;
|
|
return res;
|
|
}
|
|
|
|
bool PeerConnection::ReconfigurePortAllocator_n(
|
|
const cricket::ServerAddresses& stun_servers,
|
|
const std::vector<cricket::RelayServerConfig>& turn_servers,
|
|
IceTransportsType type,
|
|
int candidate_pool_size,
|
|
PortPrunePolicy turn_port_prune_policy,
|
|
webrtc::TurnCustomizer* turn_customizer,
|
|
absl::optional<int> stun_candidate_keepalive_interval,
|
|
bool have_local_description) {
|
|
port_allocator_->SetCandidateFilter(
|
|
ConvertIceTransportTypeToCandidateFilter(type));
|
|
// 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.
|
|
if (have_local_description) {
|
|
port_allocator_->FreezeCandidatePool();
|
|
}
|
|
// Add the custom tls turn servers if they exist.
|
|
auto turn_servers_copy = turn_servers;
|
|
for (auto& turn_server : turn_servers_copy) {
|
|
turn_server.tls_cert_verifier = tls_cert_verifier_.get();
|
|
}
|
|
// Call this last since it may create pooled allocator sessions using the
|
|
// candidate filter set above.
|
|
return port_allocator_->SetConfiguration(
|
|
stun_servers, std::move(turn_servers_copy), candidate_pool_size,
|
|
turn_port_prune_policy, turn_customizer,
|
|
stun_candidate_keepalive_interval);
|
|
}
|
|
|
|
cricket::ChannelManager* PeerConnection::channel_manager() const {
|
|
return factory_->channel_manager();
|
|
}
|
|
|
|
bool PeerConnection::StartRtcEventLog_w(
|
|
std::unique_ptr<RtcEventLogOutput> output,
|
|
int64_t output_period_ms) {
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
if (!event_log_) {
|
|
return false;
|
|
}
|
|
return event_log_->StartLogging(std::move(output), output_period_ms);
|
|
}
|
|
|
|
void PeerConnection::StopRtcEventLog_w() {
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
if (event_log_) {
|
|
event_log_->StopLogging();
|
|
}
|
|
}
|
|
|
|
cricket::ChannelInterface* PeerConnection::GetChannel(
|
|
const std::string& content_name) {
|
|
for (const auto& transceiver : transceivers_) {
|
|
cricket::ChannelInterface* channel = transceiver->internal()->channel();
|
|
if (channel && channel->content_name() == content_name) {
|
|
return channel;
|
|
}
|
|
}
|
|
if (rtp_data_channel() &&
|
|
rtp_data_channel()->content_name() == content_name) {
|
|
return rtp_data_channel();
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
bool PeerConnection::GetSctpSslRole(rtc::SSLRole* role) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (!local_description() || !remote_description()) {
|
|
RTC_LOG(LS_VERBOSE)
|
|
<< "Local and Remote descriptions must be applied to get the "
|
|
"SSL Role of the SCTP transport.";
|
|
return false;
|
|
}
|
|
if (!data_channel_controller_.data_channel_transport()) {
|
|
RTC_LOG(LS_INFO) << "Non-rejected SCTP m= section is needed to get the "
|
|
"SSL Role of the SCTP transport.";
|
|
return false;
|
|
}
|
|
|
|
absl::optional<rtc::SSLRole> dtls_role;
|
|
if (sctp_mid_s_) {
|
|
dtls_role = transport_controller_->GetDtlsRole(*sctp_mid_s_);
|
|
if (!dtls_role && is_caller_.has_value()) {
|
|
dtls_role = *is_caller_ ? rtc::SSL_SERVER : rtc::SSL_CLIENT;
|
|
}
|
|
*role = *dtls_role;
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
bool PeerConnection::GetSslRole(const std::string& content_name,
|
|
rtc::SSLRole* role) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
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;
|
|
}
|
|
|
|
auto dtls_role = transport_controller_->GetDtlsRole(content_name);
|
|
if (dtls_role) {
|
|
*role = *dtls_role;
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
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,
|
|
const cricket::SessionDescription* description) {
|
|
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 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);
|
|
transceiver_stable_states_by_transceivers_.clear();
|
|
have_pending_rtp_data_channel_ = false;
|
|
}
|
|
|
|
// Update internal objects according to the session description's media
|
|
// descriptions.
|
|
RTCError error = PushdownMediaDescription(type, source);
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
|
|
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 (const auto& transceiver : transceivers_) {
|
|
const ContentInfo* content_info =
|
|
FindMediaSectionForTransceiver(transceiver, sdesc);
|
|
cricket::ChannelInterface* 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, error);
|
|
}
|
|
}
|
|
|
|
// If using the RtpDataChannel, push down the new SDP section for it too.
|
|
if (data_channel_controller_.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)
|
|
? data_channel_controller_.rtp_data_channel()->SetLocalContent(
|
|
data_desc, type, &error)
|
|
: data_channel_controller_.rtp_data_channel()->SetRemoteContent(
|
|
data_desc, type, &error);
|
|
if (!success) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, 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_mid_s_ && local_description() && remote_description()) {
|
|
rtc::scoped_refptr<SctpTransport> sctp_transport =
|
|
transport_controller_->GetSctpTransport(*sctp_mid_s_);
|
|
auto local_sctp_description = cricket::GetFirstSctpDataContentDescription(
|
|
local_description()->description());
|
|
auto remote_sctp_description = cricket::GetFirstSctpDataContentDescription(
|
|
remote_description()->description());
|
|
if (sctp_transport && local_sctp_description && remote_sctp_description) {
|
|
int max_message_size;
|
|
// A remote max message size of zero means "any size supported".
|
|
// We configure the connection with our own max message size.
|
|
if (remote_sctp_description->max_message_size() == 0) {
|
|
max_message_size = local_sctp_description->max_message_size();
|
|
} else {
|
|
max_message_size =
|
|
std::min(local_sctp_description->max_message_size(),
|
|
remote_sctp_description->max_message_size());
|
|
}
|
|
sctp_transport->Start(local_sctp_description->port(),
|
|
remote_sctp_description->port(), max_message_size);
|
|
}
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
RTCError PeerConnection::PushdownTransportDescription(
|
|
cricket::ContentSource source,
|
|
SdpType type) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
|
|
if (source == cricket::CS_LOCAL) {
|
|
const SessionDescriptionInterface* sdesc = local_description();
|
|
RTC_DCHECK(sdesc);
|
|
return transport_controller_->SetLocalDescription(type,
|
|
sdesc->description());
|
|
} else {
|
|
const SessionDescriptionInterface* sdesc = remote_description();
|
|
RTC_DCHECK(sdesc);
|
|
return transport_controller_->SetRemoteDescription(type,
|
|
sdesc->description());
|
|
}
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
cricket::IceConfig PeerConnection::ParseIceConfig(
|
|
const PeerConnectionInterface::RTCConfiguration& config) const {
|
|
cricket::ContinualGatheringPolicy gathering_policy;
|
|
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 = RTCConfigurationToIceConfigOptionalInt(
|
|
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 =
|
|
RTCConfigurationToIceConfigOptionalInt(
|
|
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.surface_ice_candidates_on_ice_transport_type_changed =
|
|
config.surface_ice_candidates_on_ice_transport_type_changed;
|
|
ice_config.ice_check_interval_strong_connectivity =
|
|
config.ice_check_interval_strong_connectivity;
|
|
ice_config.ice_check_interval_weak_connectivity =
|
|
config.ice_check_interval_weak_connectivity;
|
|
ice_config.ice_check_min_interval = config.ice_check_min_interval;
|
|
ice_config.ice_unwritable_timeout = config.ice_unwritable_timeout;
|
|
ice_config.ice_unwritable_min_checks = config.ice_unwritable_min_checks;
|
|
ice_config.ice_inactive_timeout = config.ice_inactive_timeout;
|
|
ice_config.stun_keepalive_interval = config.stun_candidate_keepalive_interval;
|
|
ice_config.network_preference = config.network_preference;
|
|
return ice_config;
|
|
}
|
|
|
|
std::vector<DataChannelStats> PeerConnection::GetDataChannelStats() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return data_channel_controller_.GetDataChannelStats();
|
|
}
|
|
|
|
absl::optional<std::string> PeerConnection::sctp_transport_name() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (sctp_mid_s_ && transport_controller_) {
|
|
auto dtls_transport = transport_controller_->GetDtlsTransport(*sctp_mid_s_);
|
|
if (dtls_transport) {
|
|
return dtls_transport->transport_name();
|
|
}
|
|
return absl::optional<std::string>();
|
|
}
|
|
return absl::optional<std::string>();
|
|
}
|
|
|
|
cricket::CandidateStatsList PeerConnection::GetPooledCandidateStats() const {
|
|
cricket::CandidateStatsList candidate_states_list;
|
|
network_thread()->Invoke<void>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&cricket::PortAllocator::GetCandidateStatsFromPooledSessions,
|
|
port_allocator_.get(), &candidate_states_list));
|
|
return candidate_states_list;
|
|
}
|
|
|
|
std::map<std::string, std::string> PeerConnection::GetTransportNamesByMid()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
std::map<std::string, std::string> transport_names_by_mid;
|
|
for (const auto& transceiver : transceivers_) {
|
|
cricket::ChannelInterface* channel = transceiver->internal()->channel();
|
|
if (channel) {
|
|
transport_names_by_mid[channel->content_name()] =
|
|
channel->transport_name();
|
|
}
|
|
}
|
|
if (data_channel_controller_.rtp_data_channel()) {
|
|
transport_names_by_mid[data_channel_controller_.rtp_data_channel()
|
|
->content_name()] =
|
|
data_channel_controller_.rtp_data_channel()->transport_name();
|
|
}
|
|
if (data_channel_controller_.data_channel_transport()) {
|
|
absl::optional<std::string> transport_name = sctp_transport_name();
|
|
RTC_DCHECK(transport_name);
|
|
transport_names_by_mid[*sctp_mid_s_] = *transport_name;
|
|
}
|
|
return transport_names_by_mid;
|
|
}
|
|
|
|
std::map<std::string, cricket::TransportStats>
|
|
PeerConnection::GetTransportStatsByNames(
|
|
const std::set<std::string>& transport_names) {
|
|
if (!network_thread()->IsCurrent()) {
|
|
return network_thread()
|
|
->Invoke<std::map<std::string, cricket::TransportStats>>(
|
|
RTC_FROM_HERE,
|
|
[&] { return GetTransportStatsByNames(transport_names); });
|
|
}
|
|
RTC_DCHECK_RUN_ON(network_thread());
|
|
std::map<std::string, cricket::TransportStats> transport_stats_by_name;
|
|
for (const std::string& transport_name : transport_names) {
|
|
cricket::TransportStats transport_stats;
|
|
bool success =
|
|
transport_controller_->GetStats(transport_name, &transport_stats);
|
|
if (success) {
|
|
transport_stats_by_name[transport_name] = std::move(transport_stats);
|
|
} else {
|
|
RTC_LOG(LS_ERROR) << "Failed to get transport stats for transport_name="
|
|
<< transport_name;
|
|
}
|
|
}
|
|
return transport_stats_by_name;
|
|
}
|
|
|
|
bool PeerConnection::GetLocalCertificate(
|
|
const std::string& transport_name,
|
|
rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
|
|
if (!certificate) {
|
|
return false;
|
|
}
|
|
*certificate = transport_controller_->GetLocalCertificate(transport_name);
|
|
return *certificate != nullptr;
|
|
}
|
|
|
|
std::unique_ptr<rtc::SSLCertChain> PeerConnection::GetRemoteSSLCertChain(
|
|
const std::string& transport_name) {
|
|
return transport_controller_->GetRemoteSSLCertChain(transport_name);
|
|
}
|
|
|
|
cricket::DataChannelType PeerConnection::data_channel_type() const {
|
|
return data_channel_controller_.data_channel_type();
|
|
}
|
|
|
|
bool PeerConnection::IceRestartPending(const std::string& content_name) const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
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);
|
|
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
|
|
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);
|
|
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
|
|
ReportTransportStats();
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED();
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerCandidatesGathered(
|
|
const std::string& transport_name,
|
|
const cricket::Candidates& candidates) {
|
|
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::OnTransportControllerCandidateError(
|
|
const cricket::IceCandidateErrorEvent& event) {
|
|
OnIceCandidateError(event.address, event.port, event.url, event.error_code,
|
|
event.error_text);
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerCandidatesRemoved(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
// 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::OnTransportControllerCandidateChanged(
|
|
const cricket::CandidatePairChangeEvent& event) {
|
|
OnSelectedCandidatePairChanged(event);
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerDtlsHandshakeError(
|
|
rtc::SSLHandshakeError error) {
|
|
RTC_HISTOGRAM_ENUMERATION(
|
|
"WebRTC.PeerConnection.DtlsHandshakeError", static_cast<int>(error),
|
|
static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
|
|
}
|
|
|
|
void PeerConnection::EnableSending() {
|
|
for (const auto& transceiver : transceivers_) {
|
|
cricket::ChannelInterface* channel = transceiver->internal()->channel();
|
|
if (channel && !channel->enabled()) {
|
|
channel->Enable(true);
|
|
}
|
|
}
|
|
|
|
if (data_channel_controller_.rtp_data_channel() &&
|
|
!data_channel_controller_.rtp_data_channel()->enabled()) {
|
|
data_channel_controller_.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) {
|
|
RTCErrorOr<const cricket::ContentInfo*> result =
|
|
FindContentInfo(remote_description(), candidate);
|
|
if (!result.ok()) {
|
|
RTC_LOG(LS_ERROR) << "UseCandidate: Invalid candidate. "
|
|
<< result.error().message();
|
|
return false;
|
|
}
|
|
std::vector<cricket::Candidate> candidates;
|
|
candidates.push_back(candidate->candidate());
|
|
// Invoking BaseSession method to handle remote candidates.
|
|
RTCError error = transport_controller_->AddRemoteCandidates(
|
|
result.value()->name, candidates);
|
|
if (error.ok()) {
|
|
ReportRemoteIceCandidateAdded(candidate->candidate());
|
|
// 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 {
|
|
RTC_LOG(LS_WARNING) << error.message();
|
|
}
|
|
return true;
|
|
}
|
|
|
|
RTCErrorOr<const cricket::ContentInfo*> PeerConnection::FindContentInfo(
|
|
const SessionDescriptionInterface* description,
|
|
const IceCandidateInterface* candidate) {
|
|
if (candidate->sdp_mline_index() >= 0) {
|
|
size_t mediacontent_index =
|
|
static_cast<size_t>(candidate->sdp_mline_index());
|
|
size_t content_size = description->description()->contents().size();
|
|
if (mediacontent_index < content_size) {
|
|
return &description->description()->contents()[mediacontent_index];
|
|
} else {
|
|
return RTCError(RTCErrorType::INVALID_RANGE,
|
|
"Media line index (" +
|
|
rtc::ToString(candidate->sdp_mline_index()) +
|
|
") out of range (number of mlines: " +
|
|
rtc::ToString(content_size) + ").");
|
|
}
|
|
} else if (!candidate->sdp_mid().empty()) {
|
|
auto& contents = description->description()->contents();
|
|
auto it = absl::c_find_if(
|
|
contents, [candidate](const cricket::ContentInfo& content_info) {
|
|
return content_info.mid() == candidate->sdp_mid();
|
|
});
|
|
if (it == contents.end()) {
|
|
return RTCError(
|
|
RTCErrorType::INVALID_PARAMETER,
|
|
"Mid " + candidate->sdp_mid() +
|
|
" specified but no media section with that mid found.");
|
|
} else {
|
|
return &*it;
|
|
}
|
|
}
|
|
|
|
return RTCError(RTCErrorType::INVALID_PARAMETER,
|
|
"Neither sdp_mline_index nor sdp_mid specified.");
|
|
}
|
|
|
|
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) {
|
|
DestroyDataChannelTransport();
|
|
}
|
|
}
|
|
|
|
RTCErrorOr<const cricket::ContentGroup*> PeerConnection::GetEarlyBundleGroup(
|
|
const SessionDescription& desc) const {
|
|
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");
|
|
}
|
|
}
|
|
return bundle_group;
|
|
}
|
|
|
|
RTCError PeerConnection::CreateChannels(const SessionDescription& desc) {
|
|
// Creating the media channels. Transports should already have been created
|
|
// at this point.
|
|
const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(&desc);
|
|
if (voice && !voice->rejected &&
|
|
!GetAudioTransceiver()->internal()->channel()) {
|
|
cricket::VoiceChannel* voice_channel = CreateVoiceChannel(voice->name);
|
|
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);
|
|
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 &&
|
|
!data_channel_controller_.rtp_data_channel() &&
|
|
!data_channel_controller_.data_channel_transport()) {
|
|
if (!CreateDataChannel(data->name)) {
|
|
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) {
|
|
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
|
|
|
|
cricket::VoiceChannel* voice_channel = channel_manager()->CreateVoiceChannel(
|
|
call_ptr_, configuration_.media_config, rtp_transport, signaling_thread(),
|
|
mid, SrtpRequired(), GetCryptoOptions(), &ssrc_generator_,
|
|
audio_options_);
|
|
if (!voice_channel) {
|
|
return nullptr;
|
|
}
|
|
voice_channel->SignalDtlsSrtpSetupFailure.connect(
|
|
this, &PeerConnection::OnDtlsSrtpSetupFailure);
|
|
voice_channel->SignalSentPacket.connect(this,
|
|
&PeerConnection::OnSentPacket_w);
|
|
voice_channel->SetRtpTransport(rtp_transport);
|
|
|
|
return voice_channel;
|
|
}
|
|
|
|
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
|
|
cricket::VideoChannel* PeerConnection::CreateVideoChannel(
|
|
const std::string& mid) {
|
|
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
|
|
|
|
cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel(
|
|
call_ptr_, configuration_.media_config, rtp_transport, signaling_thread(),
|
|
mid, SrtpRequired(), GetCryptoOptions(), &ssrc_generator_, video_options_,
|
|
video_bitrate_allocator_factory_.get());
|
|
if (!video_channel) {
|
|
return nullptr;
|
|
}
|
|
video_channel->SignalDtlsSrtpSetupFailure.connect(
|
|
this, &PeerConnection::OnDtlsSrtpSetupFailure);
|
|
video_channel->SignalSentPacket.connect(this,
|
|
&PeerConnection::OnSentPacket_w);
|
|
video_channel->SetRtpTransport(rtp_transport);
|
|
|
|
return video_channel;
|
|
}
|
|
|
|
bool PeerConnection::CreateDataChannel(const std::string& mid) {
|
|
switch (data_channel_type()) {
|
|
case cricket::DCT_SCTP:
|
|
if (network_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&PeerConnection::SetupDataChannelTransport_n, this,
|
|
mid))) {
|
|
sctp_mid_s_ = mid;
|
|
} else {
|
|
return false;
|
|
}
|
|
return true;
|
|
case cricket::DCT_RTP:
|
|
default:
|
|
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
|
|
// TODO(bugs.webrtc.org/9987): set_rtp_data_channel() should be called on
|
|
// the network thread like set_data_channel_transport is.
|
|
data_channel_controller_.set_rtp_data_channel(
|
|
channel_manager()->CreateRtpDataChannel(
|
|
configuration_.media_config, rtp_transport, signaling_thread(),
|
|
mid, SrtpRequired(), GetCryptoOptions(), &ssrc_generator_));
|
|
if (!data_channel_controller_.rtp_data_channel()) {
|
|
return false;
|
|
}
|
|
data_channel_controller_.rtp_data_channel()
|
|
->SignalDtlsSrtpSetupFailure.connect(
|
|
this, &PeerConnection::OnDtlsSrtpSetupFailure);
|
|
data_channel_controller_.rtp_data_channel()->SignalSentPacket.connect(
|
|
this, &PeerConnection::OnSentPacket_w);
|
|
data_channel_controller_.rtp_data_channel()->SetRtpTransport(
|
|
rtp_transport);
|
|
have_pending_rtp_data_channel_ = true;
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
Call::Stats PeerConnection::GetCallStats() {
|
|
if (!worker_thread()->IsCurrent()) {
|
|
return worker_thread()->Invoke<Call::Stats>(
|
|
RTC_FROM_HERE, rtc::Bind(&PeerConnection::GetCallStats, this));
|
|
}
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
|
|
if (call_) {
|
|
return call_->GetStats();
|
|
} else {
|
|
return Call::Stats();
|
|
}
|
|
}
|
|
|
|
bool PeerConnection::SetupDataChannelTransport_n(const std::string& mid) {
|
|
DataChannelTransportInterface* transport =
|
|
transport_controller_->GetDataChannelTransport(mid);
|
|
if (!transport) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "Data channel transport is not available for data channels, mid="
|
|
<< mid;
|
|
return false;
|
|
}
|
|
RTC_LOG(LS_INFO) << "Setting up data channel transport for mid=" << mid;
|
|
|
|
data_channel_controller_.set_data_channel_transport(transport);
|
|
data_channel_controller_.SetupDataChannelTransport_n();
|
|
sctp_mid_n_ = mid;
|
|
|
|
// Note: setting the data sink and checking initial state must be done last,
|
|
// after setting up the data channel. Setting the data sink may trigger
|
|
// callbacks to PeerConnection which require the transport to be completely
|
|
// set up (eg. OnReadyToSend()).
|
|
transport->SetDataSink(&data_channel_controller_);
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::TeardownDataChannelTransport_n() {
|
|
if (!sctp_mid_n_ && !data_channel_controller_.data_channel_transport()) {
|
|
return;
|
|
}
|
|
RTC_LOG(LS_INFO) << "Tearing down data channel transport for mid="
|
|
<< *sctp_mid_n_;
|
|
|
|
// |sctp_mid_| may still be active through an SCTP transport. If not, unset
|
|
// it.
|
|
sctp_mid_n_.reset();
|
|
data_channel_controller_.TeardownDataChannelTransport_n();
|
|
}
|
|
|
|
// 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();
|
|
}
|
|
|
|
static RTCError ValidateMids(const cricket::SessionDescription& description) {
|
|
std::set<std::string> mids;
|
|
for (const cricket::ContentInfo& content : description.contents()) {
|
|
if (content.name.empty()) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"A media section is missing a MID attribute.");
|
|
}
|
|
if (!mids.insert(content.name).second) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"Duplicate a=mid value '" + content.name + "'.");
|
|
}
|
|
}
|
|
return RTCError::OK();
|
|
}
|
|
|
|
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_STATE,
|
|
"Called in wrong state: " + GetSignalingStateString(signaling_state()));
|
|
}
|
|
|
|
RTCError error = ValidateMids(*sdesc->description());
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
|
|
// 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) {
|
|
// With an answer we want to compare the new answer session description with
|
|
// the offer's session description from the current negotiation.
|
|
const cricket::SessionDescription* offer_desc =
|
|
(source == cricket::CS_LOCAL) ? remote_description()->description()
|
|
: local_description()->description();
|
|
if (!MediaSectionsHaveSameCount(*offer_desc, *sdesc->description()) ||
|
|
!MediaSectionsInSameOrder(*offer_desc, nullptr, *sdesc->description(),
|
|
type)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
kMlineMismatchInAnswer);
|
|
}
|
|
} else {
|
|
// The re-offers should respect the order of m= sections in current
|
|
// description. See RFC3264 Section 8 paragraph 4 for more details.
|
|
// With a re-offer, either the current local or current remote descriptions
|
|
// could be the most up to date, so we would like to check against both of
|
|
// them if they exist. It could be the case that one of them has a 0 port
|
|
// for a media section, but the other does not. This is important to check
|
|
// against in the case that we are recycling an m= section.
|
|
const cricket::SessionDescription* current_desc = nullptr;
|
|
const cricket::SessionDescription* secondary_current_desc = nullptr;
|
|
if (local_description()) {
|
|
current_desc = local_description()->description();
|
|
if (remote_description()) {
|
|
secondary_current_desc = remote_description()->description();
|
|
}
|
|
} else if (remote_description()) {
|
|
current_desc = remote_description()->description();
|
|
}
|
|
if (current_desc &&
|
|
!MediaSectionsInSameOrder(*current_desc, secondary_current_desc,
|
|
*sdesc->description(), type)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
kMlineMismatchInSubsequentOffer);
|
|
}
|
|
}
|
|
|
|
if (IsUnifiedPlan()) {
|
|
// Ensure that each audio and video media section has at most one
|
|
// "StreamParams". This will return an error if receiving a session
|
|
// description from a "Plan B" endpoint which adds multiple tracks of the
|
|
// same type. With Unified Plan, there can only be at most one track per
|
|
// media section.
|
|
for (const ContentInfo& content : sdesc->description()->contents()) {
|
|
const MediaContentDescription& desc = *content.media_description();
|
|
if ((desc.type() == cricket::MEDIA_TYPE_AUDIO ||
|
|
desc.type() == cricket::MEDIA_TYPE_VIDEO) &&
|
|
desc.streams().size() > 1u) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"Media section has more than one track specified "
|
|
"with a=ssrc lines which is not supported with "
|
|
"Unified Plan.");
|
|
}
|
|
}
|
|
}
|
|
|
|
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() {
|
|
rtc::StringBuilder desc;
|
|
desc << kSessionError << SessionErrorToString(session_error()) << ". ";
|
|
desc << kSessionErrorDesc << session_error_desc() << ".";
|
|
return desc.Release();
|
|
}
|
|
|
|
void PeerConnection::ReportSdpFormatReceived(
|
|
const SessionDescriptionInterface& remote_offer) {
|
|
int num_audio_mlines = 0;
|
|
int num_video_mlines = 0;
|
|
int num_audio_tracks = 0;
|
|
int num_video_tracks = 0;
|
|
for (const ContentInfo& content : remote_offer.description()->contents()) {
|
|
cricket::MediaType media_type = content.media_description()->type();
|
|
int num_tracks = std::max(
|
|
1, static_cast<int>(content.media_description()->streams().size()));
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
|
num_audio_mlines += 1;
|
|
num_audio_tracks += num_tracks;
|
|
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
|
|
num_video_mlines += 1;
|
|
num_video_tracks += num_tracks;
|
|
}
|
|
}
|
|
SdpFormatReceived format = kSdpFormatReceivedNoTracks;
|
|
if (num_audio_mlines > 1 || num_video_mlines > 1) {
|
|
format = kSdpFormatReceivedComplexUnifiedPlan;
|
|
} else if (num_audio_tracks > 1 || num_video_tracks > 1) {
|
|
format = kSdpFormatReceivedComplexPlanB;
|
|
} else if (num_audio_tracks > 0 || num_video_tracks > 0) {
|
|
format = kSdpFormatReceivedSimple;
|
|
}
|
|
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpFormatReceived", format,
|
|
kSdpFormatReceivedMax);
|
|
}
|
|
|
|
void PeerConnection::ReportIceCandidateCollected(
|
|
const cricket::Candidate& candidate) {
|
|
NoteUsageEvent(UsageEvent::CANDIDATE_COLLECTED);
|
|
if (candidate.address().IsPrivateIP()) {
|
|
NoteUsageEvent(UsageEvent::PRIVATE_CANDIDATE_COLLECTED);
|
|
}
|
|
if (candidate.address().IsUnresolvedIP()) {
|
|
NoteUsageEvent(UsageEvent::MDNS_CANDIDATE_COLLECTED);
|
|
}
|
|
if (candidate.address().family() == AF_INET6) {
|
|
NoteUsageEvent(UsageEvent::IPV6_CANDIDATE_COLLECTED);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::ReportRemoteIceCandidateAdded(
|
|
const cricket::Candidate& candidate) {
|
|
NoteUsageEvent(UsageEvent::REMOTE_CANDIDATE_ADDED);
|
|
if (candidate.address().IsPrivateIP()) {
|
|
NoteUsageEvent(UsageEvent::REMOTE_PRIVATE_CANDIDATE_ADDED);
|
|
}
|
|
if (candidate.address().IsUnresolvedIP()) {
|
|
NoteUsageEvent(UsageEvent::REMOTE_MDNS_CANDIDATE_ADDED);
|
|
}
|
|
if (candidate.address().family() == AF_INET6) {
|
|
NoteUsageEvent(UsageEvent::REMOTE_IPV6_CANDIDATE_ADDED);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::NoteUsageEvent(UsageEvent event) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
usage_event_accumulator_ |= static_cast<int>(event);
|
|
}
|
|
|
|
void PeerConnection::ReportUsagePattern() const {
|
|
RTC_DLOG(LS_INFO) << "Usage signature is " << usage_event_accumulator_;
|
|
RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.PeerConnection.UsagePattern",
|
|
usage_event_accumulator_,
|
|
static_cast<int>(UsageEvent::MAX_VALUE));
|
|
const int bad_bits =
|
|
static_cast<int>(UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED) |
|
|
static_cast<int>(UsageEvent::CANDIDATE_COLLECTED);
|
|
const int good_bits =
|
|
static_cast<int>(UsageEvent::SET_REMOTE_DESCRIPTION_SUCCEEDED) |
|
|
static_cast<int>(UsageEvent::REMOTE_CANDIDATE_ADDED) |
|
|
static_cast<int>(UsageEvent::ICE_STATE_CONNECTED);
|
|
if ((usage_event_accumulator_ & bad_bits) == bad_bits &&
|
|
(usage_event_accumulator_ & good_bits) == 0) {
|
|
// If called after close(), we can't report, because observer may have
|
|
// been deallocated, and therefore pointer is null. Write to log instead.
|
|
if (observer_) {
|
|
Observer()->OnInterestingUsage(usage_event_accumulator_);
|
|
} else {
|
|
RTC_LOG(LS_INFO) << "Interesting usage signature "
|
|
<< usage_event_accumulator_
|
|
<< " observed after observer shutdown";
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::ReportNegotiatedSdpSemantics(
|
|
const SessionDescriptionInterface& answer) {
|
|
SdpSemanticNegotiated semantics_negotiated;
|
|
switch (answer.description()->msid_signaling()) {
|
|
case 0:
|
|
semantics_negotiated = kSdpSemanticNegotiatedNone;
|
|
break;
|
|
case cricket::kMsidSignalingMediaSection:
|
|
semantics_negotiated = kSdpSemanticNegotiatedUnifiedPlan;
|
|
break;
|
|
case cricket::kMsidSignalingSsrcAttribute:
|
|
semantics_negotiated = kSdpSemanticNegotiatedPlanB;
|
|
break;
|
|
case cricket::kMsidSignalingMediaSection |
|
|
cricket::kMsidSignalingSsrcAttribute:
|
|
semantics_negotiated = kSdpSemanticNegotiatedMixed;
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED();
|
|
}
|
|
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpSemanticNegotiated",
|
|
semantics_negotiated, kSdpSemanticNegotiatedMax);
|
|
}
|
|
|
|
// 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;
|
|
}
|
|
|
|
RTCErrorOr<const cricket::ContentInfo*> result =
|
|
FindContentInfo(current_remote_desc, candidate);
|
|
if (!result.ok()) {
|
|
RTC_LOG(LS_ERROR) << "ReadyToUseRemoteCandidate: Invalid candidate. "
|
|
<< result.error().message();
|
|
|
|
*valid = false;
|
|
return false;
|
|
}
|
|
|
|
std::string transport_name = GetTransportName(result.value()->name);
|
|
return !transport_name.empty();
|
|
}
|
|
|
|
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() {
|
|
std::map<std::string, std::set<cricket::MediaType>>
|
|
media_types_by_transport_name;
|
|
for (const auto& transceiver : transceivers_) {
|
|
if (transceiver->internal()->channel()) {
|
|
const std::string& transport_name =
|
|
transceiver->internal()->channel()->transport_name();
|
|
media_types_by_transport_name[transport_name].insert(
|
|
transceiver->media_type());
|
|
}
|
|
}
|
|
if (rtp_data_channel()) {
|
|
media_types_by_transport_name[rtp_data_channel()->transport_name()].insert(
|
|
cricket::MEDIA_TYPE_DATA);
|
|
}
|
|
|
|
absl::optional<std::string> transport_name = sctp_transport_name();
|
|
if (transport_name) {
|
|
media_types_by_transport_name[*transport_name].insert(
|
|
cricket::MEDIA_TYPE_DATA);
|
|
}
|
|
|
|
for (const auto& entry : media_types_by_transport_name) {
|
|
const std::string& transport_name = entry.first;
|
|
const std::set<cricket::MediaType> media_types = entry.second;
|
|
cricket::TransportStats stats;
|
|
if (transport_controller_->GetStats(transport_name, &stats)) {
|
|
ReportBestConnectionState(stats);
|
|
ReportNegotiatedCiphers(stats, media_types);
|
|
}
|
|
}
|
|
}
|
|
// Walk through the ConnectionInfos to gather best connection usage
|
|
// for IPv4 and IPv6.
|
|
void PeerConnection::ReportBestConnectionState(
|
|
const cricket::TransportStats& stats) {
|
|
for (const cricket::TransportChannelStats& channel_stats :
|
|
stats.channel_stats) {
|
|
for (const cricket::ConnectionInfo& connection_info :
|
|
channel_stats.ice_transport_stats.connection_infos) {
|
|
if (!connection_info.best_connection) {
|
|
continue;
|
|
}
|
|
|
|
const cricket::Candidate& local = connection_info.local_candidate;
|
|
const cricket::Candidate& remote = connection_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)) {
|
|
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_TCP",
|
|
GetIceCandidatePairCounter(local, remote),
|
|
kIceCandidatePairMax);
|
|
} else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
|
|
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_UDP",
|
|
GetIceCandidatePairCounter(local, remote),
|
|
kIceCandidatePairMax);
|
|
} else {
|
|
RTC_CHECK(0);
|
|
}
|
|
|
|
// Increment the counter for IP type.
|
|
if (local.address().family() == AF_INET) {
|
|
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
|
|
kBestConnections_IPv4,
|
|
kPeerConnectionAddressFamilyCounter_Max);
|
|
} else if (local.address().family() == AF_INET6) {
|
|
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
|
|
kBestConnections_IPv6,
|
|
kPeerConnectionAddressFamilyCounter_Max);
|
|
} else {
|
|
RTC_CHECK(!local.address().hostname().empty() &&
|
|
local.address().IsUnresolvedIP());
|
|
}
|
|
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::ReportNegotiatedCiphers(
|
|
const cricket::TransportStats& stats,
|
|
const std::set<cricket::MediaType>& media_types) {
|
|
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;
|
|
}
|
|
|
|
if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE) {
|
|
for (cricket::MediaType media_type : media_types) {
|
|
switch (media_type) {
|
|
case cricket::MEDIA_TYPE_AUDIO:
|
|
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
|
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio", srtp_crypto_suite,
|
|
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
|
|
break;
|
|
case cricket::MEDIA_TYPE_VIDEO:
|
|
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
|
"WebRTC.PeerConnection.SrtpCryptoSuite.Video", srtp_crypto_suite,
|
|
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
|
|
break;
|
|
case cricket::MEDIA_TYPE_DATA:
|
|
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
|
"WebRTC.PeerConnection.SrtpCryptoSuite.Data", srtp_crypto_suite,
|
|
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED();
|
|
continue;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
|
|
for (cricket::MediaType media_type : media_types) {
|
|
switch (media_type) {
|
|
case cricket::MEDIA_TYPE_AUDIO:
|
|
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
|
"WebRTC.PeerConnection.SslCipherSuite.Audio", ssl_cipher_suite,
|
|
rtc::SSL_CIPHER_SUITE_MAX_VALUE);
|
|
break;
|
|
case cricket::MEDIA_TYPE_VIDEO:
|
|
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
|
"WebRTC.PeerConnection.SslCipherSuite.Video", ssl_cipher_suite,
|
|
rtc::SSL_CIPHER_SUITE_MAX_VALUE);
|
|
break;
|
|
case cricket::MEDIA_TYPE_DATA:
|
|
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
|
"WebRTC.PeerConnection.SslCipherSuite.Data", ssl_cipher_suite,
|
|
rtc::SSL_CIPHER_SUITE_MAX_VALUE);
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED();
|
|
continue;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
RTC_DCHECK(call_);
|
|
call_->OnSentPacket(sent_packet);
|
|
}
|
|
|
|
const std::string PeerConnection::GetTransportName(
|
|
const std::string& content_name) {
|
|
cricket::ChannelInterface* channel = GetChannel(content_name);
|
|
if (channel) {
|
|
return channel->transport_name();
|
|
}
|
|
if (data_channel_controller_.data_channel_transport()) {
|
|
RTC_DCHECK(sctp_mid_s_);
|
|
if (content_name == *sctp_mid_s_) {
|
|
return *sctp_transport_name();
|
|
}
|
|
}
|
|
// Return an empty string if failed to retrieve the transport name.
|
|
return "";
|
|
}
|
|
|
|
void PeerConnection::DestroyTransceiverChannel(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver) {
|
|
RTC_DCHECK(transceiver);
|
|
|
|
cricket::ChannelInterface* channel = transceiver->internal()->channel();
|
|
if (channel) {
|
|
transceiver->internal()->SetChannel(nullptr);
|
|
DestroyChannelInterface(channel);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::DestroyDataChannelTransport() {
|
|
if (data_channel_controller_.rtp_data_channel()) {
|
|
data_channel_controller_.OnTransportChannelClosed();
|
|
DestroyChannelInterface(data_channel_controller_.rtp_data_channel());
|
|
data_channel_controller_.set_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_mid_s_) {
|
|
data_channel_controller_.OnTransportChannelClosed();
|
|
network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
|
|
RTC_DCHECK_RUN_ON(network_thread());
|
|
TeardownDataChannelTransport_n();
|
|
});
|
|
sctp_mid_s_.reset();
|
|
}
|
|
}
|
|
|
|
void PeerConnection::DestroyChannelInterface(
|
|
cricket::ChannelInterface* channel) {
|
|
RTC_DCHECK(channel);
|
|
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;
|
|
}
|
|
}
|
|
|
|
bool PeerConnection::OnTransportChanged(
|
|
const std::string& mid,
|
|
RtpTransportInternal* rtp_transport,
|
|
rtc::scoped_refptr<DtlsTransport> dtls_transport,
|
|
DataChannelTransportInterface* data_channel_transport) {
|
|
RTC_DCHECK_RUN_ON(network_thread());
|
|
bool ret = true;
|
|
auto base_channel = GetChannel(mid);
|
|
if (base_channel) {
|
|
ret = base_channel->SetRtpTransport(rtp_transport);
|
|
}
|
|
if (mid == sctp_mid_n_) {
|
|
data_channel_controller_.OnTransportChanged(data_channel_transport);
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
void PeerConnection::OnSetStreams() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (IsUnifiedPlan())
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
PeerConnectionObserver* PeerConnection::Observer() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(observer_);
|
|
return observer_;
|
|
}
|
|
|
|
CryptoOptions PeerConnection::GetCryptoOptions() {
|
|
// TODO(bugs.webrtc.org/9891) - Remove PeerConnectionFactory::CryptoOptions
|
|
// after it has been removed.
|
|
return configuration_.crypto_options.has_value()
|
|
? *configuration_.crypto_options
|
|
: factory_->options().crypto_options;
|
|
}
|
|
|
|
void PeerConnection::ClearStatsCache() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (stats_collector_) {
|
|
stats_collector_->ClearCachedStatsReport();
|
|
}
|
|
}
|
|
|
|
void PeerConnection::RequestUsagePatternReportForTesting() {
|
|
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_REPORT_USAGE_PATTERN,
|
|
nullptr);
|
|
}
|
|
|
|
void PeerConnection::UpdateNegotiationNeeded() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (!IsUnifiedPlan()) {
|
|
Observer()->OnRenegotiationNeeded();
|
|
return;
|
|
}
|
|
|
|
// If connection's [[IsClosed]] slot is true, abort these steps.
|
|
if (IsClosed())
|
|
return;
|
|
|
|
// If connection's signaling state is not "stable", abort these steps.
|
|
if (signaling_state() != kStable)
|
|
return;
|
|
|
|
// NOTE
|
|
// The negotiation-needed flag will be updated once the state transitions to
|
|
// "stable", as part of the steps for setting an RTCSessionDescription.
|
|
|
|
// If the result of checking if negotiation is needed is false, clear the
|
|
// negotiation-needed flag by setting connection's [[NegotiationNeeded]] slot
|
|
// to false, and abort these steps.
|
|
bool is_negotiation_needed = CheckIfNegotiationIsNeeded();
|
|
if (!is_negotiation_needed) {
|
|
is_negotiation_needed_ = false;
|
|
return;
|
|
}
|
|
|
|
// If connection's [[NegotiationNeeded]] slot is already true, abort these
|
|
// steps.
|
|
if (is_negotiation_needed_)
|
|
return;
|
|
|
|
// Set connection's [[NegotiationNeeded]] slot to true.
|
|
is_negotiation_needed_ = true;
|
|
|
|
// Queue a task that runs the following steps:
|
|
// If connection's [[IsClosed]] slot is true, abort these steps.
|
|
// If connection's [[NegotiationNeeded]] slot is false, abort these steps.
|
|
// Fire an event named negotiationneeded at connection.
|
|
Observer()->OnRenegotiationNeeded();
|
|
}
|
|
|
|
bool PeerConnection::CheckIfNegotiationIsNeeded() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// 1. If any implementation-specific negotiation is required, as described at
|
|
// the start of this section, return true.
|
|
|
|
// 2. If connection's [[RestartIce]] internal slot is true, return true.
|
|
if (local_ice_credentials_to_replace_->HasIceCredentials()) {
|
|
return true;
|
|
}
|
|
|
|
// 3. Let description be connection.[[CurrentLocalDescription]].
|
|
const SessionDescriptionInterface* description = current_local_description();
|
|
if (!description)
|
|
return true;
|
|
|
|
// 4. If connection has created any RTCDataChannels, and no m= section in
|
|
// description has been negotiated yet for data, return true.
|
|
if (data_channel_controller_.HasSctpDataChannels()) {
|
|
if (!cricket::GetFirstDataContent(description->description()->contents()))
|
|
return true;
|
|
}
|
|
|
|
// 5. For each transceiver in connection's set of transceivers, perform the
|
|
// following checks:
|
|
for (const auto& transceiver : transceivers_) {
|
|
const ContentInfo* current_local_msection =
|
|
FindTransceiverMSection(transceiver.get(), description);
|
|
|
|
const ContentInfo* current_remote_msection = FindTransceiverMSection(
|
|
transceiver.get(), current_remote_description());
|
|
|
|
// 5.3 If transceiver is stopped and is associated with an m= section,
|
|
// but the associated m= section is not yet rejected in
|
|
// connection.[[CurrentLocalDescription]] or
|
|
// connection.[[CurrentRemoteDescription]], return true.
|
|
if (transceiver->stopped()) {
|
|
if (current_local_msection && !current_local_msection->rejected &&
|
|
((current_remote_msection && !current_remote_msection->rejected) ||
|
|
!current_remote_msection)) {
|
|
return true;
|
|
}
|
|
continue;
|
|
}
|
|
|
|
// 5.1 If transceiver isn't stopped and isn't yet associated with an m=
|
|
// section in description, return true.
|
|
if (!current_local_msection)
|
|
return true;
|
|
|
|
const MediaContentDescription* current_local_media_description =
|
|
current_local_msection->media_description();
|
|
// 5.2 If transceiver isn't stopped and is associated with an m= section
|
|
// in description then perform the following checks:
|
|
|
|
// 5.2.1 If transceiver.[[Direction]] is "sendrecv" or "sendonly", and the
|
|
// associated m= section in description either doesn't contain a single
|
|
// "a=msid" line, or the number of MSIDs from the "a=msid" lines in this
|
|
// m= section, or the MSID values themselves, differ from what is in
|
|
// transceiver.sender.[[AssociatedMediaStreamIds]], return true.
|
|
if (RtpTransceiverDirectionHasSend(transceiver->direction())) {
|
|
if (current_local_media_description->streams().size() == 0)
|
|
return true;
|
|
|
|
std::vector<std::string> msection_msids;
|
|
for (const auto& stream : current_local_media_description->streams()) {
|
|
for (const std::string& msid : stream.stream_ids())
|
|
msection_msids.push_back(msid);
|
|
}
|
|
|
|
std::vector<std::string> transceiver_msids =
|
|
transceiver->sender()->stream_ids();
|
|
if (msection_msids.size() != transceiver_msids.size())
|
|
return true;
|
|
|
|
absl::c_sort(transceiver_msids);
|
|
absl::c_sort(msection_msids);
|
|
if (transceiver_msids != msection_msids)
|
|
return true;
|
|
}
|
|
|
|
// 5.2.2 If description is of type "offer", and the direction of the
|
|
// associated m= section in neither connection.[[CurrentLocalDescription]]
|
|
// nor connection.[[CurrentRemoteDescription]] matches
|
|
// transceiver.[[Direction]], return true.
|
|
if (description->GetType() == SdpType::kOffer) {
|
|
if (!current_remote_description())
|
|
return true;
|
|
|
|
if (!current_remote_msection)
|
|
return true;
|
|
|
|
RtpTransceiverDirection current_local_direction =
|
|
current_local_media_description->direction();
|
|
RtpTransceiverDirection current_remote_direction =
|
|
current_remote_msection->media_description()->direction();
|
|
if (transceiver->direction() != current_local_direction &&
|
|
transceiver->direction() !=
|
|
RtpTransceiverDirectionReversed(current_remote_direction)) {
|
|
return true;
|
|
}
|
|
}
|
|
|
|
// 5.2.3 If description is of type "answer", and the direction of the
|
|
// associated m= section in the description does not match
|
|
// transceiver.[[Direction]] intersected with the offered direction (as
|
|
// described in [JSEP] (section 5.3.1.)), return true.
|
|
if (description->GetType() == SdpType::kAnswer) {
|
|
if (!remote_description())
|
|
return true;
|
|
|
|
const ContentInfo* offered_remote_msection =
|
|
FindTransceiverMSection(transceiver.get(), remote_description());
|
|
|
|
RtpTransceiverDirection offered_direction =
|
|
offered_remote_msection
|
|
? offered_remote_msection->media_description()->direction()
|
|
: RtpTransceiverDirection::kInactive;
|
|
|
|
if (current_local_media_description->direction() !=
|
|
(RtpTransceiverDirectionIntersection(
|
|
transceiver->direction(),
|
|
RtpTransceiverDirectionReversed(offered_direction)))) {
|
|
return true;
|
|
}
|
|
}
|
|
}
|
|
|
|
// If all the preceding checks were performed and true was not returned,
|
|
// nothing remains to be negotiated; return false.
|
|
return false;
|
|
}
|
|
|
|
RTCError PeerConnection::Rollback(SdpType sdp_type) {
|
|
auto state = signaling_state();
|
|
if (state != PeerConnectionInterface::kHaveLocalOffer &&
|
|
state != PeerConnectionInterface::kHaveRemoteOffer) {
|
|
return RTCError(RTCErrorType::INVALID_STATE,
|
|
"Called in wrong signalingState: " +
|
|
GetSignalingStateString(signaling_state()));
|
|
}
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> all_added_streams;
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> all_removed_streams;
|
|
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> removed_receivers;
|
|
|
|
for (auto&& transceivers_stable_state_pair :
|
|
transceiver_stable_states_by_transceivers_) {
|
|
auto transceiver = transceivers_stable_state_pair.first;
|
|
auto state = transceivers_stable_state_pair.second;
|
|
|
|
if (state.remote_stream_ids()) {
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> added_streams;
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
|
|
SetAssociatedRemoteStreams(transceiver->internal()->receiver_internal(),
|
|
state.remote_stream_ids().value(),
|
|
&added_streams, &removed_streams);
|
|
all_added_streams.insert(all_added_streams.end(), added_streams.begin(),
|
|
added_streams.end());
|
|
all_removed_streams.insert(all_removed_streams.end(),
|
|
removed_streams.begin(),
|
|
removed_streams.end());
|
|
if (!state.has_m_section() && !state.newly_created()) {
|
|
continue;
|
|
}
|
|
}
|
|
|
|
RTC_DCHECK(transceiver->internal()->mid().has_value());
|
|
DestroyTransceiverChannel(transceiver);
|
|
|
|
if (signaling_state() == PeerConnectionInterface::kHaveRemoteOffer &&
|
|
transceiver->receiver()) {
|
|
removed_receivers.push_back(transceiver->receiver());
|
|
}
|
|
if (state.newly_created()) {
|
|
if (transceiver->internal()->reused_for_addtrack()) {
|
|
transceiver->internal()->set_created_by_addtrack(true);
|
|
} else {
|
|
int remaining_transceiver_count = 0;
|
|
for (auto&& t : transceivers_) {
|
|
if (t != transceiver) {
|
|
transceivers_[remaining_transceiver_count++] = t;
|
|
}
|
|
}
|
|
transceivers_.resize(remaining_transceiver_count);
|
|
}
|
|
}
|
|
transceiver->internal()->sender_internal()->set_transport(nullptr);
|
|
transceiver->internal()->receiver_internal()->set_transport(nullptr);
|
|
transceiver->internal()->set_mid(state.mid());
|
|
transceiver->internal()->set_mline_index(state.mline_index());
|
|
}
|
|
transport_controller_->RollbackTransports();
|
|
if (have_pending_rtp_data_channel_) {
|
|
DestroyDataChannelTransport();
|
|
have_pending_rtp_data_channel_ = false;
|
|
}
|
|
transceiver_stable_states_by_transceivers_.clear();
|
|
pending_local_description_.reset();
|
|
pending_remote_description_.reset();
|
|
ChangeSignalingState(PeerConnectionInterface::kStable);
|
|
|
|
// Once all processing has finished, fire off callbacks.
|
|
for (const auto& receiver : removed_receivers) {
|
|
Observer()->OnRemoveTrack(receiver);
|
|
}
|
|
for (const auto& stream : all_added_streams) {
|
|
Observer()->OnAddStream(stream);
|
|
}
|
|
for (const auto& stream : all_removed_streams) {
|
|
Observer()->OnRemoveStream(stream);
|
|
}
|
|
|
|
// The assumption is that in case of implicit rollback UpdateNegotiationNeeded
|
|
// gets called in SetRemoteDescription.
|
|
if (sdp_type == SdpType::kRollback) {
|
|
UpdateNegotiationNeeded();
|
|
if (is_negotiation_needed_) {
|
|
Observer()->OnRenegotiationNeeded();
|
|
}
|
|
}
|
|
return RTCError::OK();
|
|
}
|
|
|
|
} // namespace webrtc
|