| /* |
| * Copyright 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "pc/peer_connection.h" |
| |
| #include <algorithm> |
| #include <limits> |
| #include <queue> |
| #include <set> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/memory/memory.h" |
| #include "absl/strings/match.h" |
| #include "api/jsep_ice_candidate.h" |
| #include "api/jsep_session_description.h" |
| #include "api/media_stream_proxy.h" |
| #include "api/media_stream_track_proxy.h" |
| #include "api/uma_metrics.h" |
| #include "call/call.h" |
| #include "logging/rtc_event_log/ice_logger.h" |
| #include "logging/rtc_event_log/output/rtc_event_log_output_file.h" |
| #include "logging/rtc_event_log/rtc_event_log.h" |
| #include "media/sctp/sctp_transport.h" |
| #include "pc/audio_track.h" |
| #include "pc/channel.h" |
| #include "pc/channel_manager.h" |
| #include "pc/dtmf_sender.h" |
| #include "pc/media_stream.h" |
| #include "pc/media_stream_observer.h" |
| #include "pc/remote_audio_source.h" |
| #include "pc/rtp_media_utils.h" |
| #include "pc/rtp_receiver.h" |
| #include "pc/rtp_sender.h" |
| #include "pc/sctp_utils.h" |
| #include "pc/sdp_utils.h" |
| #include "pc/stream_collection.h" |
| #include "pc/video_capturer_track_source.h" |
| #include "pc/video_track.h" |
| #include "rtc_base/bind.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/string_encode.h" |
| #include "rtc_base/strings/string_builder.h" |
| #include "rtc_base/trace_event.h" |
| #include "system_wrappers/include/clock.h" |
| #include "system_wrappers/include/field_trial.h" |
| #include "system_wrappers/include/metrics.h" |
| |
| using cricket::ContentInfo; |
| using cricket::ContentInfos; |
| using cricket::MediaContentDescription; |
| using cricket::MediaProtocolType; |
| using cricket::SessionDescription; |
| using cricket::SimulcastLayerList; |
| using cricket::TransportInfo; |
| |
| using cricket::LOCAL_PORT_TYPE; |
| using cricket::STUN_PORT_TYPE; |
| using cricket::RELAY_PORT_TYPE; |
| using cricket::PRFLX_PORT_TYPE; |
| |
| namespace webrtc { |
| |
| // Error messages |
| const char kBundleWithoutRtcpMux[] = |
| "rtcp-mux must be enabled when BUNDLE " |
| "is enabled."; |
| const char kInvalidCandidates[] = "Description contains invalid candidates."; |
| const char kInvalidSdp[] = "Invalid session description."; |
| const char kMlineMismatchInAnswer[] = |
| "The order of m-lines in answer doesn't match order in offer. Rejecting " |
| "answer."; |
| const char kMlineMismatchInSubsequentOffer[] = |
| "The order of m-lines in subsequent offer doesn't match order from " |
| "previous offer/answer."; |
| const char kSdpWithoutDtlsFingerprint[] = |
| "Called with SDP without DTLS fingerprint."; |
| const char kSdpWithoutSdesCrypto[] = "Called with SDP without SDES crypto."; |
| const char kSdpWithoutIceUfragPwd[] = |
| "Called with SDP without ice-ufrag and ice-pwd."; |
| const char kSessionError[] = "Session error code: "; |
| const char kSessionErrorDesc[] = "Session error description: "; |
| const char kDtlsSrtpSetupFailureRtp[] = |
| "Couldn't set up DTLS-SRTP on RTP channel."; |
| const char kDtlsSrtpSetupFailureRtcp[] = |
| "Couldn't set up DTLS-SRTP on RTCP channel."; |
| |
| namespace { |
| |
| static const char kDefaultStreamId[] = "default"; |
| static const char kDefaultAudioSenderId[] = "defaulta0"; |
| static const char kDefaultVideoSenderId[] = "defaultv0"; |
| |
| // The length of RTCP CNAMEs. |
| static const int kRtcpCnameLength = 16; |
| |
| enum { |
| MSG_SET_SESSIONDESCRIPTION_SUCCESS = 0, |
| MSG_SET_SESSIONDESCRIPTION_FAILED, |
| MSG_CREATE_SESSIONDESCRIPTION_FAILED, |
| MSG_GETSTATS, |
| MSG_FREE_DATACHANNELS, |
| MSG_REPORT_USAGE_PATTERN, |
| }; |
| |
| static const int REPORT_USAGE_PATTERN_DELAY_MS = 60000; |
| |
| struct SetSessionDescriptionMsg : public rtc::MessageData { |
| explicit SetSessionDescriptionMsg( |
| webrtc::SetSessionDescriptionObserver* observer) |
| : observer(observer) {} |
| |
| rtc::scoped_refptr<webrtc::SetSessionDescriptionObserver> observer; |
| RTCError error; |
| }; |
| |
| struct CreateSessionDescriptionMsg : public rtc::MessageData { |
| explicit CreateSessionDescriptionMsg( |
| webrtc::CreateSessionDescriptionObserver* observer) |
| : observer(observer) {} |
| |
| rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer; |
| RTCError error; |
| }; |
| |
| struct GetStatsMsg : public rtc::MessageData { |
| GetStatsMsg(webrtc::StatsObserver* observer, |
| webrtc::MediaStreamTrackInterface* track) |
| : observer(observer), track(track) {} |
| rtc::scoped_refptr<webrtc::StatsObserver> observer; |
| rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track; |
| }; |
| |
| // Check if we can send |new_stream| on a PeerConnection. |
| bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams, |
| webrtc::MediaStreamInterface* new_stream) { |
| if (!new_stream || !current_streams) { |
| return false; |
| } |
| if (current_streams->find(new_stream->id()) != nullptr) { |
| RTC_LOG(LS_ERROR) << "MediaStream with ID " << new_stream->id() |
| << " is already added."; |
| return false; |
| } |
| return true; |
| } |
| |
| // If the direction is "recvonly" or "inactive", treat the description |
| // as containing no streams. |
| // See: https://code.google.com/p/webrtc/issues/detail?id=5054 |
| std::vector<cricket::StreamParams> GetActiveStreams( |
| const cricket::MediaContentDescription* desc) { |
| return RtpTransceiverDirectionHasSend(desc->direction()) |
| ? desc->streams() |
| : std::vector<cricket::StreamParams>(); |
| } |
| |
| bool IsValidOfferToReceiveMedia(int value) { |
| typedef PeerConnectionInterface::RTCOfferAnswerOptions Options; |
| return (value >= Options::kUndefined) && |
| (value <= Options::kMaxOfferToReceiveMedia); |
| } |
| |
| // Add options to |[audio/video]_media_description_options| from |senders|. |
| void AddPlanBRtpSenderOptions( |
| const std::vector<rtc::scoped_refptr< |
| RtpSenderProxyWithInternal<RtpSenderInternal>>>& senders, |
| cricket::MediaDescriptionOptions* audio_media_description_options, |
| cricket::MediaDescriptionOptions* video_media_description_options, |
| int num_sim_layers) { |
| for (const auto& sender : senders) { |
| if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) { |
| if (audio_media_description_options) { |
| audio_media_description_options->AddAudioSender( |
| sender->id(), sender->internal()->stream_ids()); |
| } |
| } else { |
| RTC_DCHECK(sender->media_type() == cricket::MEDIA_TYPE_VIDEO); |
| if (video_media_description_options) { |
| video_media_description_options->AddVideoSender( |
| sender->id(), sender->internal()->stream_ids(), {}, |
| SimulcastLayerList(), num_sim_layers); |
| } |
| } |
| } |
| } |
| |
| // Add options to |session_options| from |rtp_data_channels|. |
| void AddRtpDataChannelOptions( |
| const std::map<std::string, rtc::scoped_refptr<DataChannel>>& |
| rtp_data_channels, |
| cricket::MediaDescriptionOptions* data_media_description_options) { |
| if (!data_media_description_options) { |
| return; |
| } |
| // Check for data channels. |
| for (const auto& kv : rtp_data_channels) { |
| const DataChannel* channel = kv.second; |
| if (channel->state() == DataChannel::kConnecting || |
| channel->state() == DataChannel::kOpen) { |
| // Legacy RTP data channels are signaled with the track/stream ID set to |
| // the data channel's label. |
| data_media_description_options->AddRtpDataChannel(channel->label(), |
| channel->label()); |
| } |
| } |
| } |
| |
| uint32_t ConvertIceTransportTypeToCandidateFilter( |
| PeerConnectionInterface::IceTransportsType type) { |
| switch (type) { |
| case PeerConnectionInterface::kNone: |
| return cricket::CF_NONE; |
| case PeerConnectionInterface::kRelay: |
| return cricket::CF_RELAY; |
| case PeerConnectionInterface::kNoHost: |
| return (cricket::CF_ALL & ~cricket::CF_HOST); |
| case PeerConnectionInterface::kAll: |
| return cricket::CF_ALL; |
| default: |
| RTC_NOTREACHED(); |
| } |
| return cricket::CF_NONE; |
| } |
| |
| // Helper to set an error and return from a method. |
| bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) { |
| if (error) { |
| error->set_type(type); |
| } |
| return type == webrtc::RTCErrorType::NONE; |
| } |
| |
| bool SafeSetError(webrtc::RTCError error, webrtc::RTCError* error_out) { |
| bool ok = error.ok(); |
| if (error_out) { |
| *error_out = std::move(error); |
| } |
| return ok; |
| } |
| |
| std::string GetSignalingStateString( |
| PeerConnectionInterface::SignalingState state) { |
| switch (state) { |
| case PeerConnectionInterface::kStable: |
| return "kStable"; |
| case PeerConnectionInterface::kHaveLocalOffer: |
| return "kHaveLocalOffer"; |
| case PeerConnectionInterface::kHaveLocalPrAnswer: |
| return "kHavePrAnswer"; |
| case PeerConnectionInterface::kHaveRemoteOffer: |
| return "kHaveRemoteOffer"; |
| case PeerConnectionInterface::kHaveRemotePrAnswer: |
| return "kHaveRemotePrAnswer"; |
| case PeerConnectionInterface::kClosed: |
| return "kClosed"; |
| } |
| RTC_NOTREACHED(); |
| return ""; |
| } |
| |
| IceCandidatePairType GetIceCandidatePairCounter( |
| const cricket::Candidate& local, |
| const cricket::Candidate& remote) { |
| const auto& l = local.type(); |
| const auto& r = remote.type(); |
| const auto& host = LOCAL_PORT_TYPE; |
| const auto& srflx = STUN_PORT_TYPE; |
| const auto& relay = RELAY_PORT_TYPE; |
| const auto& prflx = PRFLX_PORT_TYPE; |
| if (l == host && r == host) { |
| bool local_hostname = |
| !local.address().hostname().empty() && local.address().IsUnresolvedIP(); |
| bool remote_hostname = !remote.address().hostname().empty() && |
| remote.address().IsUnresolvedIP(); |
| bool local_private = IPIsPrivate(local.address().ipaddr()); |
| bool remote_private = IPIsPrivate(remote.address().ipaddr()); |
| if (local_hostname) { |
| if (remote_hostname) { |
| return kIceCandidatePairHostNameHostName; |
| } else if (remote_private) { |
| return kIceCandidatePairHostNameHostPrivate; |
| } else { |
| return kIceCandidatePairHostNameHostPublic; |
| } |
| } else if (local_private) { |
| if (remote_hostname) { |
| return kIceCandidatePairHostPrivateHostName; |
| } else if (remote_private) { |
| return kIceCandidatePairHostPrivateHostPrivate; |
| } else { |
| return kIceCandidatePairHostPrivateHostPublic; |
| } |
| } else { |
| if (remote_hostname) { |
| return kIceCandidatePairHostPublicHostName; |
| } else if (remote_private) { |
| return kIceCandidatePairHostPublicHostPrivate; |
| } else { |
| return kIceCandidatePairHostPublicHostPublic; |
| } |
| } |
| } |
| if (l == host && r == srflx) |
| return kIceCandidatePairHostSrflx; |
| if (l == host && r == relay) |
| return kIceCandidatePairHostRelay; |
| if (l == host && r == prflx) |
| return kIceCandidatePairHostPrflx; |
| if (l == srflx && r == host) |
| return kIceCandidatePairSrflxHost; |
| if (l == srflx && r == srflx) |
| return kIceCandidatePairSrflxSrflx; |
| if (l == srflx && r == relay) |
| return kIceCandidatePairSrflxRelay; |
| if (l == srflx && r == prflx) |
| return kIceCandidatePairSrflxPrflx; |
| if (l == relay && r == host) |
| return kIceCandidatePairRelayHost; |
| if (l == relay && r == srflx) |
| return kIceCandidatePairRelaySrflx; |
| if (l == relay && r == relay) |
| return kIceCandidatePairRelayRelay; |
| if (l == relay && r == prflx) |
| return kIceCandidatePairRelayPrflx; |
| if (l == prflx && r == host) |
| return kIceCandidatePairPrflxHost; |
| if (l == prflx && r == srflx) |
| return kIceCandidatePairPrflxSrflx; |
| if (l == prflx && r == relay) |
| return kIceCandidatePairPrflxRelay; |
| return kIceCandidatePairMax; |
| } |
| |
| // Logic to decide if an m= section can be recycled. This means that the new |
| // m= section is not rejected, but the old local or remote m= section is |
| // rejected. |old_content_one| and |old_content_two| refer to the m= section |
| // of the old remote and old local descriptions in no particular order. |
| // We need to check both the old local and remote because either |
| // could be the most current from the latest negotation. |
| bool IsMediaSectionBeingRecycled(SdpType type, |
| const ContentInfo& content, |
| const ContentInfo* old_content_one, |
| const ContentInfo* old_content_two) { |
| return type == SdpType::kOffer && !content.rejected && |
| ((old_content_one && old_content_one->rejected) || |
| (old_content_two && old_content_two->rejected)); |
| } |
| |
| // Verify that the order of media sections in |new_desc| matches |
| // |current_desc|. The number of m= sections in |new_desc| should be no |
| // less than |current_desc|. In the case of checking an answer's |
| // |new_desc|, the |current_desc| is the last offer that was set as the |
| // local or remote. In the case of checking an offer's |new_desc| we |
| // check against the local and remote descriptions stored from the last |
| // negotiation, because either of these could be the most up to date for |
| // possible rejected m sections. These are the |current_desc| and |
| // |secondary_current_desc|. |
| bool MediaSectionsInSameOrder(const SessionDescription& current_desc, |
| const SessionDescription* secondary_current_desc, |
| const SessionDescription& new_desc, |
| const SdpType type) { |
| if (current_desc.contents().size() > new_desc.contents().size()) { |
| return false; |
| } |
| |
| for (size_t i = 0; i < current_desc.contents().size(); ++i) { |
| const cricket::ContentInfo* secondary_content_info = nullptr; |
| if (secondary_current_desc && |
| i < secondary_current_desc->contents().size()) { |
| secondary_content_info = &secondary_current_desc->contents()[i]; |
| } |
| if (IsMediaSectionBeingRecycled(type, new_desc.contents()[i], |
| ¤t_desc.contents()[i], |
| secondary_content_info)) { |
| // For new offer descriptions, if the media section can be recycled, it's |
| // valid for the MID and media type to change. |
| continue; |
| } |
| if (new_desc.contents()[i].name != current_desc.contents()[i].name) { |
| return false; |
| } |
| const MediaContentDescription* new_desc_mdesc = |
| new_desc.contents()[i].media_description(); |
| const MediaContentDescription* current_desc_mdesc = |
| current_desc.contents()[i].media_description(); |
| if (new_desc_mdesc->type() != current_desc_mdesc->type()) { |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| bool MediaSectionsHaveSameCount(const SessionDescription& desc1, |
| const SessionDescription& desc2) { |
| return desc1.contents().size() == desc2.contents().size(); |
| } |
| |
| void NoteKeyProtocolAndMedia(KeyExchangeProtocolType protocol_type, |
| cricket::MediaType media_type) { |
| // Array of structs needed to map {KeyExchangeProtocolType, |
| // cricket::MediaType} to KeyExchangeProtocolMedia without using std::map in |
| // order to avoid -Wglobal-constructors and -Wexit-time-destructors. |
| static constexpr struct { |
| KeyExchangeProtocolType protocol_type; |
| cricket::MediaType media_type; |
| KeyExchangeProtocolMedia protocol_media; |
| } kEnumCounterKeyProtocolMediaMap[] = { |
| {kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_AUDIO, |
| kEnumCounterKeyProtocolMediaTypeDtlsAudio}, |
| {kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_VIDEO, |
| kEnumCounterKeyProtocolMediaTypeDtlsVideo}, |
| {kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_DATA, |
| kEnumCounterKeyProtocolMediaTypeDtlsData}, |
| {kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_AUDIO, |
| kEnumCounterKeyProtocolMediaTypeSdesAudio}, |
| {kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_VIDEO, |
| kEnumCounterKeyProtocolMediaTypeSdesVideo}, |
| {kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_DATA, |
| kEnumCounterKeyProtocolMediaTypeSdesData}, |
| }; |
| |
| RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocol", protocol_type, |
| kEnumCounterKeyProtocolMax); |
| |
| for (const auto& i : kEnumCounterKeyProtocolMediaMap) { |
| if (i.protocol_type == protocol_type && i.media_type == media_type) { |
| RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocolByMedia", |
| i.protocol_media, |
| kEnumCounterKeyProtocolMediaTypeMax); |
| } |
| } |
| } |
| |
| void NoteAddIceCandidateResult(int result) { |
| RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.AddIceCandidate", result, |
| kAddIceCandidateMax); |
| } |
| |
| // Checks that each non-rejected content has SDES crypto keys or a DTLS |
| // fingerprint, 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. Mismatches, such as replying with a DTLS fingerprint |
| // to SDES keys, will be caught in JsepTransport negotiation, and backstopped |
| // by Channel's |srtp_required| check. |
| RTCError VerifyCrypto(const SessionDescription* desc, bool dtls_enabled) { |
| 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; |
| } |
| |
| // Get the SCTP port out of a SessionDescription. |
| // Return -1 if not found. |
| int GetSctpPort(const SessionDescription* session_description) { |
| const cricket::DataContentDescription* data_desc = |
| GetFirstDataContentDescription(session_description); |
| RTC_DCHECK(data_desc); |
| if (!data_desc) { |
| return -1; |
| } |
| std::string value; |
| cricket::DataCodec match_pattern(cricket::kGoogleSctpDataCodecPlType, |
| cricket::kGoogleSctpDataCodecName); |
| for (const cricket::DataCodec& codec : data_desc->codecs()) { |
| if (!codec.Matches(match_pattern)) { |
| continue; |
| } |
| if (codec.GetParam(cricket::kCodecParamPort, &value)) { |
| return rtc::FromString<int>(value); |
| } |
| } |
| return -1; |
| } |
| |
| // Returns true if |new_desc| requests an ICE restart (i.e., new ufrag/pwd). |
| bool CheckForRemoteIceRestart(const SessionDescriptionInterface* old_desc, |
| const SessionDescriptionInterface* new_desc, |
| const std::string& content_name) { |
| if (!old_desc) { |
| return false; |
| } |
| const SessionDescription* new_sd = new_desc->description(); |
| const SessionDescription* old_sd = old_desc->description(); |
| const ContentInfo* cinfo = new_sd->GetContentByName(content_name); |
| if (!cinfo || cinfo->rejected) { |
| return false; |
| } |
| // If the content isn't rejected, check if ufrag and password has changed. |
| const cricket::TransportDescription* new_transport_desc = |
| new_sd->GetTransportDescriptionByName(content_name); |
| const cricket::TransportDescription* old_transport_desc = |
| old_sd->GetTransportDescriptionByName(content_name); |
| if (!new_transport_desc || !old_transport_desc) { |
| // No transport description exists. This is not an ICE restart. |
| return false; |
| } |
| if (cricket::IceCredentialsChanged( |
| old_transport_desc->ice_ufrag, old_transport_desc->ice_pwd, |
| new_transport_desc->ice_ufrag, new_transport_desc->ice_pwd)) { |
| RTC_LOG(LS_INFO) << "Remote peer requests ICE restart for " << content_name |
| << "."; |
| return true; |
| } |
| return false; |
| } |
| |
| // 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; |
| } |
| |
| cricket::DataMessageType ToCricketDataMessageType(DataMessageType type) { |
| switch (type) { |
| case DataMessageType::kText: |
| return cricket::DMT_TEXT; |
| case DataMessageType::kBinary: |
| return cricket::DMT_BINARY; |
| case DataMessageType::kControl: |
| return cricket::DMT_CONTROL; |
| default: |
| return cricket::DMT_NONE; |
| } |
| return cricket::DMT_NONE; |
| } |
| |
| DataMessageType ToWebrtcDataMessageType(cricket::DataMessageType type) { |
| switch (type) { |
| case cricket::DMT_TEXT: |
| return DataMessageType::kText; |
| case cricket::DMT_BINARY: |
| return DataMessageType::kBinary; |
| case cricket::DMT_CONTROL: |
| return DataMessageType::kControl; |
| case cricket::DMT_NONE: |
| default: |
| RTC_NOTREACHED(); |
| } |
| return DataMessageType::kControl; |
| } |
| |
| } // namespace |
| |
| // Upon completion, posts a task to execute the callback of the |
| // SetSessionDescriptionObserver asynchronously on the same thread. At this |
| // point, the state of the peer connection might no longer reflect the effects |
| // of the SetRemoteDescription operation, as the peer connection could have been |
| // modified during the post. |
| // TODO(hbos): Remove this class once we remove the version of |
| // PeerConnectionInterface::SetRemoteDescription() that takes a |
| // SetSessionDescriptionObserver as an argument. |
| class PeerConnection::SetRemoteDescriptionObserverAdapter |
| : public rtc::RefCountedObject<SetRemoteDescriptionObserverInterface> { |
| public: |
| SetRemoteDescriptionObserverAdapter( |
| rtc::scoped_refptr<PeerConnection> pc, |
| rtc::scoped_refptr<SetSessionDescriptionObserver> wrapper) |
| : pc_(std::move(pc)), wrapper_(std::move(wrapper)) {} |
| |
| // SetRemoteDescriptionObserverInterface implementation. |
| void OnSetRemoteDescriptionComplete(RTCError error) override { |
| if (error.ok()) |
| pc_->PostSetSessionDescriptionSuccess(wrapper_); |
| else |
| pc_->PostSetSessionDescriptionFailure(wrapper_, 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; |
| bool presume_writable_when_fully_relayed; |
| bool enable_ice_renomination; |
| bool redetermine_role_on_ice_restart; |
| 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; |
| absl::optional<rtc::IntervalRange> ice_regather_interval_range; |
| webrtc::TurnCustomizer* turn_customizer; |
| SdpSemantics sdp_semantics; |
| absl::optional<rtc::AdapterType> network_preference; |
| bool active_reset_srtp_params; |
| bool use_media_transport; |
| bool use_media_transport_for_data_channels; |
| absl::optional<CryptoOptions> crypto_options; |
| bool offer_extmap_allow_mixed; |
| }; |
| 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 && |
| presume_writable_when_fully_relayed == |
| o.presume_writable_when_fully_relayed && |
| enable_ice_renomination == o.enable_ice_renomination && |
| redetermine_role_on_ice_restart == o.redetermine_role_on_ice_restart && |
| ice_check_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 && |
| ice_regather_interval_range == o.ice_regather_interval_range && |
| turn_customizer == o.turn_customizer && |
| sdp_semantics == o.sdp_semantics && |
| network_preference == o.network_preference && |
| active_reset_srtp_params == o.active_reset_srtp_params && |
| use_media_transport == o.use_media_transport && |
| use_media_transport_for_data_channels == |
| o.use_media_transport_for_data_channels && |
| crypto_options == o.crypto_options && |
| offer_extmap_allow_mixed == o.offer_extmap_allow_mixed; |
| } |
| |
| bool PeerConnectionInterface::RTCConfiguration::operator!=( |
| const PeerConnectionInterface::RTCConfiguration& o) const { |
| return !(*this == o); |
| } |
| |
| // Generate a RTCP CNAME when a PeerConnection is created. |
| std::string GenerateRtcpCname() { |
| std::string cname; |
| if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) { |
| RTC_LOG(LS_ERROR) << "Failed to generate CNAME."; |
| RTC_NOTREACHED(); |
| } |
| return cname; |
| } |
| |
| bool ValidateOfferAnswerOptions( |
| const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options) { |
| return IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_audio) && |
| IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_video); |
| } |
| |
| // From |rtc_options|, fill parts of |session_options| shared by all generated |
| // m= sections (in other words, nothing that involves a map/array). |
| void ExtractSharedMediaSessionOptions( |
| const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options, |
| cricket::MediaSessionOptions* session_options) { |
| session_options->vad_enabled = rtc_options.voice_activity_detection; |
| session_options->bundle_enabled = rtc_options.use_rtp_mux; |
| } |
| |
| PeerConnection::PeerConnection(PeerConnectionFactory* factory, |
| std::unique_ptr<RtcEventLog> event_log, |
| std::unique_ptr<Call> call) |
| : factory_(factory), |
| event_log_(std::move(event_log)), |
| rtcp_cname_(GenerateRtcpCname()), |
| local_streams_(StreamCollection::Create()), |
| remote_streams_(StreamCollection::Create()), |
| call_(std::move(call)) {} |
| |
| PeerConnection::~PeerConnection() { |
| TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection"); |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| |
| // Need to stop transceivers before destroying the stats collector because |
| // AudioRtpSender has a reference to the StatsCollector it will update when |
| // stopping. |
| for (auto transceiver : transceivers_) { |
| transceiver->Stop(); |
| } |
| |
| stats_.reset(nullptr); |
| if (stats_collector_) { |
| stats_collector_->WaitForPendingRequest(); |
| stats_collector_ = nullptr; |
| } |
| |
| // Don't destroy BaseChannels until after stats has been cleaned up so that |
| // the last stats request can still read from the channels. |
| DestroyAllChannels(); |
| |
| RTC_LOG(LS_INFO) << "Session: " << session_id() << " is destroyed."; |
| |
| webrtc_session_desc_factory_.reset(); |
| sctp_invoker_.reset(); |
| sctp_factory_.reset(); |
| media_transport_invoker_.reset(); |
| transport_controller_.reset(); |
| |
| // port_allocator_ lives on the network thread and should be destroyed there. |
| network_thread()->Invoke<void>(RTC_FROM_HERE, |
| [this] { port_allocator_.reset(); }); |
| // call_ and event_log_ must be destroyed on the worker thread. |
| worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] { |
| call_.reset(); |
| // The event log must outlive call (and any other object that uses it). |
| event_log_.reset(); |
| }); |
| } |
| |
| void PeerConnection::DestroyAllChannels() { |
| // Destroy video channels first since they may have a pointer to a voice |
| // channel. |
| for (auto transceiver : transceivers_) { |
| if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) { |
| DestroyTransceiverChannel(transceiver); |
| } |
| } |
| for (auto transceiver : transceivers_) { |
| if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) { |
| DestroyTransceiverChannel(transceiver); |
| } |
| } |
| DestroyDataChannel(); |
| } |
| |
| bool PeerConnection::Initialize( |
| const PeerConnectionInterface::RTCConfiguration& configuration, |
| PeerConnectionDependencies dependencies) { |
| 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); |
| 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; |
| } |
| |
| // The port allocator lives on the network thread and should be initialized |
| // there. |
| if (!network_thread()->Invoke<bool>( |
| RTC_FROM_HERE, |
| rtc::Bind(&PeerConnection::InitializePortAllocator_n, this, |
| stun_servers, turn_servers, configuration))) { |
| return false; |
| } |
| // 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 (port_allocator_flags_ & cricket::PORTALLOCATOR_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; |
| config.event_log = event_log_.get(); |
| #if defined(ENABLE_EXTERNAL_AUTH) |
| config.enable_external_auth = true; |
| #endif |
| config.active_reset_srtp_params = configuration.active_reset_srtp_params; |
| |
| if (configuration.use_media_transport || |
| configuration.use_media_transport_for_data_channels) { |
| if (!factory_->media_transport_factory()) { |
| RTC_DCHECK(false) |
| << "PeerConnecton is initialized with use_media_transport = true or " |
| << "use_media_transport_for_data_channels = true " |
| << "but media transport factory is not set in PeerConnectionFactory"; |
| return false; |
| } |
| |
| config.media_transport_factory = factory_->media_transport_factory(); |
| } |
| |
| 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_->SignalIceCandidatesRemoved.connect( |
| this, &PeerConnection::OnTransportControllerCandidatesRemoved); |
| transport_controller_->SignalDtlsHandshakeError.connect( |
| this, &PeerConnection::OnTransportControllerDtlsHandshakeError); |
| |
| sctp_factory_ = factory_->CreateSctpTransportInternalFactory(); |
| |
| stats_.reset(new StatsCollector(this)); |
| stats_collector_ = RTCStatsCollector::Create(this); |
| |
| configuration_ = configuration; |
| |
| // Obtain a certificate from RTCConfiguration if any were provided (optional). |
| rtc::scoped_refptr<rtc::RTCCertificate> certificate; |
| if (!configuration.certificates.empty()) { |
| // TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of |
| // just picking the first one. The decision should be made based on the DTLS |
| // handshake. The DTLS negotiations need to know about all certificates. |
| certificate = configuration.certificates[0]; |
| } |
| |
| transport_controller_->SetIceConfig(ParseIceConfig(configuration)); |
| |
| if (options.disable_encryption) { |
| dtls_enabled_ = false; |
| } else { |
| // Enable DTLS by default if we have an identity store or a certificate. |
| dtls_enabled_ = (dependencies.cert_generator || certificate); |
| // |configuration| can override the default |dtls_enabled_| value. |
| if (configuration.enable_dtls_srtp) { |
| dtls_enabled_ = *(configuration.enable_dtls_srtp); |
| } |
| } |
| |
| if (configuration.use_media_transport_for_data_channels) { |
| if (configuration.enable_rtp_data_channel) { |
| RTC_LOG(LS_ERROR) << "enable_rtp_data_channel and " |
| "use_media_transport_for_data_channels are " |
| "incompatible and cannot both be set to true"; |
| return false; |
| } |
| data_channel_type_ = cricket::DCT_MEDIA_TRANSPORT; |
| } else 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_type_ = cricket::DCT_RTP; |
| } else { |
| // DTLS has to be enabled to use SCTP. |
| if (!options.disable_sctp_data_channels && dtls_enabled_) { |
| data_channel_type_ = cricket::DCT_SCTP; |
| } |
| } |
| |
| video_options_.screencast_min_bitrate_kbps = |
| configuration.screencast_min_bitrate; |
| audio_options_.combined_audio_video_bwe = |
| configuration.combined_audio_video_bwe; |
| |
| audio_options_.audio_jitter_buffer_max_packets = |
| configuration.audio_jitter_buffer_max_packets; |
| |
| audio_options_.audio_jitter_buffer_fast_accelerate = |
| configuration.audio_jitter_buffer_fast_accelerate; |
| |
| 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)); |
| 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); |
| return true; |
| } |
| |
| RTCError PeerConnection::ValidateConfiguration( |
| const RTCConfiguration& config) const { |
| if (config.ice_regather_interval_range && |
| config.continual_gathering_policy == GATHER_ONCE) { |
| return RTCError(RTCErrorType::INVALID_PARAMETER, |
| "ice_regather_interval_range specified but continual " |
| "gathering policy is GATHER_ONCE"); |
| } |
| auto result = |
| cricket::P2PTransportChannel::ValidateIceConfig(ParseIceConfig(config)); |
| return result; |
| } |
| |
| rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::local_streams() { |
| 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_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_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); |
| Observer()->OnRenegotiationNeeded(); |
| return true; |
| } |
| |
| void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) { |
| 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; |
| } |
| Observer()->OnRenegotiationNeeded(); |
| } |
| |
| RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack( |
| rtc::scoped_refptr<MediaStreamTrackInterface> track, |
| const std::vector<std::string>& stream_ids) { |
| 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()) { |
| Observer()->OnRenegotiationNeeded(); |
| 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); |
| } 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) { |
| 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."); |
| } |
| } |
| Observer()->OnRenegotiationNeeded(); |
| return RTCError::OK(); |
| } |
| |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| PeerConnection::FindTransceiverBySender( |
| rtc::scoped_refptr<RtpSenderInterface> sender) { |
| for (auto transceiver : transceivers_) { |
| if (transceiver->sender() == sender) { |
| return transceiver; |
| } |
| } |
| return nullptr; |
| } |
| |
| RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> |
| PeerConnection::AddTransceiver( |
| rtc::scoped_refptr<MediaStreamTrackInterface> track) { |
| return AddTransceiver(track, RtpTransceiverInit()); |
| } |
| |
| RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> |
| PeerConnection::AddTransceiver( |
| rtc::scoped_refptr<MediaStreamTrackInterface> track, |
| const RtpTransceiverInit& init) { |
| 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_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 fire_callback) { |
| RTC_DCHECK((media_type == cricket::MEDIA_TYPE_AUDIO || |
| media_type == cricket::MEDIA_TYPE_VIDEO)); |
| if (track) { |
| RTC_DCHECK_EQ(media_type, |
| (track->kind() == MediaStreamTrackInterface::kAudioKind |
| ? cricket::MEDIA_TYPE_AUDIO |
| : cricket::MEDIA_TYPE_VIDEO)); |
| } |
| |
| // TODO(bugs.webrtc.org/7600): Verify init. |
| if (init.send_encodings.size() > 1) { |
| LOG_AND_RETURN_ERROR( |
| RTCErrorType::UNSUPPORTED_PARAMETER, |
| "Attempted to create an encoder with more than 1 encoding parameter."); |
| } |
| |
| for (const auto& encoding : init.send_encodings) { |
| if (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; |
| if (UnimplementedRtpParameterHasValue(parameters)) { |
| LOG_AND_RETURN_ERROR( |
| RTCErrorType::UNSUPPORTED_PARAMETER, |
| "Attempted to set an unimplemented parameter of RtpParameters."); |
| } |
| |
| 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, |
| init.send_encodings); |
| auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid()); |
| auto transceiver = CreateAndAddTransceiver(sender, receiver); |
| transceiver->internal()->set_direction(init.direction); |
| |
| if (fire_callback) { |
| Observer()->OnRenegotiationNeeded(); |
| } |
| |
| 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::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(), |
| new AudioRtpSender(worker_thread(), id, stats_.get())); |
| 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(), new VideoRtpSender(worker_thread(), id)); |
| 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)); |
| 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()); |
| Observer()->OnRenegotiationNeeded(); |
| } |
| |
| rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender( |
| const std::string& kind, |
| const std::string& stream_id) { |
| 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 = new AudioRtpSender( |
| worker_thread(), rtc::CreateRandomUuid(), stats_.get()); |
| 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 = |
| new VideoRtpSender(worker_thread(), rtc::CreateRandomUuid()); |
| 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 { |
| std::vector<rtc::scoped_refptr<RtpSenderInterface>> ret; |
| for (auto sender : GetSendersInternal()) { |
| ret.push_back(sender); |
| } |
| return ret; |
| } |
| |
| std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>> |
| PeerConnection::GetSendersInternal() const { |
| std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>> |
| all_senders; |
| for (auto transceiver : transceivers_) { |
| auto senders = transceiver->internal()->senders(); |
| all_senders.insert(all_senders.end(), senders.begin(), senders.end()); |
| } |
| return all_senders; |
| } |
| |
| std::vector<rtc::scoped_refptr<RtpReceiverInterface>> |
| PeerConnection::GetReceivers() const { |
| std::vector<rtc::scoped_refptr<RtpReceiverInterface>> ret; |
| for (const auto& receiver : GetReceiversInternal()) { |
| ret.push_back(receiver); |
| } |
| return ret; |
| } |
| |
| std::vector< |
| rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>> |
| PeerConnection::GetReceiversInternal() const { |
| std::vector< |
| rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>> |
| all_receivers; |
| for (auto transceiver : transceivers_) { |
| auto receivers = transceiver->internal()->receivers(); |
| all_receivers.insert(all_receivers.end(), receivers.begin(), |
| receivers.end()); |
| } |
| return all_receivers; |
| } |
| |
| std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> |
| PeerConnection::GetTransceivers() const { |
| RTC_CHECK(IsUnifiedPlan()) |
| << "GetTransceivers is only supported with Unified Plan SdpSemantics."; |
| std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers; |
| for (auto transceiver : transceivers_) { |
| all_transceivers.push_back(transceiver); |
| } |
| return all_transceivers; |
| } |
| |
| bool PeerConnection::GetStats(StatsObserver* observer, |
| MediaStreamTrackInterface* track, |
| StatsOutputLevel level) { |
| TRACE_EVENT0("webrtc", "PeerConnection::GetStats"); |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| if (!observer) { |
| RTC_LOG(LS_ERROR) << "GetStats - observer is NULL."; |
| return false; |
| } |
| |
| stats_->UpdateStats(level); |
| // The StatsCollector is used to tell if a track is valid because it may |
| // remember tracks that the PeerConnection previously removed. |
| if (track && !stats_->IsValidTrack(track->id())) { |
| RTC_LOG(LS_WARNING) << "GetStats is called with an invalid track: " |
| << track->id(); |
| return false; |
| } |
| signaling_thread()->Post(RTC_FROM_HERE, this, MSG_GETSTATS, |
| new GetStatsMsg(observer, track)); |
| return true; |
| } |
| |
| void PeerConnection::GetStats(RTCStatsCollectorCallback* callback) { |
| TRACE_EVENT0("webrtc", "PeerConnection::GetStats"); |
| 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(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(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() { |
| return signaling_state_; |
| } |
| |
| PeerConnectionInterface::IceConnectionState |
| PeerConnection::ice_connection_state() { |
| return ice_connection_state_; |
| } |
| |
| PeerConnectionInterface::IceConnectionState |
| PeerConnection::standardized_ice_connection_state() { |
| return standardized_ice_connection_state_; |
| } |
| |
| PeerConnectionInterface::PeerConnectionState |
| PeerConnection::peer_connection_state() { |
| return connection_state_; |
| } |
| |
| PeerConnectionInterface::IceGatheringState |
| PeerConnection::ice_gathering_state() { |
| return ice_gathering_state_; |
| } |
| |
| rtc::scoped_refptr<DataChannelInterface> PeerConnection::CreateDataChannel( |
| const std::string& label, |
| const DataChannelInit* config) { |
| TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel"); |
| |
| bool first_datachannel = !HasDataChannels(); |
| |
| std::unique_ptr<InternalDataChannelInit> internal_config; |
| if (config) { |
| internal_config.reset(new InternalDataChannelInit(*config)); |
| } |
| rtc::scoped_refptr<DataChannelInterface> channel( |
| InternalCreateDataChannel(label, internal_config.get())); |
| if (!channel.get()) { |
| return nullptr; |
| } |
| |
| // Trigger the onRenegotiationNeeded event for every new RTP DataChannel, or |
| // the first SCTP DataChannel. |
| if (data_channel_type() == cricket::DCT_RTP || first_datachannel) { |
| Observer()->OnRenegotiationNeeded(); |
| } |
| NoteUsageEvent(UsageEvent::DATA_ADDED); |
| return DataChannelProxy::Create(signaling_thread(), channel.get()); |
| } |
| |
| void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer, |
| const RTCOfferAnswerOptions& options) { |
| TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer"); |
| |
| if (!observer) { |
| RTC_LOG(LS_ERROR) << "CreateOffer - observer is NULL."; |
| return; |
| } |
| |
| if (IsClosed()) { |
| std::string error = "CreateOffer called when PeerConnection is closed."; |
| RTC_LOG(LS_ERROR) << error; |
| PostCreateSessionDescriptionFailure( |
| observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error))); |
| 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 (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) { |
| 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, /*fire_callback=*/false); |
| } |
| } |
| |
| std::vector<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>> |
| PeerConnection::GetReceivingTransceiversOfType(cricket::MediaType media_type) { |
| std::vector< |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>> |
| receiving_transceivers; |
| for (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) { |
| TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer"); |
| if (!observer) { |
| RTC_LOG(LS_ERROR) << "CreateAnswer - observer is NULL."; |
| 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) { |
| TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription"); |
| |
| // The SetLocalDescription contract is that we take ownership of the session |
| // description regardless of the outcome, so wrap it in a unique_ptr right |
| // away. Ideally, SetLocalDescription's signature will be changed to take the |
| // description as a unique_ptr argument to formalize this agreement. |
| std::unique_ptr<SessionDescriptionInterface> desc(desc_ptr); |
| |
| 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; |
| } |
| |
| 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()); |
| } |
| NoteUsageEvent(UsageEvent::SET_LOCAL_DESCRIPTION_CALLED); |
| } |
| |
| 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()); |
| |
| 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 (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 (auto transceiver : remove_list) { |
| observer->OnRemoveTrack(transceiver->receiver()); |
| } |
| for (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 (DataChannel::IsSctpLike(data_channel_type_) && GetSctpSslRole(&role)) { |
| AllocateSctpSids(role); |
| } |
| |
| if (IsUnifiedPlan()) { |
| for (auto transceiver : transceivers_) { |
| const ContentInfo* content = |
| FindMediaSectionForTransceiver(transceiver, local_description()); |
| if (!content) { |
| continue; |
| } |
| const auto& streams = content->media_description()->streams(); |
| if (!content->rejected && !streams.empty()) { |
| transceiver->internal()->sender_internal()->set_stream_ids( |
| streams[0].stream_ids()); |
| transceiver->internal()->sender_internal()->SetSsrc( |
| streams[0].first_ssrc()); |
| } else { |
| // 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 { |
| // 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::DataContentDescription* data_desc = |
| data_content->media_description()->as_data(); |
| if (absl::StartsWith(data_desc->protocol(), |
| cricket::kMediaProtocolRtpPrefix)) { |
| UpdateLocalRtpDataChannels(data_desc->streams()); |
| } |
| } |
| |
| 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* remote_description) { |
| RTC_DCHECK(remote_description); |
| const cricket::ContentInfos& local_contents = |
| (local_description() ? local_description()->description()->contents() |
| : cricket::ContentInfos()); |
| for (size_t i = 0; i < remote_description->contents().size(); ++i) { |
| cricket::ContentInfo& content = 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 { |
| 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"; |
| } |
| content.name = new_mid; |
| remote_description->transport_infos()[i].content_name = |
| std::string(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) { |
| SetRemoteDescription( |
| std::unique_ptr<SessionDescriptionInterface>(desc), |
| rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>( |
| new SetRemoteDescriptionObserverAdapter(this, observer))); |
| } |
| |
| void PeerConnection::SetRemoteDescription( |
| std::unique_ptr<SessionDescriptionInterface> desc, |
| rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) { |
| TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription"); |
| |
| if (!observer) { |
| RTC_LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL."; |
| return; |
| } |
| |
| if (!desc) { |
| observer->OnSetRemoteDescriptionComplete(RTCError( |
| RTCErrorType::INVALID_PARAMETER, "SessionDescription is NULL.")); |
| return; |
| } |
| |
| // 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 (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()); |
| } |
| |
| observer->OnSetRemoteDescriptionComplete(RTCError::OK()); |
| NoteUsageEvent(UsageEvent::SET_REMOTE_DESCRIPTION_CALLED); |
| } |
| |
| 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()); |
| |
| 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 (DataChannel::IsSctpLike(data_channel_type_) && GetSctpSslRole(&role)) { |
| 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 (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(); |
| } |
| 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)) { |
| // Set ssrc to 0 in the case of an unsignalled ssrc. |
| uint32_t ssrc = 0; |
| if (!media_desc->streams().empty() && |
| media_desc->streams()[0].has_ssrcs()) { |
| ssrc = media_desc->streams()[0].first_ssrc(); |
| } |
| transceiver->internal()->receiver_internal()->SetupMediaChannel(ssrc); |
| } |
| } |
| // Once all processing has finished, fire off callbacks. |
| auto observer = Observer(); |
| for (auto transceiver : now_receiving_transceivers) { |
| stats_->AddTrack(transceiver->receiver()->track()); |
| observer->OnTrack(transceiver); |
| observer->OnAddTrack(transceiver->receiver(), |
| transceiver->receiver()->streams()); |
| } |
| for (auto stream : added_streams) { |
| observer->OnAddStream(stream); |
| } |
| for (auto transceiver : remove_list) { |
| observer->OnRemoveTrack(transceiver->receiver()); |
| } |
| for (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::DataContentDescription* data_desc = |
| GetFirstDataContentDescription(remote_description()->description()); |
| |
| // Check if the descriptions include streams, just in case the peer supports |
| // MSID, but doesn't indicate so with "a=msid-semantic". |
| if (remote_description()->description()->msid_supported() || |
| (audio_desc && !audio_desc->streams().empty()) || |
| (video_desc && !video_desc->streams().empty())) { |
| remote_peer_supports_msid_ = true; |
| } |
| |
| // We wait to signal new streams until we finish processing the description, |
| // since only at that point will new streams have all their tracks. |
| rtc::scoped_refptr<StreamCollection> new_streams(StreamCollection::Create()); |
| |
| 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); |
| } |
| } |
| |
| // Update the DataChannels with the information from the remote peer. |
| if (data_desc) { |
| if (absl::StartsWith(data_desc->protocol(), |
| cricket::kMediaProtocolRtpPrefix)) { |
| UpdateRemoteRtpDataChannels(GetActiveStreams(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(); |
| } |
| |
| 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 (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) { |
| DestroyDataChannel(); |
| } else { |
| if (!rtp_data_channel_ && !sctp_transport_ && !media_transport_) { |
| 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())) { |
| UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc)); |
| } |
| } |
| } |
| return RTCError::OK(); |
| } |
| |
| 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... |
| if (!transceiver && |
| RtpTransceiverDirectionHasRecv(media_desc->direction())) { |
| 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(); |
| auto sender = |
| CreateSender(media_desc->type(), sender_id, nullptr, {}, {}); |
| 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); |
| } |
| } |
| 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."); |
| } |
| // 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() { |
| return configuration_; |
| } |
| |
| bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration, |
| RTCError* error) { |
| TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration"); |
| if (IsClosed()) { |
| RTC_LOG(LS_ERROR) << "SetConfiguration: PeerConnection is closed."; |
| return SafeSetError(RTCErrorType::INVALID_STATE, error); |
| } |
| |
| // 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) { |
| RTC_LOG(LS_ERROR) << "Can't change candidate pool size after calling " |
| "SetLocalDescription."; |
| return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error); |
| } |
| |
| if (local_description() && |
| configuration.use_media_transport != configuration_.use_media_transport) { |
| RTC_LOG(LS_ERROR) << "Can't change media_transport after calling " |
| "SetLocalDescription."; |
| return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error); |
| } |
| |
| if (remote_description() && |
| configuration.use_media_transport != configuration_.use_media_transport) { |
| RTC_LOG(LS_ERROR) << "Can't change media_transport after calling " |
| "SetRemoteDescription."; |
| return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error); |
| } |
| |
| if (local_description() && |
| configuration.use_media_transport_for_data_channels != |
| configuration_.use_media_transport_for_data_channels) { |
| RTC_LOG(LS_ERROR) << "Can't change media_transport_for_data_channels " |
| "after calling SetLocalDescription."; |
| return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error); |
| } |
| |
| if (remote_description() && |
| configuration.use_media_transport_for_data_channels != |
| configuration_.use_media_transport_for_data_channels) { |
| RTC_LOG(LS_ERROR) << "Can't change media_transport_for_data_channels " |
| "after calling SetRemoteDescription."; |
| return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error); |
| } |
| |
| if (local_description() && |
| configuration.crypto_options != configuration_.crypto_options) { |
| RTC_LOG(LS_ERROR) << "Can't change crypto_options after calling " |
| "SetLocalDescription."; |
| return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error); |
| } |
| |
| if (configuration.use_media_transport_for_data_channels || |
| configuration.use_media_transport) { |
| RTC_CHECK(configuration.bundle_policy == kBundlePolicyMaxBundle) |
| << "Media transport requires MaxBundle policy."; |
| } |
| |
| // The simplest (and most future-compatible) way to tell if the config was |
| // modified in an invalid way is to copy each property we do support |
| // modifying, then use operator==. There are far more properties we don't |
| // support modifying than those we do, and more could be added. |
| RTCConfiguration modified_config = configuration_; |
| modified_config.servers = configuration.servers; |
| modified_config.type = configuration.type; |
| modified_config.ice_candidate_pool_size = |
| configuration.ice_candidate_pool_size; |
| modified_config.prune_turn_ports = configuration.prune_turn_ports; |
| modified_config.ice_check_min_interval = configuration.ice_check_min_interval; |
| modified_config.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.use_media_transport = configuration.use_media_transport; |
| modified_config.use_media_transport_for_data_channels = |
| configuration.use_media_transport_for_data_channels; |
| if (configuration != modified_config) { |
| RTC_LOG(LS_ERROR) << "Modifying the configuration in an unsupported way."; |
| return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error); |
| } |
| |
| // Validate the modified configuration. |
| RTCError validate_error = ValidateConfiguration(modified_config); |
| if (!validate_error.ok()) { |
| return SafeSetError(std::move(validate_error), error); |
| } |
| |
| // Note that this isn't possible through chromium, since it's an unsigned |
| // short in WebIDL. |
| if (configuration.ice_candidate_pool_size < 0 || |
| configuration.ice_candidate_pool_size > static_cast<int>(UINT16_MAX)) { |
| return SafeSetError(RTCErrorType::INVALID_RANGE, error); |
| } |
| |
| // Parse ICE servers before hopping to network thread. |
| cricket::ServerAddresses stun_servers; |
| std::vector<cricket::RelayServerConfig> turn_servers; |
| RTCErrorType parse_error = |
| ParseIceServers(configuration.servers, &stun_servers, &turn_servers); |
| if (parse_error != RTCErrorType::NONE) { |
| return SafeSetError(parse_error, error); |
| } |
| // 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.prune_turn_ports, |
| modified_config.turn_customizer, |
| modified_config.stun_candidate_keepalive_interval))) { |
| RTC_LOG(LS_ERROR) << "Failed to apply configuration to PortAllocator."; |
| return SafeSetError(RTCErrorType::INTERNAL_ERROR, error); |
| } |
| |
| // As described in JSEP, calling setConfiguration with new ICE servers or |
| // candidate policy must set a "needs-ice-restart" bit so that the next offer |
| // triggers an ICE restart which will pick up the changes. |
| if (modified_config.servers != configuration_.servers || |
| modified_config.type != configuration_.type || |
| modified_config.prune_turn_ports != configuration_.prune_turn_ports) { |
| transport_controller_->SetNeedsIceRestartFlag(); |
| } |
| |
| transport_controller_->SetIceConfig(ParseIceConfig(modified_config)); |
| transport_controller_->SetMediaTransportFactory( |
| modified_config.use_media_transport || |
| modified_config.use_media_transport_for_data_channels |
| ? factory_->media_transport_factory() |
| : nullptr); |
| |
| if (configuration_.active_reset_srtp_params != |
| modified_config.active_reset_srtp_params) { |
| transport_controller_->SetActiveResetSrtpParams( |
| modified_config.active_reset_srtp_params); |
| } |
| |
| configuration_ = modified_config; |
| return SafeSetError(RTCErrorType::NONE, error); |
| } |
| |
| bool PeerConnection::AddIceCandidate( |
| const IceCandidateInterface* ice_candidate) { |
| 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::REMOTE_CANDIDATE_ADDED); |
| NoteAddIceCandidateResult(kAddIceCandidateSuccess); |
| } else { |
| NoteAddIceCandidateResult(kAddIceCandidateFailNotUsable); |
| } |
| return result; |
| } else { |
| RTC_LOG(LS_INFO) << "AddIceCandidate: Not ready to use candidate."; |
| NoteAddIceCandidateResult(kAddIceCandidateFailNotReady); |
| return true; |
| } |
| } |
| |
| bool PeerConnection::RemoveIceCandidates( |
| const std::vector<cricket::Candidate>& candidates) { |
| TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates"); |
| 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); }); |
| } |
| |
| 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_->GetTransportControllerSend()->SetClientBitratePreferences(bitrate); |
| |
| return RTCError::OK(); |
| } |
| |
| void PeerConnection::SetBitrateAllocationStrategy( |
| std::unique_ptr<rtc::BitrateAllocationStrategy> |
| bitrate_allocation_strategy) { |
| rtc::Thread* worker_thread = factory_->worker_thread(); |
| if (!worker_thread->IsCurrent()) { |
| rtc::BitrateAllocationStrategy* strategy_raw = |
| bitrate_allocation_strategy.release(); |
| auto functor = [this, strategy_raw]() { |
| call_->SetBitrateAllocationStrategy( |
| absl::WrapUnique<rtc::BitrateAllocationStrategy>(strategy_raw)); |
| }; |
| worker_thread->Invoke<void>(RTC_FROM_HERE, functor); |
| return; |
| } |
| RTC_DCHECK(call_.get()); |
| call_->SetBitrateAllocationStrategy(std::move(bitrate_allocation_strategy)); |
| } |
| |
| void PeerConnection::SetAudioPlayout(bool playout) { |
| if (!worker_thread()->IsCurrent()) { |
| worker_thread()->Invoke<void>( |
| RTC_FROM_HERE, |
| rtc::Bind(&PeerConnection::SetAudioPlayout, this, playout)); |
| return; |
| } |
| auto audio_state = |
| factory_->channel_manager()->media_engine()->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() { |
| 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; |
| } |
| |
| bool PeerConnection::StartRtcEventLog(rtc::PlatformFile file, |
| int64_t max_size_bytes) { |
| // TODO(eladalon): It would be better to not allow negative values into PC. |
| const size_t max_size = (max_size_bytes < 0) |
| ? RtcEventLog::kUnlimitedOutput |
| : rtc::saturated_cast<size_t>(max_size_bytes); |
| int64_t output_period_ms = webrtc::RtcEventLog::kImmediateOutput; |
| if (field_trial::IsEnabled("WebRTC-RtcEventLogNewFormat")) { |
| output_period_ms = 5000; |
| } |
| return StartRtcEventLog( |
| absl::make_unique<RtcEventLogOutputFile>(file, max_size), |
| output_period_ms); |
| } |
| |
| bool PeerConnection::StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output, |
| int64_t output_period_ms) { |
| // TODO(eladalon): In C++14, this can be done with a lambda. |
| struct Functor { |
| bool operator()() { |
| return pc->StartRtcEventLog_w(std::move(output), output_period_ms); |
| } |
| PeerConnection* const pc; |
| std::unique_ptr<RtcEventLogOutput> output; |
| const int64_t output_period_ms; |
| }; |
| return worker_thread()->Invoke<bool>( |
| RTC_FROM_HERE, Functor{this, std::move(output), output_period_ms}); |
| } |
| |
| void PeerConnection::StopRtcEventLog() { |
| worker_thread()->Invoke<void>( |
| RTC_FROM_HERE, rtc::Bind(&PeerConnection::StopRtcEventLog_w, this)); |
| } |
| |
| rtc::scoped_refptr<DtlsTransportInterface> |
| PeerConnection::LookupDtlsTransportByMid(const std::string& mid) { |
| return transport_controller_->LookupDtlsTransportByMid(mid); |
| } |
| |
| rtc::scoped_refptr<DtlsTransport> |
| PeerConnection::LookupDtlsTransportByMidInternal(const std::string& mid) { |
| return transport_controller_->LookupDtlsTransportByMid(mid); |
| } |
| |
| const SessionDescriptionInterface* PeerConnection::local_description() const { |
| return pending_local_description_ ? pending_local_description_.get() |
| : current_local_description_.get(); |
| } |
| |
| const SessionDescriptionInterface* PeerConnection::remote_description() const { |
| return pending_remote_description_ ? pending_remote_description_.get() |
| : current_remote_description_.get(); |
| } |
| |
| const SessionDescriptionInterface* PeerConnection::current_local_description() |
| const { |
| return current_local_description_.get(); |
| } |
| |
| const SessionDescriptionInterface* PeerConnection::current_remote_description() |
| const { |
| return current_remote_description_.get(); |
| } |
| |
| const SessionDescriptionInterface* PeerConnection::pending_local_description() |
| const { |
| return pending_local_description_.get(); |
| } |
| |
| const SessionDescriptionInterface* PeerConnection::pending_remote_description() |
| const { |
| return pending_remote_description_.get(); |
| } |
| |
| void PeerConnection::Close() { |
| TRACE_EVENT0("webrtc", "PeerConnection::Close"); |
| // Update stats here so that we have the most recent stats for tracks and |
| // streams before the channels are closed. |
| stats_->UpdateStats(kStatsOutputLevelStandard); |
| |
| ChangeSignalingState(PeerConnectionInterface::kClosed); |
| NoteUsageEvent(UsageEvent::CLOSE_CALLED); |
| |
| for (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] { |
| 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) { |
| 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_FREE_DATACHANNELS: { |
| sctp_data_channels_to_free_.clear(); |
| 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()); |
| audio_receiver->SetupMediaChannel(remote_sender_info.first_ssrc); |
| auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create( |
| signaling_thread(), audio_receiver); |
| GetAudioTransceiver()->internal()->AddReceiver(receiver); |
| Observer()->OnAddTrack(receiver, std::move(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()); |
| video_receiver->SetupMediaChannel(remote_sender_info.first_ssrc); |
| auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create( |
| signaling_thread(), video_receiver); |
| GetVideoTransceiver()->internal()->AddReceiver(receiver); |
| Observer()->OnAddTrack(receiver, std::move(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) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| if (ice_connection_state_ == new_state) { |
| return; |
| } |
| |
| // After transitioning to "closed", ignore any additional states from |
| // TransportController (such as "disconnected"). |
| if (IsClosed()) { |
| return; |
| } |
| |
| RTC_LOG(LS_INFO) << "Changing IceConnectionState " << ice_connection_state_ |
| << " => " << new_state; |
| RTC_DCHECK(ice_connection_state_ != |
| PeerConnectionInterface::kIceConnectionClosed); |
| |
| ice_connection_state_ = new_state; |
| Observer()->OnIceConnectionChange(ice_connection_state_); |
| } |
| |
| void PeerConnection::SetStandardizedIceConnectionState( |
| PeerConnectionInterface::IceConnectionState new_state) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| if (standardized_ice_connection_state_ == new_state) |
| return; |
| if (IsClosed()) |
| return; |
| standardized_ice_connection_state_ = new_state; |
| Observer()->OnStandardizedIceConnectionChange(new_state); |
| } |
| |
| void PeerConnection::SetConnectionState( |
| PeerConnectionInterface::PeerConnectionState new_state) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| if (connection_state_ == new_state) |
| return; |
| if (IsClosed()) |
| return; |
| connection_state_ = new_state; |
| Observer()->OnConnectionChange(new_state); |
| } |
| |
| void PeerConnection::OnIceGatheringChange( |
| PeerConnectionInterface::IceGatheringState new_state) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| if (IsClosed()) { |
| return; |
| } |
| ice_gathering_state_ = new_state; |
| Observer()->OnIceGatheringChange(ice_gathering_state_); |
| } |
| |
| void PeerConnection::OnIceCandidate( |
| std::unique_ptr<IceCandidateInterface> candidate) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| if (IsClosed()) { |
| return; |
| } |
| NoteUsageEvent(UsageEvent::CANDIDATE_COLLECTED); |
| if (candidate->candidate().type() == LOCAL_PORT_TYPE && |
| candidate->candidate().address().IsPrivateIP()) { |
| NoteUsageEvent(UsageEvent::PRIVATE_CANDIDATE_COLLECTED); |
| } |
| Observer()->OnIceCandidate(candidate.get()); |
| } |
| |
| void PeerConnection::OnIceCandidatesRemoved( |
| const std::vector<cricket::Candidate>& candidates) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| if (IsClosed()) { |
| return; |
| } |
| Observer()->OnIceCandidatesRemoved(candidates); |
| } |
| |
| void PeerConnection::ChangeSignalingState( |
| PeerConnectionInterface::SignalingState signaling_state) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| if (signaling_state_ == signaling_state) { |
| return; |
| } |
| RTC_LOG(LS_INFO) << "Session: " << session_id() << " Old state: " |
| << GetSignalingStateString(signaling_state_) |
| << " New state: " |
| << GetSignalingStateString(signaling_state); |
| signaling_state_ = signaling_state; |
| if (signaling_state == kClosed) { |
| ice_connection_state_ = kIceConnectionClosed; |
| Observer()->OnIceConnectionChange(ice_connection_state_); |
| standardized_ice_connection_state_ = |
| PeerConnectionInterface::IceConnectionState::kIceConnectionClosed; |
| connection_state_ = PeerConnectionInterface::PeerConnectionState::kClosed; |
| Observer()->OnConnectionChange(connection_state_); |
| if (ice_gathering_state_ != kIceGatheringComplete) { |
| ice_gathering_state_ = kIceGatheringComplete; |
| Observer()->OnIceGatheringChange(ice_gathering_state_); |
| } |
| } |
| Observer()->OnSignalingChange(signaling_state_); |
| } |
| |
| void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track, |
| MediaStreamInterface* stream) { |
| if (IsClosed()) { |
| return; |
| } |
| AddAudioTrack(track, stream); |
| Observer()->OnRenegotiationNeeded(); |
| } |
| |
| void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track, |
| MediaStreamInterface* stream) { |
| if (IsClosed()) { |
| return; |
| } |
| RemoveAudioTrack(track, stream); |
| Observer()->OnRenegotiationNeeded(); |
| } |
| |
| void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track, |
| MediaStreamInterface* stream) { |
| if (IsClosed()) { |
| return; |
| } |
| AddVideoTrack(track, stream); |
| Observer()->OnRenegotiationNeeded(); |
| } |
| |
| void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track, |
| MediaStreamInterface* stream) { |
| if (IsClosed()) { |
| return; |
| } |
| RemoveVideoTrack(track, stream); |
| Observer()->OnRenegotiationNeeded(); |
| } |
| |
| void PeerConnection::PostSetSessionDescriptionSuccess( |
| SetSessionDescriptionObserver* observer) { |
| SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer); |
| signaling_thread()->Post(RTC_FROM_HERE, this, |
| MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg); |
| } |
| |
| void PeerConnection::PostSetSessionDescriptionFailure( |
| SetSessionDescriptionObserver* observer, |
| 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 (!rtp_data_channels_.empty() || data_channel_type() != cricket::DCT_RTP) { |
| session_options->data_channel_type = data_channel_type(); |
| } |
| |
| // Apply ICE restart flag and renomination flag. |
| for (auto& options : session_options->media_description_options) { |
| options.transport_options.ice_restart = offer_answer_options.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; |
| } |
| |
| void PeerConnection::GetOptionsForPlanBOffer( |
| const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options, |
| cricket::MediaSessionOptions* session_options) { |
| // Figure out transceiver directional preferences. |
| bool send_audio = HasRtpSender(cricket::MEDIA_TYPE_AUDIO); |
| bool send_video = HasRtpSender(cricket::MEDIA_TYPE_VIDEO); |
| |
| // By default, generate sendrecv/recvonly m= sections. |
| bool recv_audio = true; |
| bool recv_video = true; |
| |
| // By default, only offer a new m= section if we have media to send with it. |
| bool offer_new_audio_description = send_audio; |
| bool offer_new_video_description = send_video; |
| bool offer_new_data_description = HasDataChannels(); |
| |
| // The "offer_to_receive_X" options allow those defaults to be overridden. |
| if (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) { |
| session_options->media_description_options.push_back( |
| cricket::MediaDescriptionOptions( |
| cricket::MEDIA_TYPE_AUDIO, cricket::CN_AUDIO, |
| RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio), |
| false)); |
| audio_index = session_options->media_description_options.size() - 1; |
| } |
| if (!video_index && offer_new_video_description) { |
| session_options->media_description_options.push_back( |
| cricket::MediaDescriptionOptions( |
| cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO, |
| RtpTransceiverDirectionFromSendRecv(send_video, recv_video), |
| false)); |
| video_index = session_options->media_description_options.size() - 1; |
| } |
| if (!data_index && offer_new_data_description) { |
| session_options->media_description_options.push_back( |
| 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()); |
| // 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())) { |
| cricket::SenderOptions sender_options; |
| sender_options.track_id = transceiver->sender()->id(); |
| sender_options.stream_ids = transceiver->sender()->stream_ids(); |
| int num_send_encoding_layers = |
| transceiver->sender()->init_send_encodings().size(); |
| sender_options.num_sim_layers = |
| !num_send_encoding_layers ? 1 : num_send_encoding_layers; |
| 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& local_contents = |
| (local_description() ? local_description()->description()->contents() |
| : ContentInfos()); |
| const ContentInfos& remote_contents = |
| (remote_description() ? remote_description()->description()->contents() |
| : ContentInfos()); |
| // 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); |
| RTC_CHECK(GetDataMid()); |
| if (had_been_rejected || mid != *GetDataMid()) { |
| session_options->media_description_options.push_back( |
| GetMediaDescriptionOptionsForRejectedData(mid)); |
| } else { |
| session_options->media_description_options.push_back( |
| GetMediaDescriptionOptionsForActiveData(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 (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() && 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 (!rtp_data_channels_.empty() || 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 = HasRtpSender(cricket::MEDIA_TYPE_AUDIO); |
| bool send_video = HasRtpSender(cricket::MEDIA_TYPE_VIDEO); |
| |
| // By default, generate sendrecv/recvonly m= sections. The direction is also |
| // restricted by the direction in the offer. |
| bool recv_audio = true; |
| bool recv_video = true; |
| |
| // The "offer_to_receive_X" options allow those defaults to be overridden. |
| if (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; |
| } |
| } 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; |
| } |
| } 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(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(rtp_data_channels_, &options); |
| return options; |
| } |
| |
| absl::optional<std::string> PeerConnection::GetDataMid() const { |
| switch (data_channel_type_) { |
| case cricket::DCT_RTP: |
| if (!rtp_data_channel_) { |
| return absl::nullopt; |
| } |
| return rtp_data_channel_->content_name(); |
| case cricket::DCT_SCTP: |
| return sctp_mid_; |
| case cricket::DCT_MEDIA_TRANSPORT: |
| return media_transport_data_mid_; |
| 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, 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::UpdateLocalRtpDataChannels( |
| const cricket::StreamParamsVec& streams) { |
| std::vector<std::string> existing_channels; |
| |
| // Find new and active data channels. |
| for (const cricket::StreamParams& params : streams) { |
| // |it->sync_label| is actually the data channel label. The reason is that |
| // we use the same naming of data channels as we do for |
| // MediaStreams and Tracks. |
| // For MediaStreams, the sync_label is the MediaStream label and the |
| // track label is the same as |streamid|. |
| const std::string& channel_label = params.first_stream_id(); |
| auto data_channel_it = rtp_data_channels_.find(channel_label); |
| if (data_channel_it == rtp_data_channels_.end()) { |
| RTC_LOG(LS_ERROR) << "channel label not found"; |
| continue; |
| } |
| // Set the SSRC the data channel should use for sending. |
| data_channel_it->second->SetSendSsrc(params.first_ssrc()); |
| existing_channels.push_back(data_channel_it->first); |
| } |
| |
| UpdateClosingRtpDataChannels(existing_channels, true); |
| } |
| |
| void PeerConnection::UpdateRemoteRtpDataChannels( |
| const cricket::StreamParamsVec& streams) { |
| std::vector<std::string> existing_channels; |
| |
| // Find new and active data channels. |
| for (const cricket::StreamParams& params : streams) { |
| // The data channel label is either the mslabel or the SSRC if the mslabel |
| // does not exist. Ex a=ssrc:444330170 mslabel:test1. |
| std::string label = params.first_stream_id().empty() |
| ? rtc::ToString(params.first_ssrc()) |
| : params.first_stream_id(); |
| auto data_channel_it = rtp_data_channels_.find(label); |
| if (data_channel_it == rtp_data_channels_.end()) { |
| // This is a new data channel. |
| CreateRemoteRtpDataChannel(label, params.first_ssrc()); |
| } else { |
| data_channel_it->second->SetReceiveSsrc(params.first_ssrc()); |
| } |
| existing_channels.push_back(label); |
| } |
| |
| UpdateClosingRtpDataChannels(existing_channels, false); |
| } |
| |
| void PeerConnection::UpdateClosingRtpDataChannels( |
| const std::vector<std::string>& active_channels, |
| bool is_local_update) { |
| auto it = rtp_data_channels_.begin(); |
| while (it != rtp_data_channels_.end()) { |
| DataChannel* data_channel = it->second; |
| if (std::find(active_channels.begin(), active_channels.end(), |
| data_channel->label()) != active_channels.end()) { |
| ++it; |
| continue; |
| } |
| |
| if (is_local_update) { |
| data_channel->SetSendSsrc(0); |
| } else { |
| data_channel->RemotePeerRequestClose(); |
| } |
| |
| if (data_channel->state() == DataChannel::kClosed) { |
| rtp_data_channels_.erase(it); |
| it = rtp_data_channels_.begin(); |
| } else { |
| ++it; |
| } |
| } |
| } |
| |
| void PeerConnection::CreateRemoteRtpDataChannel(const std::string& label, |
| uint32_t remote_ssrc) { |
| rtc::scoped_refptr<DataChannel> channel( |
| InternalCreateDataChannel(label, nullptr)); |
| if (!channel.get()) { |
| RTC_LOG(LS_WARNING) << "Remote peer requested a DataChannel but" |
| "CreateDataChannel failed."; |
| return; |
| } |
| channel->SetReceiveSsrc(remote_ssrc); |
| rtc::scoped_refptr<DataChannelInterface> proxy_channel = |
| DataChannelProxy::Create(signaling_thread(), channel); |
| Observer()->OnDataChannel(std::move(proxy_channel)); |
| } |
| |
| rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel( |
| const std::string& label, |
| const InternalDataChannelInit* config) { |
| if (IsClosed()) { |
| return nullptr; |
| } |
| if (data_channel_type() == cricket::DCT_NONE) { |
| RTC_LOG(LS_ERROR) |
| << "InternalCreateDataChannel: Data is not supported in this call."; |
| return nullptr; |
| } |
| InternalDataChannelInit new_config = |
| config ? (*config) : InternalDataChannelInit(); |
| if (DataChannel::IsSctpLike(data_channel_type_)) { |
| if (new_config.id < 0) { |
| rtc::SSLRole role; |
| if ((GetSctpSslRole(&role)) && |
| !sid_allocator_.AllocateSid(role, &new_config.id)) { |
| RTC_LOG(LS_ERROR) |
| << "No id can be allocated for the SCTP data channel."; |
| return nullptr; |
| } |
| } else if (!sid_allocator_.ReserveSid(new_config.id)) { |
| RTC_LOG(LS_ERROR) << "Failed to create a SCTP data channel " |
| "because the id is already in use or out of range."; |
| return nullptr; |
| } |
| } |
| |
| rtc::scoped_refptr<DataChannel> channel( |
| DataChannel::Create(this, data_channel_type(), label, new_config)); |
| if (!channel) { |
| sid_allocator_.ReleaseSid(new_config.id); |
| return nullptr; |
| } |
| |
| if (channel->data_channel_type() == cricket::DCT_RTP) { |
| if (rtp_data_channels_.find(channel->label()) != rtp_data_channels_.end()) { |
| RTC_LOG(LS_ERROR) << "DataChannel with label " << channel->label() |
| << " already exists."; |
| return nullptr; |
| } |
| rtp_data_channels_[channel->label()] = channel; |
| } else { |
| RTC_DCHECK(DataChannel::IsSctpLike(data_channel_type_)); |
| sctp_data_channels_.push_back(channel); |
| channel->SignalClosed.connect(this, |
| &PeerConnection::OnSctpDataChannelClosed); |
| } |
| |
| SignalDataChannelCreated_(channel.get()); |
| return channel; |
| } |
| |
| bool PeerConnection::HasDataChannels() const { |
| return !rtp_data_channels_.empty() || !sctp_data_channels_.empty(); |
| } |
| |
| void PeerConnection::AllocateSctpSids(rtc::SSLRole role) { |
| for (const auto& channel : sctp_data_channels_) { |
| if (channel->id() < 0) { |
| int sid; |
| if (!sid_allocator_.AllocateSid(role, &sid)) { |
| RTC_LOG(LS_ERROR) << "Failed to allocate SCTP sid."; |
| continue; |
| } |
| channel->SetSctpSid(sid); |
| } |
| } |
| } |
| |
| void PeerConnection::OnSctpDataChannelClosed(DataChannel* channel) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| for (auto it = sctp_data_channels_.begin(); it != sctp_data_channels_.end(); |
| ++it) { |
| if (it->get() == channel) { |
| if (channel->id() >= 0) { |
| // After the closing procedure is done, it's safe to use this ID for |
| // another data channel. |
| sid_allocator_.ReleaseSid(channel->id()); |
| } |
| // Since this method is triggered by a signal from the DataChannel, |
| // we can't free it directly here; we need to free it asynchronously. |
| sctp_data_channels_to_free_.push_back(*it); |
| sctp_data_channels_.erase(it); |
| signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FREE_DATACHANNELS, |
| nullptr); |
| return; |
| } |
| } |
| } |
| |
| void PeerConnection::OnDataChannelDestroyed() { |
| // Use a temporary copy of the RTP/SCTP DataChannel list because the |
| // DataChannel may callback to us and try to modify the list. |
| std::map<std::string, rtc::scoped_refptr<DataChannel>> temp_rtp_dcs; |
| temp_rtp_dcs.swap(rtp_data_channels_); |
| for (const auto& kv : temp_rtp_dcs) { |
| kv.second->OnTransportChannelDestroyed(); |
| } |
| |
| std::vector<rtc::scoped_refptr<DataChannel>> temp_sctp_dcs; |
| temp_sctp_dcs.swap(sctp_data_channels_); |
| for (const auto& channel : temp_sctp_dcs) { |
| channel->OnTransportChannelDestroyed(); |
| } |
| } |
| |
| void PeerConnection::OnDataChannelOpenMessage( |
| const std::string& label, |
| const InternalDataChannelInit& config) { |
| rtc::scoped_refptr<DataChannel> channel( |
| InternalCreateDataChannel(label, &config)); |
| if (!channel.get()) { |
| RTC_LOG(LS_ERROR) << "Failed to create DataChannel from the OPEN message."; |
| return; |
| } |
| |
| rtc::scoped_refptr<DataChannelInterface> proxy_channel = |
| DataChannelProxy::Create(signaling_thread(), channel); |
| Observer()->OnDataChannel(std::move(proxy_channel)); |
| NoteUsageEvent(UsageEvent::DATA_ADDED); |
| } |
| |
| bool PeerConnection::HandleOpenMessage_s( |
| const cricket::ReceiveDataParams& params, |
| const rtc::CopyOnWriteBuffer& buffer) { |
| if (params.type == cricket::DMT_CONTROL && IsOpenMessage(buffer)) { |
| // Received OPEN message; parse and signal that a new data channel should |
| // be created. |
| std::string label; |
| InternalDataChannelInit config; |
| config.id = params.ssrc; |
| if (!ParseDataChannelOpenMessage(buffer, &label, &config)) { |
| RTC_LOG(LS_WARNING) << "Failed to parse the OPEN message for ssrc " |
| << params.ssrc; |
| return true; |
| } |
| config.open_handshake_role = InternalDataChannelInit::kAcker; |
| OnDataChannelOpenMessage(label, config); |
| return true; |
| } |
| return false; |
| } |
| |
| 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; |
| } |
| |
| // TODO(bugs.webrtc.org/7600): Remove this when multiple transceivers with |
| // individual transceiver directions are supported. |
| bool PeerConnection::HasRtpSender(cricket::MediaType type) const { |
| switch (type) { |
| case cricket::MEDIA_TYPE_AUDIO: |
| return !GetAudioTransceiver()->internal()->senders().empty(); |
| case cricket::MEDIA_TYPE_VIDEO: |
| return !GetVideoTransceiver()->internal()->senders().empty(); |
| case cricket::MEDIA_TYPE_DATA: |
| return false; |
| } |
| RTC_NOTREACHED(); |
| return false; |
| } |
| |
| rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> |
| PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) const { |
| for (auto transceiver : transceivers_) { |
| for (auto sender : transceiver->internal()->senders()) { |
| if (sender->track() == track) { |
| return sender; |
| } |
| } |
| } |
| return nullptr; |
| } |
| |
| rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> |
| PeerConnection::FindSenderById(const std::string& sender_id) const { |
| for (auto transceiver : transceivers_) { |
| for (auto sender : transceiver->internal()->senders()) { |
| if (sender->id() == sender_id) { |
| return sender; |
| } |
| } |
| } |
| return nullptr; |
| } |
| |
| rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>> |
| PeerConnection::FindReceiverById(const std::string& receiver_id) const { |
| for (auto transceiver : transceivers_) { |
| for (auto receiver : transceiver->internal()->receivers()) { |
| if (receiver->id() == receiver_id) { |
| return receiver; |
| } |
| } |
| } |
| return nullptr; |
| } |
| |
| std::vector<PeerConnection::RtpSenderInfo>* |
| PeerConnection::GetRemoteSenderInfos(cricket::MediaType media_type) { |
| RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO || |
| media_type == cricket::MEDIA_TYPE_VIDEO); |
| return (media_type == cricket::MEDIA_TYPE_AUDIO) |
| ? &remote_audio_sender_infos_ |
| : &remote_video_sender_infos_; |
| } |
| |
| std::vector<PeerConnection::RtpSenderInfo>* PeerConnection::GetLocalSenderInfos( |
| cricket::MediaType media_type) { |
| RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO || |
| media_type == cricket::MEDIA_TYPE_VIDEO); |
| return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &local_audio_sender_infos_ |
| : &local_video_sender_infos_; |
| } |
| |
| const PeerConnection::RtpSenderInfo* PeerConnection::FindSenderInfo( |
| const std::vector<PeerConnection::RtpSenderInfo>& infos, |
| const std::string& stream_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; |
| } |
| |
| DataChannel* PeerConnection::FindDataChannelBySid(int sid) const { |
| for (const auto& channel : sctp_data_channels_) { |
| if (channel->id() == sid) { |
| return channel; |
| } |
| } |
| return nullptr; |
| } |
| |
| bool PeerConnection::InitializePortAllocator_n( |
| const cricket::ServerAddresses& stun_servers, |
| const std::vector<cricket::RelayServerConfig>& turn_servers, |
| const RTCConfiguration& configuration) { |
| port_allocator_->Initialize(); |
| // To handle both internal and externally created port allocator, we will |
| // enable BUNDLE here. |
| 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 (webrtc::field_trial::FindFullName("WebRTC-IPv6Default") |
| .find("Disabled") == 0) { |
| 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_->set_candidate_filter( |
| 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.prune_turn_ports, |
| configuration.turn_customizer, |
| configuration.stun_candidate_keepalive_interval); |
| return true; |
| } |
| |
| bool PeerConnection::ReconfigurePortAllocator_n( |
| const cricket::ServerAddresses& stun_servers, |
| const std::vector<cricket::RelayServerConfig>& turn_servers, |
| IceTransportsType type, |
| int candidate_pool_size, |
| bool prune_turn_ports, |
| webrtc::TurnCustomizer* turn_customizer, |
| absl::optional<int> stun_candidate_keepalive_interval) { |
| port_allocator_->set_candidate_filter( |
| 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 (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, |
| prune_turn_ports, 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) { |
| if (!event_log_) { |
| return false; |
| } |
| return event_log_->StartLogging(std::move(output), output_period_ms); |
| } |
| |
| void PeerConnection::StopRtcEventLog_w() { |
| if (event_log_) { |
| event_log_->StopLogging(); |
| } |
| } |
| |
| cricket::ChannelInterface* PeerConnection::GetChannel( |
| const std::string& content_name) { |
| for (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_INFO) |
| << "Local and Remote descriptions must be applied to get the " |
| "SSL Role of the SCTP transport."; |
| return false; |
| } |
| if (!sctp_transport_ && !media_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_) { |
| dtls_role = transport_controller_->GetDtlsRole(*sctp_mid_); |
| } else if (is_caller_) { |
| dtls_role = *is_caller_ ? rtc::SSL_SERVER : rtc::SSL_CLIENT; |
| } |
| if (dtls_role) { |
| *role = *dtls_role; |
| return true; |
| } |
| return false; |
| } |
| |
| bool PeerConnection::GetSslRole(const std::string& content_name, |
| rtc::SSLRole* role) { |
| if (!local_description() || !remote_description()) { |
| RTC_LOG(LS_INFO) |
| << "Local and Remote descriptions must be applied to get the " |
| "SSL Role of the session."; |
| return false; |
| } |
| |
| 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); |
| } |
| |
| // 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 (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 (rtp_data_channel_) { |
| const ContentInfo* data_content = |
| cricket::GetFirstDataContent(sdesc->description()); |
| if (data_content && !data_content->rejected) { |
| const MediaContentDescription* data_desc = |
| data_content->media_description(); |
| if (data_desc) { |
| std::string error; |
| bool success = |
| (source == cricket::CS_LOCAL) |
| ? rtp_data_channel_->SetLocalContent(data_desc, type, &error) |
| : rtp_data_channel_->SetRemoteContent(data_desc, type, &error); |
| if (!success) { |
| LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, error); |
| } |
| } |
| } |
| } |
| |
| // Need complete offer/answer with an SCTP m= section before starting SCTP, |
| // according to https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-19 |
| if (sctp_transport_ && local_description() && remote_description() && |
| cricket::GetFirstDataContent(local_description()->description()) && |
| cricket::GetFirstDataContent(remote_description()->description())) { |
| bool success = network_thread()->Invoke<bool>( |
| RTC_FROM_HERE, |
| rtc::Bind(&PeerConnection::PushdownSctpParameters_n, this, source)); |
| if (!success) { |
| LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, |
| "Failed to push down SCTP parameters."); |
| } |
| } |
| |
| return RTCError::OK(); |
| } |
| |
| bool PeerConnection::PushdownSctpParameters_n(cricket::ContentSource source) { |
| RTC_DCHECK(network_thread()->IsCurrent()); |
| RTC_DCHECK(local_description()); |
| RTC_DCHECK(remote_description()); |
| // Apply the SCTP port (which is hidden inside a DataCodec structure...) |
| // When we support "max-message-size", that would also be pushed down here. |
| return sctp_transport_->Start( |
| GetSctpPort(local_description()->description()), |
| GetSctpPort(remote_description()->description())); |
| } |
| |
| RTCError PeerConnection::PushdownTransportDescription( |
| cricket::ContentSource source, |
| SdpType type) { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| |
| 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.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.regather_all_networks_interval_range = |
| config.ice_regather_interval_range; |
| ice_config.network_preference = config.network_preference; |
| return ice_config; |
| } |
| |
| static absl::string_view GetTrackIdBySsrc( |
| const SessionDescriptionInterface* session_description, |
| uint32_t ssrc) { |
| if (!session_description) { |
| return {}; |
| } |
| for (const cricket::ContentInfo& content : |
| session_description->description()->contents()) { |
| const cricket::MediaContentDescription& media = |
| *content.media_description(); |
| if (media.type() == cricket::MEDIA_TYPE_AUDIO || |
| media.type() == cricket::MEDIA_TYPE_VIDEO) { |
| const cricket::StreamParams* stream_params = |
| cricket::GetStreamBySsrc(media.streams(), ssrc); |
| if (stream_params) { |
| return stream_params->id; |
| } |
| } |
| } |
| return {}; |
| } |
| |
| absl::string_view PeerConnection::GetLocalTrackIdBySsrc(uint32_t ssrc) { |
| return GetTrackIdBySsrc(local_description(), ssrc); |
| } |
| |
| absl::string_view PeerConnection::GetRemoteTrackIdBySsrc(uint32_t ssrc) { |
| return GetTrackIdBySsrc(remote_description(), ssrc); |
| } |
| |
| bool PeerConnection::SendData(const cricket::SendDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload, |
| cricket::SendDataResult* result) { |
| if (!rtp_data_channel_ && !sctp_transport_ && !media_transport_) { |
| RTC_LOG(LS_ERROR) << "SendData called when rtp_data_channel_, " |
| "sctp_transport_, and media_transport_ are NULL."; |
| return false; |
| } |
| if (media_transport_) { |
| SendDataParams send_params; |
| send_params.type = ToWebrtcDataMessageType(params.type); |
| send_params.ordered = params.ordered; |
| if (params.max_rtx_count >= 0) { |
| send_params.max_rtx_count = params.max_rtx_count; |
| } else if (params.max_rtx_ms >= 0) { |
| send_params.max_rtx_ms = params.max_rtx_ms; |
| } |
| return media_transport_->SendData(params.sid, send_params, payload).ok(); |
| } |
| return rtp_data_channel_ |
| ? rtp_data_channel_->SendData(params, payload, result) |
| : network_thread()->Invoke<bool>( |
| RTC_FROM_HERE, |
| Bind(&cricket::SctpTransportInternal::SendData, |
| sctp_transport_.get(), params, payload, result)); |
| } |
| |
| bool PeerConnection::ConnectDataChannel(DataChannel* webrtc_data_channel) { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| if (!rtp_data_channel_ && !sctp_transport_ && !media_transport_) { |
| // Don't log an error here, because DataChannels are expected to call |
| // ConnectDataChannel in this state. It's the only way to initially tell |
| // whether or not the underlying transport is ready. |
| return false; |
| } |
| if (media_transport_) { |
| SignalMediaTransportWritable_s.connect(webrtc_data_channel, |
| &DataChannel::OnChannelReady); |
| SignalMediaTransportReceivedData_s.connect(webrtc_data_channel, |
| &DataChannel::OnDataReceived); |
| SignalMediaTransportChannelClosing_s.connect( |
| webrtc_data_channel, &DataChannel::OnClosingProcedureStartedRemotely); |
| SignalMediaTransportChannelClosed_s.connect( |
| webrtc_data_channel, &DataChannel::OnClosingProcedureComplete); |
| } else if (rtp_data_channel_) { |
| rtp_data_channel_->SignalReadyToSendData.connect( |
| webrtc_data_channel, &DataChannel::OnChannelReady); |
| rtp_data_channel_->SignalDataReceived.connect(webrtc_data_channel, |
| &DataChannel::OnDataReceived); |
| } else { |
| SignalSctpReadyToSendData.connect(webrtc_data_channel, |
| &DataChannel::OnChannelReady); |
| SignalSctpDataReceived.connect(webrtc_data_channel, |
| &DataChannel::OnDataReceived); |
| SignalSctpClosingProcedureStartedRemotely.connect( |
| webrtc_data_channel, &DataChannel::OnClosingProcedureStartedRemotely); |
| SignalSctpClosingProcedureComplete.connect( |
| webrtc_data_channel, &DataChannel::OnClosingProcedureComplete); |
| } |
| return true; |
| } |
| |
| void PeerConnection::DisconnectDataChannel(DataChannel* webrtc_data_channel) { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| if (!rtp_data_channel_ && !sctp_transport_ && !media_transport_) { |
| RTC_LOG(LS_ERROR) |
| << "DisconnectDataChannel called when rtp_data_channel_ and " |
| "sctp_transport_ are NULL."; |
| return; |
| } |
| if (media_transport_) { |
| SignalMediaTransportWritable_s.disconnect(webrtc_data_channel); |
| SignalMediaTransportReceivedData_s.disconnect(webrtc_data_channel); |
| SignalMediaTransportChannelClosing_s.disconnect(webrtc_data_channel); |
| SignalMediaTransportChannelClosed_s.disconnect(webrtc_data_channel); |
| } else if (rtp_data_channel_) { |
| rtp_data_channel_->SignalReadyToSendData.disconnect(webrtc_data_channel); |
| rtp_data_channel_->SignalDataReceived.disconnect(webrtc_data_channel); |
| } else { |
| SignalSctpReadyToSendData.disconnect(webrtc_data_channel); |
| SignalSctpDataReceived.disconnect(webrtc_data_channel); |
| SignalSctpClosingProcedureStartedRemotely.disconnect(webrtc_data_channel); |
| SignalSctpClosingProcedureComplete.disconnect(webrtc_data_channel); |
| } |
| } |
| |
| void PeerConnection::AddSctpDataStream(int sid) { |
| if (media_transport_) { |
| // No-op. Media transport does not need to add streams. |
| return; |
| } |
| if (!sctp_transport_) { |
| RTC_LOG(LS_ERROR) |
| << "AddSctpDataStream called when sctp_transport_ is NULL."; |
| return; |
| } |
| network_thread()->Invoke<void>( |
| RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::OpenStream, |
| sctp_transport_.get(), sid)); |
| } |
| |
| void PeerConnection::RemoveSctpDataStream(int sid) { |
| if (media_transport_) { |
| media_transport_->CloseChannel(sid); |
| return; |
| } |
| if (!sctp_transport_) { |
| RTC_LOG(LS_ERROR) << "RemoveSctpDataStream called when sctp_transport_ is " |
| "NULL."; |
| return; |
| } |
| network_thread()->Invoke<void>( |
| RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::ResetStream, |
| sctp_transport_.get(), sid)); |
| } |
| |
| bool PeerConnection::ReadyToSendData() const { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return (rtp_data_channel_ && rtp_data_channel_->ready_to_send_data()) || |
| (media_transport_ && media_transport_ready_to_send_data_) || |
| sctp_ready_to_send_data_; |
| } |
| |
| void PeerConnection::OnDataReceived(int channel_id, |
| DataMessageType type, |
| const rtc::CopyOnWriteBuffer& buffer) { |
| cricket::ReceiveDataParams params; |
| params.sid = channel_id; |
| params.type = ToCricketDataMessageType(type); |
| media_transport_invoker_->AsyncInvoke<void>( |
| RTC_FROM_HERE, signaling_thread(), [this, params, buffer] { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| if (!HandleOpenMessage_s(params, buffer)) { |
| SignalMediaTransportReceivedData_s(params, buffer); |
| } |
| }); |
| } |
| |
| void PeerConnection::OnChannelClosing(int channel_id) { |
| media_transport_invoker_->AsyncInvoke<void>( |
| RTC_FROM_HERE, signaling_thread(), [this, channel_id] { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| SignalMediaTransportChannelClosing_s(channel_id); |
| }); |
| } |
| |
| void PeerConnection::OnChannelClosed(int channel_id) { |
| media_transport_invoker_->AsyncInvoke<void>( |
| RTC_FROM_HERE, signaling_thread(), [this, channel_id] { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| SignalMediaTransportChannelClosed_s(channel_id); |
| }); |
| } |
| |
| absl::optional<std::string> PeerConnection::sctp_transport_name() const { |
| if (sctp_mid_ && transport_controller_) { |
| auto dtls_transport = transport_controller_->GetDtlsTransport(*sctp_mid_); |
| 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 { |
| std::map<std::string, std::string> transport_names_by_mid; |
| for (auto transceiver : transceivers_) { |
| cricket::ChannelInterface* channel = transceiver->internal()->channel(); |
| if (channel) { |
| transport_names_by_mid[channel->content_name()] = |
| channel->transport_name(); |
| } |
| } |
| if (rtp_data_channel_) { |
| transport_names_by_mid[rtp_data_channel_->content_name()] = |
| rtp_data_channel_->transport_name(); |
| } |
| if (sctp_transport_) { |
| absl::optional<std::string> transport_name = sctp_transport_name(); |
| RTC_DCHECK(transport_name); |
| transport_names_by_mid[*sctp_mid_] = *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); }); |
| } |
| 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_type_; |
| } |
| |
| bool PeerConnection::IceRestartPending(const std::string& content_name) const { |
| return pending_ice_restarts_.find(content_name) != |
| pending_ice_restarts_.end(); |
| } |
| |
| bool PeerConnection::NeedsIceRestart(const std::string& content_name) const { |
| return transport_controller_->NeedsIceRestart(content_name); |
| } |
| |
| void PeerConnection::OnCertificateReady( |
| const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) { |
| transport_controller_->SetLocalCertificate(certificate); |
| } |
| |
| void PeerConnection::OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp) { |
| SetSessionError(SessionError::kTransport, |
| rtcp ? kDtlsSrtpSetupFailureRtcp : kDtlsSrtpSetupFailureRtp); |
| } |
| |
| void PeerConnection::OnTransportControllerConnectionState( |
| cricket::IceConnectionState state) { |
| switch (state) { |
| case cricket::kIceConnectionConnecting: |
| // If the current state is Connected or Completed, then there were |
| // writable channels but now there are not, so the next state must |
| // be Disconnected. |
| // kIceConnectionConnecting is currently used as the default, |
| // un-connected state by the TransportController, so its only use is |
| // detecting disconnections. |
| if (ice_connection_state_ == |
| PeerConnectionInterface::kIceConnectionConnected || |
| ice_connection_state_ == |
| PeerConnectionInterface::kIceConnectionCompleted) { |
| SetIceConnectionState( |
| PeerConnectionInterface::kIceConnectionDisconnected); |
| } |
| break; |
| case cricket::kIceConnectionFailed: |
| SetIceConnectionState(PeerConnectionInterface::kIceConnectionFailed); |
| break; |
| case cricket::kIceConnectionConnected: |
| RTC_LOG(LS_INFO) << "Changing to ICE connected state because " |
| "all transports are writable."; |
| SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected); |
| 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) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| int sdp_mline_index; |
| if (!GetLocalCandidateMediaIndex(transport_name, &sdp_mline_index)) { |
| RTC_LOG(LS_ERROR) |
| << "OnTransportControllerCandidatesGathered: content name " |
| << transport_name << " not found"; |
| return; |
| } |
| |
| for (cricket::Candidates::const_iterator citer = candidates.begin(); |
| citer != candidates.end(); ++citer) { |
| // Use transport_name as the candidate media id. |
| std::unique_ptr<JsepIceCandidate> candidate( |
| new JsepIceCandidate(transport_name, sdp_mline_index, *citer)); |
| if (local_description()) { |
| mutable_local_description()->AddCandidate(candidate.get()); |
| } |
| OnIceCandidate(std::move(candidate)); |
| } |
| } |
| |
| void PeerConnection::OnTransportControllerCandidatesRemoved( |
| const std::vector<cricket::Candidate>& candidates) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| // Sanity check. |
| for (const cricket::Candidate& candidate : candidates) { |
| if (candidate.transport_name().empty()) { |
| RTC_LOG(LS_ERROR) << "OnTransportControllerCandidatesRemoved: " |
| "empty content name in candidate " |
| << candidate.ToString(); |
| return; |
| } |
| } |
| |
| if (local_description()) { |
| mutable_local_description()->RemoveCandidates(candidates); |
| } |
| OnIceCandidatesRemoved(candidates); |
| } |
| |
| void PeerConnection::OnTransportControllerDtlsHandshakeError( |
| rtc::SSLHandshakeError error) { |
| RTC_HISTOGRAM_ENUMERATION( |
| "WebRTC.PeerConnection.DtlsHandshakeError", static_cast<int>(error), |
| static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE)); |
| } |
| |
| void PeerConnection::EnableSending() { |
| for (auto transceiver : transceivers_) { |
| cricket::ChannelInterface* channel = transceiver->internal()->channel(); |
| if (channel && !channel->enabled()) { |
| channel->Enable(true); |
| } |
| } |
| |
| if (rtp_data_channel_ && !rtp_data_channel_->enabled()) { |
| rtp_data_channel_->Enable(true); |
| } |
| } |
| |
| // Returns the media index for a local ice candidate given the content name. |
| bool PeerConnection::GetLocalCandidateMediaIndex( |
| const std::string& content_name, |
| int* sdp_mline_index) { |
| if (!local_description() || !sdp_mline_index) { |
| return false; |
| } |
| |
| bool content_found = false; |
| const ContentInfos& contents = local_description()->description()->contents(); |
| for (size_t index = 0; index < contents.size(); ++index) { |
| if (contents[index].name == content_name) { |
| *sdp_mline_index = static_cast<int>(index); |
| content_found = true; |
| break; |
| } |
| } |
| return content_found; |
| } |
| |
| bool PeerConnection::UseCandidatesInSessionDescription( |
| const SessionDescriptionInterface* remote_desc) { |
| if (!remote_desc) { |
| return true; |
| } |
| bool ret = true; |
| |
| for (size_t m = 0; m < remote_desc->number_of_mediasections(); ++m) { |
| const IceCandidateCollection* candidates = remote_desc->candidates(m); |
| for (size_t n = 0; n < candidates->count(); ++n) { |
| const IceCandidateInterface* candidate = candidates->at(n); |
| bool valid = false; |
| if (!ReadyToUseRemoteCandidate(candidate, remote_desc, &valid)) { |
| if (valid) { |
| RTC_LOG(LS_INFO) |
| << "UseCandidatesInSessionDescription: Not ready to use " |
| "candidate."; |
| } |
| continue; |
| } |
| ret = UseCandidate(candidate); |
| if (!ret) { |
| break; |
| } |
| } |
| } |
| return ret; |
| } |
| |
| bool PeerConnection::UseCandidate(const IceCandidateInterface* candidate) { |
| size_t mediacontent_index = static_cast<size_t>(candidate->sdp_mline_index()); |
| size_t remote_content_size = |
| remote_description()->description()->contents().size(); |
| if (mediacontent_index >= remote_content_size) { |
| RTC_LOG(LS_ERROR) << "UseCandidate: Invalid candidate media index."; |
| return false; |
| } |
| |
| cricket::ContentInfo content = |
| remote_description()->description()->contents()[mediacontent_index]; |
| std::vector<cricket::Candidate> candidates; |
| candidates.push_back(candidate->candidate()); |
| // Invoking BaseSession method to handle remote candidates. |
| RTCError error = |
| transport_controller_->AddRemoteCandidates(content.name, candidates); |
| if (error.ok()) { |
| // 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; |
| } |
| |
| void PeerConnection::RemoveUnusedChannels(const SessionDescription* desc) { |
| // Destroy video channel first since it may have a pointer to the |
| // voice channel. |
| const cricket::ContentInfo* video_info = cricket::GetFirstVideoContent(desc); |
| if (!video_info || video_info->rejected) { |
| DestroyTransceiverChannel(GetVideoTransceiver()); |
| } |
| |
| const cricket::ContentInfo* audio_info = cricket::GetFirstAudioContent(desc); |
| if (!audio_info || audio_info->rejected) { |
| DestroyTransceiverChannel(GetAudioTransceiver()); |
| } |
| |
| const cricket::ContentInfo* data_info = cricket::GetFirstDataContent(desc); |
| if (!data_info || data_info->rejected) { |
| DestroyDataChannel(); |
| } |
| } |
| |
| 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 std::move(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 && |
| !rtp_data_channel_ && !sctp_transport_ && !media_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); |
| MediaTransportInterface* media_transport = nullptr; |
| if (configuration_.use_media_transport) { |
| media_transport = GetMediaTransport(mid); |
| } |
| |
| cricket::VoiceChannel* voice_channel = channel_manager()->CreateVoiceChannel( |
| call_.get(), configuration_.media_config, rtp_transport, media_transport, |
| signaling_thread(), mid, SrtpRequired(), GetCryptoOptions(), |
| 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); |
| MediaTransportInterface* media_transport = nullptr; |
| if (configuration_.use_media_transport) { |
| media_transport = GetMediaTransport(mid); |
| } |
| |
| cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel( |
| call_.get(), configuration_.media_config, rtp_transport, media_transport, |
| signaling_thread(), mid, SrtpRequired(), GetCryptoOptions(), |
| video_options_); |
| 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_MEDIA_TRANSPORT: |
| if (network_thread()->Invoke<bool>( |
| RTC_FROM_HERE, |
| rtc::Bind(&PeerConnection::SetupMediaTransportForDataChannels_n, |
| this, mid))) { |
| for (const auto& channel : sctp_data_channels_) { |
| channel->OnTransportChannelCreated(); |
| } |
| return true; |
| } |
| return false; |
| case cricket::DCT_SCTP: |
| if (!sctp_factory_) { |
| RTC_LOG(LS_ERROR) |
| << "Trying to create SCTP transport, but didn't compile with " |
| "SCTP support (HAVE_SCTP)"; |
| return false; |
| } |
| if (!network_thread()->Invoke<bool>( |
| RTC_FROM_HERE, |
| rtc::Bind(&PeerConnection::CreateSctpTransport_n, this, mid))) { |
| return false; |
| } |
| for (const auto& channel : sctp_data_channels_) { |
| channel->OnTransportChannelCreated(); |
| } |
| return true; |
| case cricket::DCT_RTP: |
| default: |
| RtpTransportInternal* rtp_transport = GetRtpTransport(mid); |
| rtp_data_channel_ = channel_manager()->CreateRtpDataChannel( |
| configuration_.media_config, rtp_transport, signaling_thread(), mid, |
| SrtpRequired(), GetCryptoOptions()); |
| if (!rtp_data_channel_) { |
| return false; |
| } |
| rtp_data_channel_->SignalDtlsSrtpSetupFailure.connect( |
| this, &PeerConnection::OnDtlsSrtpSetupFailure); |
| rtp_data_channel_->SignalSentPacket.connect( |
| this, &PeerConnection::OnSentPacket_w); |
| rtp_data_channel_->SetRtpTransport(rtp_transport); |
| return true; |
| } |
| |
| return true; |
| } |
| |
| Call::Stats PeerConnection::GetCallStats() { |
| if (!worker_thread()->IsCurrent()) { |
| return worker_thread()->Invoke<Call::Stats>( |
| RTC_FROM_HERE, rtc::Bind(&PeerConnection::GetCallStats, this)); |
| } |
| if (call_) { |
| return call_->GetStats(); |
| } else { |
| return Call::Stats(); |
| } |
| } |
| |
| bool PeerConnection::CreateSctpTransport_n(const std::string& mid) { |
| RTC_DCHECK(network_thread()->IsCurrent()); |
| RTC_DCHECK(sctp_factory_); |
| cricket::DtlsTransportInternal* dtls_transport = |
| transport_controller_->GetDtlsTransport(mid); |
| RTC_DCHECK(dtls_transport); |
| sctp_transport_ = sctp_factory_->CreateSctpTransport(dtls_transport); |
| RTC_DCHECK(sctp_transport_); |
| sctp_invoker_.reset(new rtc::AsyncInvoker()); |
| sctp_transport_->SignalReadyToSendData.connect( |
| this, &PeerConnection::OnSctpTransportReadyToSendData_n); |
| sctp_transport_->SignalDataReceived.connect( |
| this, &PeerConnection::OnSctpTransportDataReceived_n); |
| // TODO(deadbeef): All we do here is AsyncInvoke to fire the signal on |
| // another thread. Would be nice if there was a helper class similar to |
| // sigslot::repeater that did this for us, eliminating a bunch of boilerplate |
| // code. |
| sctp_transport_->SignalClosingProcedureStartedRemotely.connect( |
| this, &PeerConnection::OnSctpClosingProcedureStartedRemotely_n); |
| sctp_transport_->SignalClosingProcedureComplete.connect( |
| this, &PeerConnection::OnSctpClosingProcedureComplete_n); |
| sctp_mid_ = mid; |
| sctp_transport_->SetDtlsTransport(dtls_transport); |
| return true; |
| } |
| |
| void PeerConnection::DestroySctpTransport_n() { |
| RTC_DCHECK(network_thread()->IsCurrent()); |
| sctp_transport_.reset(nullptr); |
| sctp_mid_.reset(); |
| sctp_invoker_.reset(nullptr); |
| sctp_ready_to_send_data_ = false; |
| } |
| |
| void PeerConnection::OnSctpTransportReadyToSendData_n() { |
| RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP); |
| RTC_DCHECK(network_thread()->IsCurrent()); |
| // Note: Cannot use rtc::Bind here because it will grab a reference to |
| // PeerConnection and potentially cause PeerConnection to live longer than |
| // expected. It is safe not to grab a reference since the sctp_invoker_ will |
| // be destroyed before PeerConnection is destroyed, and at that point all |
| // pending tasks will be cleared. |
| sctp_invoker_->AsyncInvoke<void>(RTC_FROM_HERE, signaling_thread(), [this] { |
| OnSctpTransportReadyToSendData_s(true); |
| }); |
| } |
| |
| void PeerConnection::OnSctpTransportReadyToSendData_s(bool ready) { |
| RTC_DCHECK(signaling_thread()->IsCurrent()); |
| sctp_ready_to_send_data_ = ready; |
| SignalSctpReadyToSendData(ready); |
| } |
| |
| void PeerConnection::OnSctpTransportDataReceived_n( |
| const cricket::ReceiveDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload) { |
| RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP); |
| RTC_DCHECK(network_thread()->IsCurrent()); |
| // Note: Cannot use rtc::Bind here because it will grab a reference to |
| // PeerConnection and potentially cause PeerConnection to live longer than |
| // expected. It is safe not to grab a reference since the sctp_invoker_ will |
| // be destroyed before PeerConnection is destroyed, and at that point all |
| // pending tasks will be cleared. |
| sctp_invoker_->AsyncInvoke<void>( |
| RTC_FROM_HERE, signaling_thread(), [this, params, payload] { |
| OnSctpTransportDataReceived_s(params, payload); |
| }); |
| } |
| |
| void PeerConnection::OnSctpTransportDataReceived_s( |
| const cricket::ReceiveDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload) { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| if (!HandleOpenMessage_s(params, payload)) { |
| SignalSctpDataReceived(params, payload); |
| } |
| } |
| |
| void PeerConnection::OnSctpClosingProcedureStartedRemotely_n(int sid) { |
| RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP); |
| RTC_DCHECK(network_thread()->IsCurrent()); |
| sctp_invoker_->AsyncInvoke<void>( |
| RTC_FROM_HERE, signaling_thread(), |
| rtc::Bind(&sigslot::signal1<int>::operator(), |
| &SignalSctpClosingProcedureStartedRemotely, sid)); |
| } |
| |
| void PeerConnection::OnSctpClosingProcedureComplete_n(int sid) { |
| RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP); |
| RTC_DCHECK(network_thread()->IsCurrent()); |
| sctp_invoker_->AsyncInvoke<void>( |
| RTC_FROM_HERE, signaling_thread(), |
| rtc::Bind(&sigslot::signal1<int>::operator(), |
| &SignalSctpClosingProcedureComplete, sid)); |
| } |
| |
| bool PeerConnection::SetupMediaTransportForDataChannels_n( |
| const std::string& mid) { |
| media_transport_ = transport_controller_->GetMediaTransport(mid); |
| if (!media_transport_) { |
| RTC_LOG(LS_ERROR) << "Media transport is not available for data channels"; |
| return false; |
| } |
| |
| media_transport_invoker_ = absl::make_unique<rtc::AsyncInvoker>(); |
| media_transport_->SetDataSink(this); |
| media_transport_data_mid_ = mid; |
| transport_controller_->SignalMediaTransportStateChanged.connect( |
| this, &PeerConnection::OnMediaTransportStateChanged_n); |
| // Check the initial state right away, in case transport is already writable. |
| OnMediaTransportStateChanged_n(); |
| return true; |
| } |
| |
| void PeerConnection::TeardownMediaTransportForDataChannels_n() { |
| if (!media_transport_) { |
| return; |
| } |
| transport_controller_->SignalMediaTransportStateChanged.disconnect(this); |
| media_transport_data_mid_.reset(); |
| media_transport_->SetDataSink(nullptr); |
| media_transport_invoker_ = nullptr; |
| media_transport_ = nullptr; |
| } |
| |
| void PeerConnection::OnMediaTransportStateChanged_n() { |
| if (!media_transport_data_mid_ || |
| transport_controller_->GetMediaTransportState( |
| *media_transport_data_mid_) != MediaTransportState::kWritable) { |
| return; |
| } |
| media_transport_invoker_->AsyncInvoke<void>( |
| RTC_FROM_HERE, signaling_thread(), [this] { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| media_transport_ready_to_send_data_ = true; |
| SignalMediaTransportWritable_s(media_transport_ready_to_send_data_); |
| }); |
| } |
| |
| // 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.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::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_CALLED) | |
| static_cast<int>(UsageEvent::CANDIDATE_COLLECTED); |
| const int good_bits = |
| static_cast<int>(UsageEvent::SET_REMOTE_DESCRIPTION_CALLED) | |
| 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; |
| } |
| |
| size_t mediacontent_index = static_cast<size_t>(candidate->sdp_mline_index()); |
| size_t remote_content_size = |
| current_remote_desc->description()->contents().size(); |
| if (mediacontent_index >= remote_content_size) { |
| RTC_LOG(LS_ERROR) |
| << "ReadyToUseRemoteCandidate: Invalid candidate media index " |
| << mediacontent_index; |
| |
| *valid = false; |
| return false; |
| } |
| |
| cricket::ContentInfo content = |
| current_remote_desc->description()->contents()[mediacontent_index]; |
| |
| const std::string transport_name = GetTransportName(content.name); |
| if (transport_name.empty()) { |
| return false; |
| } |
| return true; |
| } |
| |
| 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 (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.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(worker_thread()->IsCurrent()); |
| 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 (sctp_transport_) { |
| RTC_DCHECK(sctp_mid_); |
| if (content_name == *sctp_mid_) { |
| 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::DestroyDataChannel() { |
| if (rtp_data_channel_) { |
| OnDataChannelDestroyed(); |
| DestroyChannelInterface(rtp_data_channel_); |
| rtp_data_channel_ = nullptr; |
| } |
| |
| // Note: Cannot use rtc::Bind to create a functor to invoke because it will |
| // grab a reference to this PeerConnection. If this is called from the |
| // PeerConnection destructor, the RefCountedObject vtable will have already |
| // been destroyed (since it is a subclass of PeerConnection) and using |
| // rtc::Bind will cause "Pure virtual function called" error to appear. |
| |
| if (sctp_transport_) { |
| OnDataChannelDestroyed(); |
| network_thread()->Invoke<void>(RTC_FROM_HERE, |
| [this] { DestroySctpTransport_n(); }); |
| } |
| |
| if (media_transport_) { |
| OnDataChannelDestroyed(); |
| network_thread()->Invoke<void>(RTC_FROM_HERE, [this] { |
| RTC_DCHECK_RUN_ON(network_thread()); |
| TeardownMediaTransportForDataChannels_n(); |
| }); |
| } |
| } |
| |
| 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, |
| cricket::DtlsTransportInternal* dtls_transport, |
| MediaTransportInterface* media_transport) { |
| bool ret = true; |
| auto base_channel = GetChannel(mid); |
| if (base_channel) { |
| ret = base_channel->SetRtpTransport(rtp_transport); |
| } |
| if (sctp_transport_ && mid == sctp_mid_) { |
| sctp_transport_->SetDtlsTransport(dtls_transport); |
| } |
| |
| call_->MediaTransportChange(media_transport); |
| |
| return ret; |
| } |
| |
| PeerConnectionObserver* PeerConnection::Observer() const { |
| // In earlier production code, the pointer was not cleared on close, |
| // which might have led to undefined behavior if the observer was not |
| // deallocated, or strange crashes if it was. |
| // We use CHECK in order to catch such behavior if it exists. |
| // TODO(hta): Remove or replace with DCHECK if nothing is found. |
| RTC_CHECK(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() { |
| if (stats_collector_) { |
| stats_collector_->ClearCachedStatsReport(); |
| } |
| } |
| |
| void PeerConnection::RequestUsagePatternReportForTesting() { |
| signaling_thread()->Post(RTC_FROM_HERE, this, MSG_REPORT_USAGE_PATTERN, |
| nullptr); |
| } |
| |
| } // namespace webrtc |