4919 lines
199 KiB
C++
4919 lines
199 KiB
C++
/*
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* Copyright 2020 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/sdp_offer_answer.h"
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#include <algorithm>
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#include <iterator>
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#include <map>
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#include <queue>
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#include <type_traits>
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#include <utility>
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#include "absl/algorithm/container.h"
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#include "absl/memory/memory.h"
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#include "absl/strings/string_view.h"
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#include "api/array_view.h"
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#include "api/crypto/crypto_options.h"
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#include "api/dtls_transport_interface.h"
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#include "api/media_stream_proxy.h"
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#include "api/rtp_parameters.h"
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#include "api/rtp_receiver_interface.h"
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#include "api/rtp_sender_interface.h"
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#include "api/video/builtin_video_bitrate_allocator_factory.h"
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#include "media/base/codec.h"
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#include "media/base/media_engine.h"
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#include "media/base/rid_description.h"
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#include "p2p/base/ice_transport_internal.h"
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#include "p2p/base/p2p_constants.h"
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#include "p2p/base/p2p_transport_channel.h"
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#include "p2p/base/port.h"
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#include "p2p/base/transport_description.h"
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#include "p2p/base/transport_description_factory.h"
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#include "p2p/base/transport_info.h"
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#include "pc/data_channel_utils.h"
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#include "pc/dtls_transport.h"
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#include "pc/media_stream.h"
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#include "pc/peer_connection.h"
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#include "pc/peer_connection_message_handler.h"
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#include "pc/rtp_media_utils.h"
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#include "pc/rtp_sender.h"
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#include "pc/rtp_transport_internal.h"
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#include "pc/simulcast_description.h"
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#include "pc/stats_collector.h"
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#include "pc/usage_pattern.h"
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#include "pc/webrtc_session_description_factory.h"
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#include "rtc_base/helpers.h"
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#include "rtc_base/location.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/ref_counted_object.h"
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#include "rtc_base/rtc_certificate.h"
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#include "rtc_base/ssl_stream_adapter.h"
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#include "rtc_base/string_encode.h"
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#include "rtc_base/strings/string_builder.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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#include "rtc_base/trace_event.h"
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#include "system_wrappers/include/metrics.h"
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using cricket::ContentInfo;
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using cricket::ContentInfos;
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using cricket::MediaContentDescription;
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using cricket::MediaProtocolType;
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using cricket::RidDescription;
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using cricket::RidDirection;
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using cricket::SessionDescription;
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using cricket::SimulcastDescription;
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using cricket::SimulcastLayer;
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using cricket::SimulcastLayerList;
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using cricket::StreamParams;
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using cricket::TransportInfo;
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using cricket::LOCAL_PORT_TYPE;
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using cricket::PRFLX_PORT_TYPE;
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using cricket::RELAY_PORT_TYPE;
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using cricket::STUN_PORT_TYPE;
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namespace webrtc {
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namespace {
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typedef webrtc::PeerConnectionInterface::RTCOfferAnswerOptions
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RTCOfferAnswerOptions;
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// Error messages
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const char kInvalidSdp[] = "Invalid session description.";
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const char kInvalidCandidates[] = "Description contains invalid candidates.";
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const char kBundleWithoutRtcpMux[] =
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"rtcp-mux must be enabled when BUNDLE "
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"is enabled.";
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const char kMlineMismatchInAnswer[] =
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"The order of m-lines in answer doesn't match order in offer. Rejecting "
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"answer.";
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const char kMlineMismatchInSubsequentOffer[] =
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"The order of m-lines in subsequent offer doesn't match order from "
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"previous offer/answer.";
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const char kSdpWithoutIceUfragPwd[] =
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"Called with SDP without ice-ufrag and ice-pwd.";
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const char kSdpWithoutDtlsFingerprint[] =
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"Called with SDP without DTLS fingerprint.";
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const char kSdpWithoutSdesCrypto[] = "Called with SDP without SDES crypto.";
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const char kSessionError[] = "Session error code: ";
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const char kSessionErrorDesc[] = "Session error description: ";
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// UMA metric names.
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const char kSimulcastVersionApplyLocalDescription[] =
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"WebRTC.PeerConnection.Simulcast.ApplyLocalDescription";
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const char kSimulcastVersionApplyRemoteDescription[] =
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"WebRTC.PeerConnection.Simulcast.ApplyRemoteDescription";
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const char kSimulcastDisabled[] = "WebRTC.PeerConnection.Simulcast.Disabled";
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// The length of RTCP CNAMEs.
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static const int kRtcpCnameLength = 16;
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const char kDefaultStreamId[] = "default";
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// NOTE: Duplicated in peer_connection.cc:
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static const char kDefaultAudioSenderId[] = "defaulta0";
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static const char kDefaultVideoSenderId[] = "defaultv0";
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void NoteAddIceCandidateResult(int result) {
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RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.AddIceCandidate", result,
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kAddIceCandidateMax);
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}
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void NoteKeyProtocolAndMedia(KeyExchangeProtocolType protocol_type,
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cricket::MediaType media_type) {
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// Array of structs needed to map {KeyExchangeProtocolType,
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// cricket::MediaType} to KeyExchangeProtocolMedia without using std::map in
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// order to avoid -Wglobal-constructors and -Wexit-time-destructors.
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static constexpr struct {
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KeyExchangeProtocolType protocol_type;
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cricket::MediaType media_type;
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KeyExchangeProtocolMedia protocol_media;
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} kEnumCounterKeyProtocolMediaMap[] = {
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{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_AUDIO,
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kEnumCounterKeyProtocolMediaTypeDtlsAudio},
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{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_VIDEO,
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kEnumCounterKeyProtocolMediaTypeDtlsVideo},
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{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_DATA,
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kEnumCounterKeyProtocolMediaTypeDtlsData},
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{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_AUDIO,
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kEnumCounterKeyProtocolMediaTypeSdesAudio},
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{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_VIDEO,
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kEnumCounterKeyProtocolMediaTypeSdesVideo},
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{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_DATA,
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kEnumCounterKeyProtocolMediaTypeSdesData},
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};
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RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocol", protocol_type,
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kEnumCounterKeyProtocolMax);
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for (const auto& i : kEnumCounterKeyProtocolMediaMap) {
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if (i.protocol_type == protocol_type && i.media_type == media_type) {
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RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocolByMedia",
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i.protocol_media,
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kEnumCounterKeyProtocolMediaTypeMax);
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}
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}
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}
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std::map<std::string, const cricket::ContentGroup*> GetBundleGroupsByMid(
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const SessionDescription* desc) {
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std::vector<const cricket::ContentGroup*> bundle_groups =
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desc->GetGroupsByName(cricket::GROUP_TYPE_BUNDLE);
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std::map<std::string, const cricket::ContentGroup*> bundle_groups_by_mid;
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for (const cricket::ContentGroup* bundle_group : bundle_groups) {
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for (const std::string& content_name : bundle_group->content_names()) {
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bundle_groups_by_mid[content_name] = bundle_group;
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}
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}
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return bundle_groups_by_mid;
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}
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// Returns true if |new_desc| requests an ICE restart (i.e., new ufrag/pwd).
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bool CheckForRemoteIceRestart(const SessionDescriptionInterface* old_desc,
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const SessionDescriptionInterface* new_desc,
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const std::string& content_name) {
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if (!old_desc) {
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return false;
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}
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const SessionDescription* new_sd = new_desc->description();
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const SessionDescription* old_sd = old_desc->description();
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const ContentInfo* cinfo = new_sd->GetContentByName(content_name);
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if (!cinfo || cinfo->rejected) {
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return false;
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}
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// If the content isn't rejected, check if ufrag and password has changed.
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const cricket::TransportDescription* new_transport_desc =
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new_sd->GetTransportDescriptionByName(content_name);
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const cricket::TransportDescription* old_transport_desc =
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old_sd->GetTransportDescriptionByName(content_name);
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if (!new_transport_desc || !old_transport_desc) {
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// No transport description exists. This is not an ICE restart.
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return false;
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}
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if (cricket::IceCredentialsChanged(
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old_transport_desc->ice_ufrag, old_transport_desc->ice_pwd,
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new_transport_desc->ice_ufrag, new_transport_desc->ice_pwd)) {
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RTC_LOG(LS_INFO) << "Remote peer requests ICE restart for " << content_name
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<< ".";
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return true;
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}
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return false;
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}
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// Generates a string error message for SetLocalDescription/SetRemoteDescription
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// from an RTCError.
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std::string GetSetDescriptionErrorMessage(cricket::ContentSource source,
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SdpType type,
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const RTCError& error) {
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rtc::StringBuilder oss;
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oss << "Failed to set " << (source == cricket::CS_LOCAL ? "local" : "remote")
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<< " " << SdpTypeToString(type) << " sdp: " << error.message();
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return oss.Release();
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}
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std::string GetStreamIdsString(rtc::ArrayView<const std::string> stream_ids) {
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std::string output = "streams=[";
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const char* separator = "";
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for (const auto& stream_id : stream_ids) {
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output.append(separator).append(stream_id);
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separator = ", ";
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}
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output.append("]");
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return output;
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}
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void ReportSimulcastApiVersion(const char* name,
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const SessionDescription& session) {
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bool has_legacy = false;
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bool has_spec_compliant = false;
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for (const ContentInfo& content : session.contents()) {
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if (!content.media_description()) {
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continue;
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}
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has_spec_compliant |= content.media_description()->HasSimulcast();
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for (const StreamParams& sp : content.media_description()->streams()) {
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has_legacy |= sp.has_ssrc_group(cricket::kSimSsrcGroupSemantics);
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}
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}
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if (has_legacy) {
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RTC_HISTOGRAM_ENUMERATION(name, kSimulcastApiVersionLegacy,
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kSimulcastApiVersionMax);
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}
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if (has_spec_compliant) {
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RTC_HISTOGRAM_ENUMERATION(name, kSimulcastApiVersionSpecCompliant,
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kSimulcastApiVersionMax);
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}
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if (!has_legacy && !has_spec_compliant) {
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RTC_HISTOGRAM_ENUMERATION(name, kSimulcastApiVersionNone,
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kSimulcastApiVersionMax);
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}
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}
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const ContentInfo* FindTransceiverMSection(
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RtpTransceiver* transceiver,
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const SessionDescriptionInterface* session_description) {
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return transceiver->mid()
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? session_description->description()->GetContentByName(
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*transceiver->mid())
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: nullptr;
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}
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// If the direction is "recvonly" or "inactive", treat the description
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// as containing no streams.
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// See: https://code.google.com/p/webrtc/issues/detail?id=5054
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std::vector<cricket::StreamParams> GetActiveStreams(
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const cricket::MediaContentDescription* desc) {
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return RtpTransceiverDirectionHasSend(desc->direction())
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? desc->streams()
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: std::vector<cricket::StreamParams>();
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}
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// Logic to decide if an m= section can be recycled. This means that the new
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// m= section is not rejected, but the old local or remote m= section is
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// rejected. |old_content_one| and |old_content_two| refer to the m= section
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// of the old remote and old local descriptions in no particular order.
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// We need to check both the old local and remote because either
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// could be the most current from the latest negotation.
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bool IsMediaSectionBeingRecycled(SdpType type,
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const ContentInfo& content,
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const ContentInfo* old_content_one,
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const ContentInfo* old_content_two) {
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return type == SdpType::kOffer && !content.rejected &&
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((old_content_one && old_content_one->rejected) ||
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(old_content_two && old_content_two->rejected));
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}
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// Verify that the order of media sections in |new_desc| matches
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// |current_desc|. The number of m= sections in |new_desc| should be no
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// less than |current_desc|. In the case of checking an answer's
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// |new_desc|, the |current_desc| is the last offer that was set as the
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// local or remote. In the case of checking an offer's |new_desc| we
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// check against the local and remote descriptions stored from the last
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// negotiation, because either of these could be the most up to date for
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// possible rejected m sections. These are the |current_desc| and
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// |secondary_current_desc|.
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bool MediaSectionsInSameOrder(const SessionDescription& current_desc,
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const SessionDescription* secondary_current_desc,
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const SessionDescription& new_desc,
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const SdpType type) {
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if (current_desc.contents().size() > new_desc.contents().size()) {
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return false;
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}
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for (size_t i = 0; i < current_desc.contents().size(); ++i) {
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const cricket::ContentInfo* secondary_content_info = nullptr;
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if (secondary_current_desc &&
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i < secondary_current_desc->contents().size()) {
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secondary_content_info = &secondary_current_desc->contents()[i];
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}
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if (IsMediaSectionBeingRecycled(type, new_desc.contents()[i],
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¤t_desc.contents()[i],
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secondary_content_info)) {
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// For new offer descriptions, if the media section can be recycled, it's
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// valid for the MID and media type to change.
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continue;
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}
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if (new_desc.contents()[i].name != current_desc.contents()[i].name) {
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return false;
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}
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const MediaContentDescription* new_desc_mdesc =
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new_desc.contents()[i].media_description();
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const MediaContentDescription* current_desc_mdesc =
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current_desc.contents()[i].media_description();
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if (new_desc_mdesc->type() != current_desc_mdesc->type()) {
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return false;
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}
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}
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return true;
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}
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bool MediaSectionsHaveSameCount(const SessionDescription& desc1,
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const SessionDescription& desc2) {
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return desc1.contents().size() == desc2.contents().size();
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}
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// Checks that each non-rejected content has SDES crypto keys or a DTLS
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// fingerprint, unless it's in a BUNDLE group, in which case only the
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// BUNDLE-tag section (first media section/description in the BUNDLE group)
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// needs a ufrag and pwd. Mismatches, such as replying with a DTLS fingerprint
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// to SDES keys, will be caught in JsepTransport negotiation, and backstopped
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// by Channel's |srtp_required| check.
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RTCError VerifyCrypto(const SessionDescription* desc,
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bool dtls_enabled,
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const std::map<std::string, const cricket::ContentGroup*>&
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bundle_groups_by_mid) {
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for (const cricket::ContentInfo& content_info : desc->contents()) {
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if (content_info.rejected) {
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continue;
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}
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// Note what media is used with each crypto protocol, for all sections.
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NoteKeyProtocolAndMedia(dtls_enabled ? webrtc::kEnumCounterKeyProtocolDtls
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: webrtc::kEnumCounterKeyProtocolSdes,
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content_info.media_description()->type());
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const std::string& mid = content_info.name;
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auto it = bundle_groups_by_mid.find(mid);
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const cricket::ContentGroup* bundle =
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it != bundle_groups_by_mid.end() ? it->second : nullptr;
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if (bundle && mid != *(bundle->FirstContentName())) {
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// This isn't the first media section in the BUNDLE group, so it's not
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// required to have crypto attributes, since only the crypto attributes
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// from the first section actually get used.
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continue;
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}
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// If the content isn't rejected or bundled into another m= section, crypto
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// must be present.
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const MediaContentDescription* media = content_info.media_description();
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const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
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if (!media || !tinfo) {
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// Something is not right.
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LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
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}
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if (dtls_enabled) {
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if (!tinfo->description.identity_fingerprint) {
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RTC_LOG(LS_WARNING)
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<< "Session description must have DTLS fingerprint if "
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"DTLS enabled.";
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return RTCError(RTCErrorType::INVALID_PARAMETER,
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kSdpWithoutDtlsFingerprint);
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}
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} else {
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if (media->cryptos().empty()) {
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RTC_LOG(LS_WARNING)
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<< "Session description must have SDES when DTLS disabled.";
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return RTCError(RTCErrorType::INVALID_PARAMETER, kSdpWithoutSdesCrypto);
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}
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}
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}
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return RTCError::OK();
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}
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// Checks that each non-rejected content has ice-ufrag and ice-pwd set, unless
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// it's in a BUNDLE group, in which case only the BUNDLE-tag section (first
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// media section/description in the BUNDLE group) needs a ufrag and pwd.
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bool VerifyIceUfragPwdPresent(
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const SessionDescription* desc,
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const std::map<std::string, const cricket::ContentGroup*>&
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bundle_groups_by_mid) {
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for (const cricket::ContentInfo& content_info : desc->contents()) {
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if (content_info.rejected) {
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continue;
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}
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const std::string& mid = content_info.name;
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auto it = bundle_groups_by_mid.find(mid);
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const cricket::ContentGroup* bundle =
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it != bundle_groups_by_mid.end() ? it->second : nullptr;
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if (bundle && mid != *(bundle->FirstContentName())) {
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// This isn't the first media section in the BUNDLE group, so it's not
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// required to have ufrag/password, since only the ufrag/password from
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// the first section actually get used.
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continue;
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}
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// If the content isn't rejected or bundled into another m= section,
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// ice-ufrag and ice-pwd must be present.
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const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
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if (!tinfo) {
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// Something is not right.
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RTC_LOG(LS_ERROR) << kInvalidSdp;
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return false;
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}
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if (tinfo->description.ice_ufrag.empty() ||
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tinfo->description.ice_pwd.empty()) {
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RTC_LOG(LS_ERROR) << "Session description must have ice ufrag and pwd.";
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return false;
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}
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}
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return true;
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}
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RTCError ValidateMids(const cricket::SessionDescription& description) {
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std::set<std::string> mids;
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for (const cricket::ContentInfo& content : description.contents()) {
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if (content.name.empty()) {
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LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
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"A media section is missing a MID attribute.");
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}
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if (!mids.insert(content.name).second) {
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LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
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"Duplicate a=mid value '" + content.name + "'.");
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}
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}
|
|
return RTCError::OK();
|
|
}
|
|
|
|
bool IsValidOfferToReceiveMedia(int value) {
|
|
typedef PeerConnectionInterface::RTCOfferAnswerOptions Options;
|
|
return (value >= Options::kUndefined) &&
|
|
(value <= Options::kMaxOfferToReceiveMedia);
|
|
}
|
|
|
|
bool ValidateOfferAnswerOptions(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options) {
|
|
return IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_audio) &&
|
|
IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_video);
|
|
}
|
|
|
|
// Map internal signaling state name to spec name:
|
|
// https://w3c.github.io/webrtc-pc/#rtcsignalingstate-enum
|
|
std::string GetSignalingStateString(
|
|
PeerConnectionInterface::SignalingState state) {
|
|
switch (state) {
|
|
case PeerConnectionInterface::kStable:
|
|
return "stable";
|
|
case PeerConnectionInterface::kHaveLocalOffer:
|
|
return "have-local-offer";
|
|
case PeerConnectionInterface::kHaveLocalPrAnswer:
|
|
return "have-local-pranswer";
|
|
case PeerConnectionInterface::kHaveRemoteOffer:
|
|
return "have-remote-offer";
|
|
case PeerConnectionInterface::kHaveRemotePrAnswer:
|
|
return "have-remote-pranswer";
|
|
case PeerConnectionInterface::kClosed:
|
|
return "closed";
|
|
}
|
|
RTC_NOTREACHED();
|
|
return "";
|
|
}
|
|
|
|
// This method will extract any send encodings that were sent by the remote
|
|
// connection. This is currently only relevant for Simulcast scenario (where
|
|
// the number of layers may be communicated by the server).
|
|
std::vector<RtpEncodingParameters> GetSendEncodingsFromRemoteDescription(
|
|
const MediaContentDescription& desc) {
|
|
if (!desc.HasSimulcast()) {
|
|
return {};
|
|
}
|
|
std::vector<RtpEncodingParameters> result;
|
|
const SimulcastDescription& simulcast = desc.simulcast_description();
|
|
|
|
// This is a remote description, the parameters we are after should appear
|
|
// as receive streams.
|
|
for (const auto& alternatives : simulcast.receive_layers()) {
|
|
RTC_DCHECK(!alternatives.empty());
|
|
// There is currently no way to specify or choose from alternatives.
|
|
// We will always use the first alternative, which is the most preferred.
|
|
const SimulcastLayer& layer = alternatives[0];
|
|
RtpEncodingParameters parameters;
|
|
parameters.rid = layer.rid;
|
|
parameters.active = !layer.is_paused;
|
|
result.push_back(parameters);
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
RTCError UpdateSimulcastLayerStatusInSender(
|
|
const std::vector<SimulcastLayer>& layers,
|
|
rtc::scoped_refptr<RtpSenderInternal> sender) {
|
|
RTC_DCHECK(sender);
|
|
RtpParameters parameters = sender->GetParametersInternal();
|
|
std::vector<std::string> disabled_layers;
|
|
|
|
// The simulcast envelope cannot be changed, only the status of the streams.
|
|
// So we will iterate over the send encodings rather than the layers.
|
|
for (RtpEncodingParameters& encoding : parameters.encodings) {
|
|
auto iter = std::find_if(layers.begin(), layers.end(),
|
|
[&encoding](const SimulcastLayer& layer) {
|
|
return layer.rid == encoding.rid;
|
|
});
|
|
// A layer that cannot be found may have been removed by the remote party.
|
|
if (iter == layers.end()) {
|
|
disabled_layers.push_back(encoding.rid);
|
|
continue;
|
|
}
|
|
|
|
encoding.active = !iter->is_paused;
|
|
}
|
|
|
|
RTCError result = sender->SetParametersInternal(parameters);
|
|
if (result.ok()) {
|
|
result = sender->DisableEncodingLayers(disabled_layers);
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
bool SimulcastIsRejected(const ContentInfo* local_content,
|
|
const MediaContentDescription& answer_media_desc) {
|
|
bool simulcast_offered = local_content &&
|
|
local_content->media_description() &&
|
|
local_content->media_description()->HasSimulcast();
|
|
bool simulcast_answered = answer_media_desc.HasSimulcast();
|
|
bool rids_supported = RtpExtension::FindHeaderExtensionByUri(
|
|
answer_media_desc.rtp_header_extensions(), RtpExtension::kRidUri);
|
|
return simulcast_offered && (!simulcast_answered || !rids_supported);
|
|
}
|
|
|
|
RTCError DisableSimulcastInSender(
|
|
rtc::scoped_refptr<RtpSenderInternal> sender) {
|
|
RTC_DCHECK(sender);
|
|
RtpParameters parameters = sender->GetParametersInternal();
|
|
if (parameters.encodings.size() <= 1) {
|
|
return RTCError::OK();
|
|
}
|
|
|
|
std::vector<std::string> disabled_layers;
|
|
std::transform(
|
|
parameters.encodings.begin() + 1, parameters.encodings.end(),
|
|
std::back_inserter(disabled_layers),
|
|
[](const RtpEncodingParameters& encoding) { return encoding.rid; });
|
|
return sender->DisableEncodingLayers(disabled_layers);
|
|
}
|
|
|
|
// The SDP parser used to populate these values by default for the 'content
|
|
// name' if an a=mid line was absent.
|
|
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;
|
|
case cricket::MEDIA_TYPE_UNSUPPORTED:
|
|
return "not supported";
|
|
}
|
|
RTC_NOTREACHED();
|
|
return "";
|
|
}
|
|
|
|
// 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);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
cricket::MediaDescriptionOptions GetMediaDescriptionOptionsForTransceiver(
|
|
RtpTransceiver* transceiver,
|
|
const std::string& mid,
|
|
bool is_create_offer) {
|
|
// NOTE: a stopping transceiver should be treated as a stopped one in
|
|
// createOffer as specified in
|
|
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-createoffer.
|
|
bool stopped =
|
|
is_create_offer ? transceiver->stopping() : transceiver->stopped();
|
|
cricket::MediaDescriptionOptions media_description_options(
|
|
transceiver->media_type(), mid, transceiver->direction(), stopped);
|
|
media_description_options.codec_preferences =
|
|
transceiver->codec_preferences();
|
|
media_description_options.header_extensions =
|
|
transceiver->HeaderExtensionsToOffer();
|
|
// This behavior is specified in JSEP. The gist is that:
|
|
// 1. The MSID is included if the RtpTransceiver's direction is sendonly or
|
|
// sendrecv.
|
|
// 2. If the MSID is included, then it must be included in any subsequent
|
|
// offer/answer exactly the same until the RtpTransceiver is stopped.
|
|
if (stopped || (!RtpTransceiverDirectionHasSend(transceiver->direction()) &&
|
|
!transceiver->has_ever_been_used_to_send())) {
|
|
return media_description_options;
|
|
}
|
|
|
|
cricket::SenderOptions sender_options;
|
|
sender_options.track_id = transceiver->sender()->id();
|
|
sender_options.stream_ids = transceiver->sender()->stream_ids();
|
|
|
|
// The following sets up RIDs and Simulcast.
|
|
// RIDs are included if Simulcast is requested or if any RID was specified.
|
|
RtpParameters send_parameters =
|
|
transceiver->sender_internal()->GetParametersInternal();
|
|
bool has_rids = std::any_of(send_parameters.encodings.begin(),
|
|
send_parameters.encodings.end(),
|
|
[](const RtpEncodingParameters& encoding) {
|
|
return !encoding.rid.empty();
|
|
});
|
|
|
|
std::vector<RidDescription> send_rids;
|
|
SimulcastLayerList send_layers;
|
|
for (const RtpEncodingParameters& encoding : send_parameters.encodings) {
|
|
if (encoding.rid.empty()) {
|
|
continue;
|
|
}
|
|
send_rids.push_back(RidDescription(encoding.rid, RidDirection::kSend));
|
|
send_layers.AddLayer(SimulcastLayer(encoding.rid, !encoding.active));
|
|
}
|
|
|
|
if (has_rids) {
|
|
sender_options.rids = send_rids;
|
|
}
|
|
|
|
sender_options.simulcast_layers = send_layers;
|
|
// When RIDs are configured, we must set num_sim_layers to 0 to.
|
|
// Otherwise, num_sim_layers must be 1 because either there is no
|
|
// simulcast, or simulcast is acheived by munging the SDP.
|
|
sender_options.num_sim_layers = has_rids ? 0 : 1;
|
|
media_description_options.sender_options.push_back(sender_options);
|
|
|
|
return media_description_options;
|
|
}
|
|
|
|
// Returns the ContentInfo at mline index |i|, or null if none exists.
|
|
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);
|
|
}
|
|
|
|
// From |rtc_options|, fill parts of |session_options| shared by all generated
|
|
// m= sectionss (in other words, nothing that involves a map/array).
|
|
void ExtractSharedMediaSessionOptions(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
session_options->vad_enabled = rtc_options.voice_activity_detection;
|
|
session_options->bundle_enabled = rtc_options.use_rtp_mux;
|
|
session_options->raw_packetization_for_video =
|
|
rtc_options.raw_packetization_for_video;
|
|
}
|
|
|
|
// 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;
|
|
}
|
|
|
|
// 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;
|
|
}
|
|
|
|
rtc::scoped_refptr<webrtc::DtlsTransport> LookupDtlsTransportByMid(
|
|
rtc::Thread* network_thread,
|
|
JsepTransportController* controller,
|
|
const std::string& mid) {
|
|
// TODO(tommi): Can we post this (and associated operations where this
|
|
// function is called) to the network thread and avoid this Invoke?
|
|
// We might be able to simplify a few things if we set the transport on
|
|
// the network thread and then update the implementation to check that
|
|
// the set_ and relevant get methods are always called on the network
|
|
// thread (we'll need to update proxy maps).
|
|
return network_thread->Invoke<rtc::scoped_refptr<webrtc::DtlsTransport>>(
|
|
RTC_FROM_HERE,
|
|
[controller, &mid] { return controller->LookupDtlsTransportByMid(mid); });
|
|
}
|
|
|
|
} // namespace
|
|
|
|
// Used by parameterless SetLocalDescription() to create an offer or answer.
|
|
// Upon completion of creating the session description, SetLocalDescription() is
|
|
// invoked with the result.
|
|
class SdpOfferAnswerHandler::ImplicitCreateSessionDescriptionObserver
|
|
: public CreateSessionDescriptionObserver {
|
|
public:
|
|
ImplicitCreateSessionDescriptionObserver(
|
|
rtc::WeakPtr<SdpOfferAnswerHandler> sdp_handler,
|
|
rtc::scoped_refptr<SetLocalDescriptionObserverInterface>
|
|
set_local_description_observer)
|
|
: sdp_handler_(std::move(sdp_handler)),
|
|
set_local_description_observer_(
|
|
std::move(set_local_description_observer)) {}
|
|
~ImplicitCreateSessionDescriptionObserver() override {
|
|
RTC_DCHECK(was_called_);
|
|
}
|
|
|
|
void SetOperationCompleteCallback(
|
|
std::function<void()> operation_complete_callback) {
|
|
operation_complete_callback_ = std::move(operation_complete_callback);
|
|
}
|
|
|
|
bool was_called() const { return was_called_; }
|
|
|
|
void OnSuccess(SessionDescriptionInterface* desc_ptr) override {
|
|
RTC_DCHECK(!was_called_);
|
|
std::unique_ptr<SessionDescriptionInterface> desc(desc_ptr);
|
|
was_called_ = true;
|
|
|
|
// Abort early if |pc_| is no longer valid.
|
|
if (!sdp_handler_) {
|
|
operation_complete_callback_();
|
|
return;
|
|
}
|
|
// DoSetLocalDescription() is a synchronous operation that invokes
|
|
// |set_local_description_observer_| with the result.
|
|
sdp_handler_->DoSetLocalDescription(
|
|
std::move(desc), std::move(set_local_description_observer_));
|
|
operation_complete_callback_();
|
|
}
|
|
|
|
void OnFailure(RTCError error) override {
|
|
RTC_DCHECK(!was_called_);
|
|
was_called_ = true;
|
|
set_local_description_observer_->OnSetLocalDescriptionComplete(RTCError(
|
|
error.type(), std::string("SetLocalDescription failed to create "
|
|
"session description - ") +
|
|
error.message()));
|
|
operation_complete_callback_();
|
|
}
|
|
|
|
private:
|
|
bool was_called_ = false;
|
|
rtc::WeakPtr<SdpOfferAnswerHandler> sdp_handler_;
|
|
rtc::scoped_refptr<SetLocalDescriptionObserverInterface>
|
|
set_local_description_observer_;
|
|
std::function<void()> operation_complete_callback_;
|
|
};
|
|
|
|
// Wraps a CreateSessionDescriptionObserver and an OperationsChain operation
|
|
// complete callback. When the observer is invoked, the wrapped observer is
|
|
// invoked followed by invoking the completion callback.
|
|
class CreateSessionDescriptionObserverOperationWrapper
|
|
: public CreateSessionDescriptionObserver {
|
|
public:
|
|
CreateSessionDescriptionObserverOperationWrapper(
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserver> observer,
|
|
std::function<void()> operation_complete_callback)
|
|
: observer_(std::move(observer)),
|
|
operation_complete_callback_(std::move(operation_complete_callback)) {
|
|
RTC_DCHECK(observer_);
|
|
}
|
|
~CreateSessionDescriptionObserverOperationWrapper() override {
|
|
#if RTC_DCHECK_IS_ON
|
|
RTC_DCHECK(was_called_);
|
|
#endif
|
|
}
|
|
|
|
void OnSuccess(SessionDescriptionInterface* desc) override {
|
|
#if RTC_DCHECK_IS_ON
|
|
RTC_DCHECK(!was_called_);
|
|
was_called_ = true;
|
|
#endif // RTC_DCHECK_IS_ON
|
|
// Completing the operation before invoking the observer allows the observer
|
|
// to execute SetLocalDescription() without delay.
|
|
operation_complete_callback_();
|
|
observer_->OnSuccess(desc);
|
|
}
|
|
|
|
void OnFailure(RTCError error) override {
|
|
#if RTC_DCHECK_IS_ON
|
|
RTC_DCHECK(!was_called_);
|
|
was_called_ = true;
|
|
#endif // RTC_DCHECK_IS_ON
|
|
operation_complete_callback_();
|
|
observer_->OnFailure(std::move(error));
|
|
}
|
|
|
|
private:
|
|
#if RTC_DCHECK_IS_ON
|
|
bool was_called_ = false;
|
|
#endif // RTC_DCHECK_IS_ON
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserver> observer_;
|
|
std::function<void()> operation_complete_callback_;
|
|
};
|
|
|
|
// Wrapper for SetSessionDescriptionObserver that invokes the success or failure
|
|
// callback in a posted message handled by the peer connection. This introduces
|
|
// a delay that prevents recursive API calls by the observer, but this also
|
|
// means that the PeerConnection can be modified before the observer sees the
|
|
// result of the operation. This is ill-advised for synchronizing states.
|
|
//
|
|
// Implements both the SetLocalDescriptionObserverInterface and the
|
|
// SetRemoteDescriptionObserverInterface.
|
|
class SdpOfferAnswerHandler::SetSessionDescriptionObserverAdapter
|
|
: public SetLocalDescriptionObserverInterface,
|
|
public SetRemoteDescriptionObserverInterface {
|
|
public:
|
|
SetSessionDescriptionObserverAdapter(
|
|
rtc::WeakPtr<SdpOfferAnswerHandler> handler,
|
|
rtc::scoped_refptr<SetSessionDescriptionObserver> inner_observer)
|
|
: handler_(std::move(handler)),
|
|
inner_observer_(std::move(inner_observer)) {}
|
|
|
|
// SetLocalDescriptionObserverInterface implementation.
|
|
void OnSetLocalDescriptionComplete(RTCError error) override {
|
|
OnSetDescriptionComplete(std::move(error));
|
|
}
|
|
// SetRemoteDescriptionObserverInterface implementation.
|
|
void OnSetRemoteDescriptionComplete(RTCError error) override {
|
|
OnSetDescriptionComplete(std::move(error));
|
|
}
|
|
|
|
private:
|
|
void OnSetDescriptionComplete(RTCError error) {
|
|
if (!handler_)
|
|
return;
|
|
if (error.ok()) {
|
|
handler_->pc_->message_handler()->PostSetSessionDescriptionSuccess(
|
|
inner_observer_);
|
|
} else {
|
|
handler_->pc_->message_handler()->PostSetSessionDescriptionFailure(
|
|
inner_observer_, std::move(error));
|
|
}
|
|
}
|
|
|
|
rtc::WeakPtr<SdpOfferAnswerHandler> handler_;
|
|
rtc::scoped_refptr<SetSessionDescriptionObserver> inner_observer_;
|
|
};
|
|
|
|
class SdpOfferAnswerHandler::LocalIceCredentialsToReplace {
|
|
public:
|
|
// Sets the ICE credentials that need restarting to the ICE credentials of
|
|
// the current and pending descriptions.
|
|
void SetIceCredentialsFromLocalDescriptions(
|
|
const SessionDescriptionInterface* current_local_description,
|
|
const SessionDescriptionInterface* pending_local_description) {
|
|
ice_credentials_.clear();
|
|
if (current_local_description) {
|
|
AppendIceCredentialsFromSessionDescription(*current_local_description);
|
|
}
|
|
if (pending_local_description) {
|
|
AppendIceCredentialsFromSessionDescription(*pending_local_description);
|
|
}
|
|
}
|
|
|
|
void ClearIceCredentials() { ice_credentials_.clear(); }
|
|
|
|
// Returns true if we have ICE credentials that need restarting.
|
|
bool HasIceCredentials() const { return !ice_credentials_.empty(); }
|
|
|
|
// Returns true if |local_description| shares no ICE credentials with the
|
|
// ICE credentials that need restarting.
|
|
bool SatisfiesIceRestart(
|
|
const SessionDescriptionInterface& local_description) const {
|
|
for (const auto& transport_info :
|
|
local_description.description()->transport_infos()) {
|
|
if (ice_credentials_.find(std::make_pair(
|
|
transport_info.description.ice_ufrag,
|
|
transport_info.description.ice_pwd)) != ice_credentials_.end()) {
|
|
return false;
|
|
}
|
|
}
|
|
return true;
|
|
}
|
|
|
|
private:
|
|
void AppendIceCredentialsFromSessionDescription(
|
|
const SessionDescriptionInterface& desc) {
|
|
for (const auto& transport_info : desc.description()->transport_infos()) {
|
|
ice_credentials_.insert(
|
|
std::make_pair(transport_info.description.ice_ufrag,
|
|
transport_info.description.ice_pwd));
|
|
}
|
|
}
|
|
|
|
std::set<std::pair<std::string, std::string>> ice_credentials_;
|
|
};
|
|
|
|
SdpOfferAnswerHandler::SdpOfferAnswerHandler(PeerConnection* pc)
|
|
: pc_(pc),
|
|
local_streams_(StreamCollection::Create()),
|
|
remote_streams_(StreamCollection::Create()),
|
|
operations_chain_(rtc::OperationsChain::Create()),
|
|
rtcp_cname_(GenerateRtcpCname()),
|
|
local_ice_credentials_to_replace_(new LocalIceCredentialsToReplace()),
|
|
weak_ptr_factory_(this) {
|
|
operations_chain_->SetOnChainEmptyCallback(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr()]() {
|
|
if (!this_weak_ptr)
|
|
return;
|
|
this_weak_ptr->OnOperationsChainEmpty();
|
|
});
|
|
}
|
|
|
|
SdpOfferAnswerHandler::~SdpOfferAnswerHandler() {}
|
|
|
|
// Static
|
|
std::unique_ptr<SdpOfferAnswerHandler> SdpOfferAnswerHandler::Create(
|
|
PeerConnection* pc,
|
|
const PeerConnectionInterface::RTCConfiguration& configuration,
|
|
PeerConnectionDependencies& dependencies) {
|
|
auto handler = absl::WrapUnique(new SdpOfferAnswerHandler(pc));
|
|
handler->Initialize(configuration, dependencies);
|
|
return handler;
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::Initialize(
|
|
const PeerConnectionInterface::RTCConfiguration& configuration,
|
|
PeerConnectionDependencies& dependencies) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
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;
|
|
|
|
// 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];
|
|
}
|
|
|
|
webrtc_session_desc_factory_ =
|
|
std::make_unique<WebRtcSessionDescriptionFactory>(
|
|
signaling_thread(), channel_manager(), this, pc_->session_id(),
|
|
pc_->dtls_enabled(), std::move(dependencies.cert_generator),
|
|
certificate, &ssrc_generator_,
|
|
[this](const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
|
|
transport_controller()->SetLocalCertificate(certificate);
|
|
});
|
|
|
|
if (pc_->options()->disable_encryption) {
|
|
webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
|
|
}
|
|
|
|
webrtc_session_desc_factory_->set_enable_encrypted_rtp_header_extensions(
|
|
pc_->GetCryptoOptions().srtp.enable_encrypted_rtp_header_extensions);
|
|
webrtc_session_desc_factory_->set_is_unified_plan(IsUnifiedPlan());
|
|
|
|
if (dependencies.video_bitrate_allocator_factory) {
|
|
video_bitrate_allocator_factory_ =
|
|
std::move(dependencies.video_bitrate_allocator_factory);
|
|
} else {
|
|
video_bitrate_allocator_factory_ =
|
|
CreateBuiltinVideoBitrateAllocatorFactory();
|
|
}
|
|
}
|
|
|
|
// ==================================================================
|
|
// Access to pc_ variables
|
|
cricket::ChannelManager* SdpOfferAnswerHandler::channel_manager() const {
|
|
return pc_->channel_manager();
|
|
}
|
|
TransceiverList* SdpOfferAnswerHandler::transceivers() {
|
|
if (!pc_->rtp_manager()) {
|
|
return nullptr;
|
|
}
|
|
return pc_->rtp_manager()->transceivers();
|
|
}
|
|
const TransceiverList* SdpOfferAnswerHandler::transceivers() const {
|
|
if (!pc_->rtp_manager()) {
|
|
return nullptr;
|
|
}
|
|
return pc_->rtp_manager()->transceivers();
|
|
}
|
|
JsepTransportController* SdpOfferAnswerHandler::transport_controller() {
|
|
return pc_->transport_controller();
|
|
}
|
|
const JsepTransportController* SdpOfferAnswerHandler::transport_controller()
|
|
const {
|
|
return pc_->transport_controller();
|
|
}
|
|
DataChannelController* SdpOfferAnswerHandler::data_channel_controller() {
|
|
return pc_->data_channel_controller();
|
|
}
|
|
const DataChannelController* SdpOfferAnswerHandler::data_channel_controller()
|
|
const {
|
|
return pc_->data_channel_controller();
|
|
}
|
|
cricket::PortAllocator* SdpOfferAnswerHandler::port_allocator() {
|
|
return pc_->port_allocator();
|
|
}
|
|
const cricket::PortAllocator* SdpOfferAnswerHandler::port_allocator() const {
|
|
return pc_->port_allocator();
|
|
}
|
|
RtpTransmissionManager* SdpOfferAnswerHandler::rtp_manager() {
|
|
return pc_->rtp_manager();
|
|
}
|
|
const RtpTransmissionManager* SdpOfferAnswerHandler::rtp_manager() const {
|
|
return pc_->rtp_manager();
|
|
}
|
|
|
|
// ===================================================================
|
|
|
|
void SdpOfferAnswerHandler::PrepareForShutdown() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
weak_ptr_factory_.InvalidateWeakPtrs();
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::Close() {
|
|
ChangeSignalingState(PeerConnectionInterface::kClosed);
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::RestartIce() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
local_ice_credentials_to_replace_->SetIceCredentialsFromLocalDescriptions(
|
|
current_local_description(), pending_local_description());
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
rtc::Thread* SdpOfferAnswerHandler::signaling_thread() const {
|
|
return pc_->signaling_thread();
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::CreateOffer(
|
|
CreateSessionDescriptionObserver* observer,
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
|
|
observer_refptr =
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserver>(observer),
|
|
options](std::function<void()> operations_chain_callback) {
|
|
// Abort early if |this_weak_ptr| is no longer valid.
|
|
if (!this_weak_ptr) {
|
|
observer_refptr->OnFailure(
|
|
RTCError(RTCErrorType::INTERNAL_ERROR,
|
|
"CreateOffer failed because the session was shut down"));
|
|
operations_chain_callback();
|
|
return;
|
|
}
|
|
// The operation completes asynchronously when the wrapper is invoked.
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserverOperationWrapper>
|
|
observer_wrapper(new rtc::RefCountedObject<
|
|
CreateSessionDescriptionObserverOperationWrapper>(
|
|
std::move(observer_refptr),
|
|
std::move(operations_chain_callback)));
|
|
this_weak_ptr->DoCreateOffer(options, observer_wrapper);
|
|
});
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::SetLocalDescription(
|
|
SetSessionDescriptionObserver* observer,
|
|
SessionDescriptionInterface* desc_ptr) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
|
|
observer_refptr =
|
|
rtc::scoped_refptr<SetSessionDescriptionObserver>(observer),
|
|
desc = std::unique_ptr<SessionDescriptionInterface>(desc_ptr)](
|
|
std::function<void()> operations_chain_callback) mutable {
|
|
// Abort early if |this_weak_ptr| is no longer valid.
|
|
if (!this_weak_ptr) {
|
|
// For consistency with SetSessionDescriptionObserverAdapter whose
|
|
// posted messages doesn't get processed when the PC is destroyed, we
|
|
// do not inform |observer_refptr| that the operation failed.
|
|
operations_chain_callback();
|
|
return;
|
|
}
|
|
// SetSessionDescriptionObserverAdapter takes care of making sure the
|
|
// |observer_refptr| is invoked in a posted message.
|
|
this_weak_ptr->DoSetLocalDescription(
|
|
std::move(desc),
|
|
rtc::scoped_refptr<SetLocalDescriptionObserverInterface>(
|
|
new rtc::RefCountedObject<SetSessionDescriptionObserverAdapter>(
|
|
this_weak_ptr, observer_refptr)));
|
|
// For backwards-compatability reasons, we declare the operation as
|
|
// completed here (rather than in a post), so that the operation chain
|
|
// is not blocked by this operation when the observer is invoked. This
|
|
// allows the observer to trigger subsequent offer/answer operations
|
|
// synchronously if the operation chain is now empty.
|
|
operations_chain_callback();
|
|
});
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::SetLocalDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc,
|
|
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(), observer,
|
|
desc = std::move(desc)](
|
|
std::function<void()> operations_chain_callback) mutable {
|
|
// Abort early if |this_weak_ptr| is no longer valid.
|
|
if (!this_weak_ptr) {
|
|
observer->OnSetLocalDescriptionComplete(RTCError(
|
|
RTCErrorType::INTERNAL_ERROR,
|
|
"SetLocalDescription failed because the session was shut down"));
|
|
operations_chain_callback();
|
|
return;
|
|
}
|
|
this_weak_ptr->DoSetLocalDescription(std::move(desc), observer);
|
|
// DoSetLocalDescription() is implemented as a synchronous operation.
|
|
// The |observer| will already have been informed that it completed, and
|
|
// we can mark this operation as complete without any loose ends.
|
|
operations_chain_callback();
|
|
});
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::SetLocalDescription(
|
|
SetSessionDescriptionObserver* observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
SetLocalDescription(
|
|
new rtc::RefCountedObject<SetSessionDescriptionObserverAdapter>(
|
|
weak_ptr_factory_.GetWeakPtr(), observer));
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::SetLocalDescription(
|
|
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// The |create_sdp_observer| handles performing DoSetLocalDescription() with
|
|
// the resulting description as well as completing the operation.
|
|
rtc::scoped_refptr<ImplicitCreateSessionDescriptionObserver>
|
|
create_sdp_observer(
|
|
new rtc::RefCountedObject<ImplicitCreateSessionDescriptionObserver>(
|
|
weak_ptr_factory_.GetWeakPtr(), observer));
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
|
|
create_sdp_observer](std::function<void()> operations_chain_callback) {
|
|
// The |create_sdp_observer| is responsible for completing the
|
|
// operation.
|
|
create_sdp_observer->SetOperationCompleteCallback(
|
|
std::move(operations_chain_callback));
|
|
// Abort early if |this_weak_ptr| is no longer valid. This triggers the
|
|
// same code path as if DoCreateOffer() or DoCreateAnswer() failed.
|
|
if (!this_weak_ptr) {
|
|
create_sdp_observer->OnFailure(RTCError(
|
|
RTCErrorType::INTERNAL_ERROR,
|
|
"SetLocalDescription failed because the session was shut down"));
|
|
return;
|
|
}
|
|
switch (this_weak_ptr->signaling_state()) {
|
|
case PeerConnectionInterface::kStable:
|
|
case PeerConnectionInterface::kHaveLocalOffer:
|
|
case PeerConnectionInterface::kHaveRemotePrAnswer:
|
|
// TODO(hbos): If [LastCreatedOffer] exists and still represents the
|
|
// current state of the system, use that instead of creating another
|
|
// offer.
|
|
this_weak_ptr->DoCreateOffer(
|
|
PeerConnectionInterface::RTCOfferAnswerOptions(),
|
|
create_sdp_observer);
|
|
break;
|
|
case PeerConnectionInterface::kHaveLocalPrAnswer:
|
|
case PeerConnectionInterface::kHaveRemoteOffer:
|
|
// TODO(hbos): If [LastCreatedAnswer] exists and still represents
|
|
// the current state of the system, use that instead of creating
|
|
// another answer.
|
|
this_weak_ptr->DoCreateAnswer(
|
|
PeerConnectionInterface::RTCOfferAnswerOptions(),
|
|
create_sdp_observer);
|
|
break;
|
|
case PeerConnectionInterface::kClosed:
|
|
create_sdp_observer->OnFailure(RTCError(
|
|
RTCErrorType::INVALID_STATE,
|
|
"SetLocalDescription called when PeerConnection is closed."));
|
|
break;
|
|
}
|
|
});
|
|
}
|
|
|
|
RTCError SdpOfferAnswerHandler::ApplyLocalDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc,
|
|
const std::map<std::string, const cricket::ContentGroup*>&
|
|
bundle_groups_by_mid) {
|
|
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.
|
|
pc_->stats()->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
|
|
|
|
// Take a reference to the old local description since it's used below to
|
|
// compare against the new local description. When setting the new local
|
|
// description, grab ownership of the replaced session description in case it
|
|
// is the same as |old_local_description|, to keep it alive for the duration
|
|
// of the method.
|
|
const SessionDescriptionInterface* old_local_description =
|
|
local_description();
|
|
std::unique_ptr<SessionDescriptionInterface> replaced_local_description;
|
|
SdpType type = desc->GetType();
|
|
if (type == SdpType::kAnswer) {
|
|
replaced_local_description = pending_local_description_
|
|
? std::move(pending_local_description_)
|
|
: std::move(current_local_description_);
|
|
current_local_description_ = std::move(desc);
|
|
pending_local_description_ = nullptr;
|
|
current_remote_description_ = std::move(pending_remote_description_);
|
|
} else {
|
|
replaced_local_description = std::move(pending_local_description_);
|
|
pending_local_description_ = std::move(desc);
|
|
}
|
|
// The session description to apply now must be accessed by
|
|
// |local_description()|.
|
|
RTC_DCHECK(local_description());
|
|
|
|
// Report statistics about any use of simulcast.
|
|
ReportSimulcastApiVersion(kSimulcastVersionApplyLocalDescription,
|
|
*local_description()->description());
|
|
|
|
if (!is_caller_) {
|
|
if (remote_description()) {
|
|
// Remote description was applied first, so this PC is the callee.
|
|
is_caller_ = false;
|
|
} else {
|
|
// Local description is applied first, so this PC is the caller.
|
|
is_caller_ = true;
|
|
}
|
|
}
|
|
|
|
RTCError error = PushdownTransportDescription(cricket::CS_LOCAL, type);
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
|
|
if (IsUnifiedPlan()) {
|
|
RTCError error = UpdateTransceiversAndDataChannels(
|
|
cricket::CS_LOCAL, *local_description(), old_local_description,
|
|
remote_description(), bundle_groups_by_mid);
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
|
|
for (const auto& transceiver_ext : transceivers()->List()) {
|
|
auto transceiver = transceiver_ext->internal();
|
|
if (transceiver->stopped()) {
|
|
continue;
|
|
}
|
|
|
|
// 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 = LookupDtlsTransportByMid(
|
|
pc_->network_thread(), transport_controller(), *transceiver->mid());
|
|
transceiver->sender_internal()->set_transport(dtls_transport);
|
|
transceiver->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->fired_direction() &&
|
|
RtpTransceiverDirectionHasRecv(*transceiver->fired_direction()))) {
|
|
ProcessRemovalOfRemoteTrack(transceiver_ext, &remove_list,
|
|
&removed_streams);
|
|
}
|
|
// 2.2.7.1.6.2: Set transceiver's [[CurrentDirection]] and
|
|
// [[FiredDirection]] slots to direction.
|
|
transceiver->set_current_direction(media_desc->direction());
|
|
transceiver->set_fired_direction(media_desc->direction());
|
|
}
|
|
}
|
|
auto observer = pc_->Observer();
|
|
for (const auto& transceiver : remove_list) {
|
|
observer->OnRemoveTrack(transceiver->receiver());
|
|
}
|
|
for (const auto& stream : removed_streams) {
|
|
observer->OnRemoveStream(stream);
|
|
}
|
|
} else {
|
|
// Media channels will be created only when offer is set. These may use new
|
|
// transports just created by PushdownTransportDescription.
|
|
if (type == SdpType::kOffer) {
|
|
// TODO(bugs.webrtc.org/4676) - Handle CreateChannel failure, as new local
|
|
// description is applied. Restore back to old description.
|
|
RTCError error = CreateChannels(*local_description()->description());
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
}
|
|
// Remove unused channels if MediaContentDescription is rejected.
|
|
RemoveUnusedChannels(local_description()->description());
|
|
}
|
|
|
|
error = UpdateSessionState(type, cricket::CS_LOCAL,
|
|
local_description()->description(),
|
|
bundle_groups_by_mid);
|
|
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 (pc_->GetSctpSslRole(&role)) {
|
|
data_channel_controller()->AllocateSctpSids(role);
|
|
}
|
|
|
|
if (IsUnifiedPlan()) {
|
|
// We must use List and not ListInternal here because
|
|
// transceivers()->StableState() is indexed by the non-internal refptr.
|
|
for (const auto& transceiver_ext : transceivers()->List()) {
|
|
auto transceiver = transceiver_ext->internal();
|
|
if (transceiver->stopped()) {
|
|
continue;
|
|
}
|
|
const ContentInfo* content =
|
|
FindMediaSectionForTransceiver(transceiver, local_description());
|
|
if (!content) {
|
|
continue;
|
|
}
|
|
cricket::ChannelInterface* channel = transceiver->channel();
|
|
if (content->rejected || !channel || channel->local_streams().empty()) {
|
|
// 0 is a special value meaning "this sender has no associated send
|
|
// stream". Need to call this so the sender won't attempt to configure
|
|
// a no longer existing stream and run into DCHECKs in the lower
|
|
// layers.
|
|
transceiver->sender_internal()->SetSsrc(0);
|
|
} else {
|
|
// Get the StreamParams from the channel which could generate SSRCs.
|
|
const std::vector<StreamParams>& streams = channel->local_streams();
|
|
transceiver->sender_internal()->set_stream_ids(streams[0].stream_ids());
|
|
auto encodings = transceiver->sender_internal()->init_send_encodings();
|
|
transceiver->sender_internal()->SetSsrc(streams[0].first_ssrc());
|
|
if (!encodings.empty()) {
|
|
transceivers()
|
|
->StableState(transceiver_ext)
|
|
->SetInitSendEncodings(encodings);
|
|
}
|
|
}
|
|
}
|
|
} 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());
|
|
}
|
|
}
|
|
}
|
|
|
|
// This function does nothing with data content.
|
|
|
|
if (type == SdpType::kAnswer &&
|
|
local_ice_credentials_to_replace_->SatisfiesIceRestart(
|
|
*current_local_description_)) {
|
|
local_ice_credentials_to_replace_->ClearIceCredentials();
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::SetRemoteDescription(
|
|
SetSessionDescriptionObserver* observer,
|
|
SessionDescriptionInterface* desc_ptr) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
|
|
observer_refptr =
|
|
rtc::scoped_refptr<SetSessionDescriptionObserver>(observer),
|
|
desc = std::unique_ptr<SessionDescriptionInterface>(desc_ptr)](
|
|
std::function<void()> operations_chain_callback) mutable {
|
|
// Abort early if |this_weak_ptr| is no longer valid.
|
|
if (!this_weak_ptr) {
|
|
// For consistency with SetSessionDescriptionObserverAdapter whose
|
|
// posted messages doesn't get processed when the PC is destroyed, we
|
|
// do not inform |observer_refptr| that the operation failed.
|
|
operations_chain_callback();
|
|
return;
|
|
}
|
|
// SetSessionDescriptionObserverAdapter takes care of making sure the
|
|
// |observer_refptr| is invoked in a posted message.
|
|
this_weak_ptr->DoSetRemoteDescription(
|
|
std::move(desc),
|
|
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>(
|
|
new rtc::RefCountedObject<SetSessionDescriptionObserverAdapter>(
|
|
this_weak_ptr, observer_refptr)));
|
|
// For backwards-compatability reasons, we declare the operation as
|
|
// completed here (rather than in a post), so that the operation chain
|
|
// is not blocked by this operation when the observer is invoked. This
|
|
// allows the observer to trigger subsequent offer/answer operations
|
|
// synchronously if the operation chain is now empty.
|
|
operations_chain_callback();
|
|
});
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::SetRemoteDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc,
|
|
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(), observer,
|
|
desc = std::move(desc)](
|
|
std::function<void()> operations_chain_callback) mutable {
|
|
// Abort early if |this_weak_ptr| is no longer valid.
|
|
if (!this_weak_ptr) {
|
|
observer->OnSetRemoteDescriptionComplete(RTCError(
|
|
RTCErrorType::INTERNAL_ERROR,
|
|
"SetRemoteDescription failed because the session was shut down"));
|
|
operations_chain_callback();
|
|
return;
|
|
}
|
|
this_weak_ptr->DoSetRemoteDescription(std::move(desc),
|
|
std::move(observer));
|
|
// DoSetRemoteDescription() is implemented as a synchronous operation.
|
|
// The |observer| will already have been informed that it completed, and
|
|
// we can mark this operation as complete without any loose ends.
|
|
operations_chain_callback();
|
|
});
|
|
}
|
|
|
|
RTCError SdpOfferAnswerHandler::ApplyRemoteDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc,
|
|
const std::map<std::string, const cricket::ContentGroup*>&
|
|
bundle_groups_by_mid) {
|
|
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.
|
|
pc_->stats()->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
|
|
|
|
// Take a reference to the old remote description since it's used below to
|
|
// compare against the new remote description. When setting the new remote
|
|
// description, grab ownership of the replaced session description in case it
|
|
// is the same as |old_remote_description|, to keep it alive for the duration
|
|
// of the method.
|
|
const SessionDescriptionInterface* old_remote_description =
|
|
remote_description();
|
|
std::unique_ptr<SessionDescriptionInterface> replaced_remote_description;
|
|
SdpType type = desc->GetType();
|
|
if (type == SdpType::kAnswer) {
|
|
replaced_remote_description = pending_remote_description_
|
|
? std::move(pending_remote_description_)
|
|
: std::move(current_remote_description_);
|
|
current_remote_description_ = std::move(desc);
|
|
pending_remote_description_ = nullptr;
|
|
current_local_description_ = std::move(pending_local_description_);
|
|
} else {
|
|
replaced_remote_description = std::move(pending_remote_description_);
|
|
pending_remote_description_ = std::move(desc);
|
|
}
|
|
// The session description to apply now must be accessed by
|
|
// |remote_description()|.
|
|
RTC_DCHECK(remote_description());
|
|
|
|
// Report statistics about any use of simulcast.
|
|
ReportSimulcastApiVersion(kSimulcastVersionApplyRemoteDescription,
|
|
*remote_description()->description());
|
|
|
|
RTCError error = PushdownTransportDescription(cricket::CS_REMOTE, type);
|
|
if (!error.ok()) {
|
|
return error;
|
|
}
|
|
// Transport and Media channels will be created only when offer is set.
|
|
if (IsUnifiedPlan()) {
|
|
RTCError error = UpdateTransceiversAndDataChannels(
|
|
cricket::CS_REMOTE, *remote_description(), local_description(),
|
|
old_remote_description, bundle_groups_by_mid);
|
|
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(),
|
|
bundle_groups_by_mid);
|
|
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 &&
|
|
pc_->ice_connection_state() ==
|
|
PeerConnectionInterface::kIceConnectionNew) {
|
|
pc_->SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
|
|
}
|
|
|
|
// If setting the description decided our SSL role, allocate any necessary
|
|
// SCTP sids.
|
|
rtc::SSLRole role;
|
|
if (pc_->GetSctpSslRole(&role)) {
|
|
data_channel_controller()->AllocateSctpSids(role);
|
|
}
|
|
|
|
if (IsUnifiedPlan()) {
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
|
|
now_receiving_transceivers;
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> added_streams;
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
|
|
for (const auto& transceiver_ext : transceivers()->List()) {
|
|
const auto transceiver = transceiver_ext->internal();
|
|
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();
|
|
}
|
|
transceivers()
|
|
->StableState(transceiver_ext)
|
|
->SetRemoteStreamIdsIfUnset(transceiver->receiver()->stream_ids());
|
|
|
|
RTC_LOG(LS_INFO) << "Processing the MSIDs for MID=" << content->name
|
|
<< " (" << GetStreamIdsString(stream_ids) << ").";
|
|
SetAssociatedRemoteStreams(transceiver->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_ext, &remove_list,
|
|
&removed_streams);
|
|
}
|
|
// 2.2.8.1.10: Set transceiver's [[FiredDirection]] slot to direction.
|
|
transceiver->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->set_current_direction(local_direction);
|
|
// 2.2.8.1.11.[3-6]: Set the transport internal slots.
|
|
if (transceiver->mid()) {
|
|
auto dtls_transport = LookupDtlsTransportByMid(pc_->network_thread(),
|
|
transport_controller(),
|
|
*transceiver->mid());
|
|
transceiver->sender_internal()->set_transport(dtls_transport);
|
|
transceiver->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->StopTransceiverProcedure();
|
|
}
|
|
if (!content->rejected &&
|
|
RtpTransceiverDirectionHasRecv(local_direction)) {
|
|
if (!media_desc->streams().empty() &&
|
|
media_desc->streams()[0].has_ssrcs()) {
|
|
uint32_t ssrc = media_desc->streams()[0].first_ssrc();
|
|
transceiver->receiver_internal()->SetupMediaChannel(ssrc);
|
|
} else {
|
|
transceiver->receiver_internal()->SetupUnsignaledMediaChannel();
|
|
}
|
|
}
|
|
}
|
|
// Once all processing has finished, fire off callbacks.
|
|
auto observer = pc_->Observer();
|
|
for (const auto& transceiver : now_receiving_transceivers) {
|
|
pc_->stats()->AddTrack(transceiver->receiver()->track());
|
|
observer->OnTrack(transceiver);
|
|
observer->OnAddTrack(transceiver->receiver(),
|
|
transceiver->receiver()->streams());
|
|
}
|
|
for (const auto& stream : added_streams) {
|
|
observer->OnAddStream(stream);
|
|
}
|
|
for (const auto& transceiver : remove_list) {
|
|
observer->OnRemoveTrack(transceiver->receiver());
|
|
}
|
|
for (const auto& stream : removed_streams) {
|
|
observer->OnRemoveStream(stream);
|
|
}
|
|
}
|
|
|
|
const cricket::ContentInfo* audio_content =
|
|
GetFirstAudioContent(remote_description()->description());
|
|
const cricket::ContentInfo* video_content =
|
|
GetFirstVideoContent(remote_description()->description());
|
|
const cricket::AudioContentDescription* audio_desc =
|
|
GetFirstAudioContentDescription(remote_description()->description());
|
|
const cricket::VideoContentDescription* video_desc =
|
|
GetFirstVideoContentDescription(remote_description()->description());
|
|
|
|
// 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);
|
|
}
|
|
}
|
|
|
|
// Iterate new_streams and notify the observer about new MediaStreams.
|
|
auto observer = pc_->Observer();
|
|
for (size_t i = 0; i < new_streams->count(); ++i) {
|
|
MediaStreamInterface* new_stream = new_streams->at(i);
|
|
pc_->stats()->AddStream(new_stream);
|
|
observer->OnAddStream(
|
|
rtc::scoped_refptr<MediaStreamInterface>(new_stream));
|
|
}
|
|
|
|
UpdateEndedRemoteMediaStreams();
|
|
}
|
|
|
|
if (type == SdpType::kAnswer &&
|
|
local_ice_credentials_to_replace_->SatisfiesIceRestart(
|
|
*current_local_description_)) {
|
|
local_ice_credentials_to_replace_->ClearIceCredentials();
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::DoSetLocalDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc,
|
|
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::DoSetLocalDescription");
|
|
|
|
if (!observer) {
|
|
RTC_LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
if (!desc) {
|
|
observer->OnSetLocalDescriptionComplete(
|
|
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;
|
|
observer->OnSetLocalDescriptionComplete(
|
|
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
|
return;
|
|
}
|
|
|
|
// For SLD we support only explicit rollback.
|
|
if (desc->GetType() == SdpType::kRollback) {
|
|
if (IsUnifiedPlan()) {
|
|
observer->OnSetLocalDescriptionComplete(Rollback(desc->GetType()));
|
|
} else {
|
|
observer->OnSetLocalDescriptionComplete(
|
|
RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
|
|
"Rollback not supported in Plan B"));
|
|
}
|
|
return;
|
|
}
|
|
|
|
std::map<std::string, const cricket::ContentGroup*> bundle_groups_by_mid =
|
|
GetBundleGroupsByMid(desc->description());
|
|
RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_LOCAL,
|
|
bundle_groups_by_mid);
|
|
if (!error.ok()) {
|
|
std::string error_message = GetSetDescriptionErrorMessage(
|
|
cricket::CS_LOCAL, desc->GetType(), error);
|
|
RTC_LOG(LS_ERROR) << error_message;
|
|
observer->OnSetLocalDescriptionComplete(
|
|
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), bundle_groups_by_mid);
|
|
// |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;
|
|
observer->OnSetLocalDescriptionComplete(
|
|
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
|
return;
|
|
}
|
|
RTC_DCHECK(local_description());
|
|
|
|
if (local_description()->GetType() == SdpType::kAnswer) {
|
|
RemoveStoppedTransceivers();
|
|
|
|
// TODO(deadbeef): We already had to hop to the network thread for
|
|
// MaybeStartGathering...
|
|
pc_->network_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, [this] { port_allocator()->DiscardCandidatePool(); });
|
|
// Make UMA notes about what was agreed to.
|
|
ReportNegotiatedSdpSemantics(*local_description());
|
|
}
|
|
|
|
observer->OnSetLocalDescriptionComplete(RTCError::OK());
|
|
pc_->NoteUsageEvent(UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED);
|
|
|
|
// Check if negotiation is needed. We must do this after informing the
|
|
// observer that SetLocalDescription() has completed to ensure negotiation is
|
|
// not needed prior to the promise resolving.
|
|
if (IsUnifiedPlan()) {
|
|
bool was_negotiation_needed = is_negotiation_needed_;
|
|
UpdateNegotiationNeeded();
|
|
if (signaling_state() == PeerConnectionInterface::kStable &&
|
|
was_negotiation_needed && is_negotiation_needed_) {
|
|
// Legacy version.
|
|
pc_->Observer()->OnRenegotiationNeeded();
|
|
// Spec-compliant version; the event may get invalidated before firing.
|
|
GenerateNegotiationNeededEvent();
|
|
}
|
|
}
|
|
|
|
// MaybeStartGathering needs to be called after informing the observer so that
|
|
// we don't signal any candidates before signaling that SetLocalDescription
|
|
// completed.
|
|
transport_controller()->MaybeStartGathering();
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::DoCreateOffer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& options,
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserver> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::DoCreateOffer");
|
|
|
|
if (!observer) {
|
|
RTC_LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
if (pc_->IsClosed()) {
|
|
std::string error = "CreateOffer called when PeerConnection is closed.";
|
|
RTC_LOG(LS_ERROR) << error;
|
|
pc_->message_handler()->PostCreateSessionDescriptionFailure(
|
|
observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
|
|
return;
|
|
}
|
|
|
|
// If a session error has occurred the PeerConnection is in a possibly
|
|
// inconsistent state so fail right away.
|
|
if (session_error() != SessionError::kNone) {
|
|
std::string error_message = GetSessionErrorMsg();
|
|
RTC_LOG(LS_ERROR) << "CreateOffer: " << error_message;
|
|
pc_->message_handler()->PostCreateSessionDescriptionFailure(
|
|
observer,
|
|
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
|
return;
|
|
}
|
|
|
|
if (!ValidateOfferAnswerOptions(options)) {
|
|
std::string error = "CreateOffer called with invalid options.";
|
|
RTC_LOG(LS_ERROR) << error;
|
|
pc_->message_handler()->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()) {
|
|
pc_->message_handler()->PostCreateSessionDescriptionFailure(
|
|
observer, std::move(error));
|
|
return;
|
|
}
|
|
}
|
|
|
|
cricket::MediaSessionOptions session_options;
|
|
GetOptionsForOffer(options, &session_options);
|
|
webrtc_session_desc_factory_->CreateOffer(observer, options, session_options);
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::CreateAnswer(
|
|
CreateSessionDescriptionObserver* observer,
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
|
|
observer_refptr =
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserver>(observer),
|
|
options](std::function<void()> operations_chain_callback) {
|
|
// Abort early if |this_weak_ptr| is no longer valid.
|
|
if (!this_weak_ptr) {
|
|
observer_refptr->OnFailure(RTCError(
|
|
RTCErrorType::INTERNAL_ERROR,
|
|
"CreateAnswer failed because the session was shut down"));
|
|
operations_chain_callback();
|
|
return;
|
|
}
|
|
// The operation completes asynchronously when the wrapper is invoked.
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserverOperationWrapper>
|
|
observer_wrapper(new rtc::RefCountedObject<
|
|
CreateSessionDescriptionObserverOperationWrapper>(
|
|
std::move(observer_refptr),
|
|
std::move(operations_chain_callback)));
|
|
this_weak_ptr->DoCreateAnswer(options, observer_wrapper);
|
|
});
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::DoCreateAnswer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& options,
|
|
rtc::scoped_refptr<CreateSessionDescriptionObserver> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::DoCreateAnswer");
|
|
if (!observer) {
|
|
RTC_LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
// If a session error has occurred the PeerConnection is in a possibly
|
|
// inconsistent state so fail right away.
|
|
if (session_error() != SessionError::kNone) {
|
|
std::string error_message = GetSessionErrorMsg();
|
|
RTC_LOG(LS_ERROR) << "CreateAnswer: " << error_message;
|
|
pc_->message_handler()->PostCreateSessionDescriptionFailure(
|
|
observer,
|
|
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
|
return;
|
|
}
|
|
|
|
if (!(signaling_state_ == PeerConnectionInterface::kHaveRemoteOffer ||
|
|
signaling_state_ == PeerConnectionInterface::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;
|
|
pc_->message_handler()->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 !=
|
|
PeerConnectionInterface::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 !=
|
|
PeerConnectionInterface::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 SdpOfferAnswerHandler::DoSetRemoteDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc,
|
|
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::DoSetRemoteDescription");
|
|
|
|
if (!observer) {
|
|
RTC_LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
if (!desc) {
|
|
observer->OnSetRemoteDescriptionComplete(RTCError(
|
|
RTCErrorType::INVALID_PARAMETER, "SessionDescription is NULL."));
|
|
return;
|
|
}
|
|
|
|
// If a session error has occurred the PeerConnection is in a possibly
|
|
// inconsistent state so fail right away.
|
|
if (session_error() != SessionError::kNone) {
|
|
std::string error_message = GetSessionErrorMsg();
|
|
RTC_LOG(LS_ERROR) << "SetRemoteDescription: " << error_message;
|
|
observer->OnSetRemoteDescriptionComplete(
|
|
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
|
return;
|
|
}
|
|
if (IsUnifiedPlan()) {
|
|
if (pc_->configuration()->enable_implicit_rollback) {
|
|
if (desc->GetType() == SdpType::kOffer &&
|
|
signaling_state() == PeerConnectionInterface::kHaveLocalOffer) {
|
|
Rollback(desc->GetType());
|
|
}
|
|
}
|
|
// Explicit rollback.
|
|
if (desc->GetType() == SdpType::kRollback) {
|
|
observer->OnSetRemoteDescriptionComplete(Rollback(desc->GetType()));
|
|
return;
|
|
}
|
|
} else if (desc->GetType() == SdpType::kRollback) {
|
|
observer->OnSetRemoteDescriptionComplete(
|
|
RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
|
|
"Rollback not supported in Plan B"));
|
|
return;
|
|
}
|
|
if (desc->GetType() == SdpType::kOffer ||
|
|
desc->GetType() == SdpType::kAnswer) {
|
|
// Report to UMA the format of the received offer or answer.
|
|
pc_->ReportSdpFormatReceived(*desc);
|
|
pc_->ReportSdpBundleUsage(*desc);
|
|
}
|
|
|
|
// Handle remote descriptions missing a=mid lines for interop with legacy end
|
|
// points.
|
|
FillInMissingRemoteMids(desc->description());
|
|
|
|
std::map<std::string, const cricket::ContentGroup*> bundle_groups_by_mid =
|
|
GetBundleGroupsByMid(desc->description());
|
|
RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_REMOTE,
|
|
bundle_groups_by_mid);
|
|
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), bundle_groups_by_mid);
|
|
// |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) {
|
|
RemoveStoppedTransceivers();
|
|
// TODO(deadbeef): We already had to hop to the network thread for
|
|
// MaybeStartGathering...
|
|
pc_->network_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, [this] { port_allocator()->DiscardCandidatePool(); });
|
|
// Make UMA notes about what was agreed to.
|
|
ReportNegotiatedSdpSemantics(*remote_description());
|
|
}
|
|
|
|
observer->OnSetRemoteDescriptionComplete(RTCError::OK());
|
|
pc_->NoteUsageEvent(UsageEvent::SET_REMOTE_DESCRIPTION_SUCCEEDED);
|
|
|
|
// Check if negotiation is needed. We must do this after informing the
|
|
// observer that SetRemoteDescription() has completed to ensure negotiation is
|
|
// not needed prior to the promise resolving.
|
|
if (IsUnifiedPlan()) {
|
|
bool was_negotiation_needed = is_negotiation_needed_;
|
|
UpdateNegotiationNeeded();
|
|
if (signaling_state() == PeerConnectionInterface::kStable &&
|
|
was_negotiation_needed && is_negotiation_needed_) {
|
|
// Legacy version.
|
|
pc_->Observer()->OnRenegotiationNeeded();
|
|
// Spec-compliant version; the event may get invalidated before firing.
|
|
GenerateNegotiationNeededEvent();
|
|
}
|
|
}
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::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) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
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);
|
|
}
|
|
|
|
bool SdpOfferAnswerHandler::AddIceCandidate(
|
|
const IceCandidateInterface* ice_candidate) {
|
|
const AddIceCandidateResult result = AddIceCandidateInternal(ice_candidate);
|
|
NoteAddIceCandidateResult(result);
|
|
// If the return value is kAddIceCandidateFailNotReady, the candidate has been
|
|
// added, although not 'ready', but that's a success.
|
|
return result == kAddIceCandidateSuccess ||
|
|
result == kAddIceCandidateFailNotReady;
|
|
}
|
|
|
|
AddIceCandidateResult SdpOfferAnswerHandler::AddIceCandidateInternal(
|
|
const IceCandidateInterface* ice_candidate) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::AddIceCandidate");
|
|
if (pc_->IsClosed()) {
|
|
RTC_LOG(LS_ERROR) << "AddIceCandidate: PeerConnection is closed.";
|
|
return kAddIceCandidateFailClosed;
|
|
}
|
|
|
|
if (!remote_description()) {
|
|
RTC_LOG(LS_ERROR) << "AddIceCandidate: ICE candidates can't be added "
|
|
"without any remote session description.";
|
|
return kAddIceCandidateFailNoRemoteDescription;
|
|
}
|
|
|
|
if (!ice_candidate) {
|
|
RTC_LOG(LS_ERROR) << "AddIceCandidate: Candidate is null.";
|
|
return kAddIceCandidateFailNullCandidate;
|
|
}
|
|
|
|
bool valid = false;
|
|
bool ready = ReadyToUseRemoteCandidate(ice_candidate, nullptr, &valid);
|
|
if (!valid) {
|
|
return kAddIceCandidateFailNotValid;
|
|
}
|
|
|
|
// Add this candidate to the remote session description.
|
|
if (!mutable_remote_description()->AddCandidate(ice_candidate)) {
|
|
RTC_LOG(LS_ERROR) << "AddIceCandidate: Candidate cannot be used.";
|
|
return kAddIceCandidateFailInAddition;
|
|
}
|
|
|
|
if (!ready) {
|
|
RTC_LOG(LS_INFO) << "AddIceCandidate: Not ready to use candidate.";
|
|
return kAddIceCandidateFailNotReady;
|
|
}
|
|
|
|
if (!UseCandidate(ice_candidate)) {
|
|
return kAddIceCandidateFailNotUsable;
|
|
}
|
|
|
|
pc_->NoteUsageEvent(UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED);
|
|
|
|
return kAddIceCandidateSuccess;
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::AddIceCandidate(
|
|
std::unique_ptr<IceCandidateInterface> candidate,
|
|
std::function<void(RTCError)> callback) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Chain this operation. If asynchronous operations are pending on the chain,
|
|
// this operation will be queued to be invoked, otherwise the contents of the
|
|
// lambda will execute immediately.
|
|
operations_chain_->ChainOperation(
|
|
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
|
|
candidate = std::move(candidate), callback = std::move(callback)](
|
|
std::function<void()> operations_chain_callback) {
|
|
auto result =
|
|
this_weak_ptr
|
|
? this_weak_ptr->AddIceCandidateInternal(candidate.get())
|
|
: kAddIceCandidateFailClosed;
|
|
NoteAddIceCandidateResult(result);
|
|
operations_chain_callback();
|
|
if (result == kAddIceCandidateFailClosed) {
|
|
callback(RTCError(
|
|
RTCErrorType::INVALID_STATE,
|
|
"AddIceCandidate failed because the session was shut down"));
|
|
} else if (result != kAddIceCandidateSuccess &&
|
|
result != kAddIceCandidateFailNotReady) {
|
|
// Fail with an error type and message consistent with Chromium.
|
|
// TODO(hbos): Fail with error types according to spec.
|
|
callback(RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
|
|
"Error processing ICE candidate"));
|
|
} else {
|
|
callback(RTCError::OK());
|
|
}
|
|
});
|
|
}
|
|
|
|
bool SdpOfferAnswerHandler::RemoveIceCandidates(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::RemoveIceCandidates");
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (pc_->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;
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::AddLocalIceCandidate(
|
|
const JsepIceCandidate* candidate) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (local_description()) {
|
|
mutable_local_description()->AddCandidate(candidate);
|
|
}
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::RemoveLocalIceCandidates(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (local_description()) {
|
|
mutable_local_description()->RemoveCandidates(candidates);
|
|
}
|
|
}
|
|
|
|
const SessionDescriptionInterface* SdpOfferAnswerHandler::local_description()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return pending_local_description_ ? pending_local_description_.get()
|
|
: current_local_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface* SdpOfferAnswerHandler::remote_description()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return pending_remote_description_ ? pending_remote_description_.get()
|
|
: current_remote_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface*
|
|
SdpOfferAnswerHandler::current_local_description() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return current_local_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface*
|
|
SdpOfferAnswerHandler::current_remote_description() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return current_remote_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface*
|
|
SdpOfferAnswerHandler::pending_local_description() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return pending_local_description_.get();
|
|
}
|
|
|
|
const SessionDescriptionInterface*
|
|
SdpOfferAnswerHandler::pending_remote_description() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return pending_remote_description_.get();
|
|
}
|
|
|
|
PeerConnectionInterface::SignalingState SdpOfferAnswerHandler::signaling_state()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return signaling_state_;
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::ChangeSignalingState(
|
|
PeerConnectionInterface::SignalingState signaling_state) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (signaling_state_ == signaling_state) {
|
|
return;
|
|
}
|
|
RTC_LOG(LS_INFO) << "Session: " << pc_->session_id() << " Old state: "
|
|
<< GetSignalingStateString(signaling_state_)
|
|
<< " New state: "
|
|
<< GetSignalingStateString(signaling_state);
|
|
signaling_state_ = signaling_state;
|
|
pc_->Observer()->OnSignalingChange(signaling_state_);
|
|
}
|
|
|
|
RTCError SdpOfferAnswerHandler::UpdateSessionState(
|
|
SdpType type,
|
|
cricket::ContentSource source,
|
|
const cricket::SessionDescription* description,
|
|
const std::map<std::string, const cricket::ContentGroup*>&
|
|
bundle_groups_by_mid) {
|
|
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);
|
|
transceivers()->DiscardStableStates();
|
|
}
|
|
|
|
// Update internal objects according to the session description's media
|
|
// descriptions.
|
|
return PushdownMediaDescription(type, source, bundle_groups_by_mid);
|
|
}
|
|
|
|
bool SdpOfferAnswerHandler::ShouldFireNegotiationNeededEvent(
|
|
uint32_t event_id) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Plan B? Always fire to conform with useless legacy behavior.
|
|
if (!IsUnifiedPlan()) {
|
|
return true;
|
|
}
|
|
// The event ID has been invalidated. Either negotiation is no longer needed
|
|
// or a newer negotiation needed event has been generated.
|
|
if (event_id != negotiation_needed_event_id_) {
|
|
return false;
|
|
}
|
|
// The chain is no longer empty, update negotiation needed when it becomes
|
|
// empty. This should generate a newer negotiation needed event, making this
|
|
// one obsolete.
|
|
if (!operations_chain_->IsEmpty()) {
|
|
// Since we just suppressed an event that would have been fired, if
|
|
// negotiation is still needed by the time the chain becomes empty again, we
|
|
// must make sure to generate another event if negotiation is needed then.
|
|
// This happens when |is_negotiation_needed_| goes from false to true, so we
|
|
// set it to false until UpdateNegotiationNeeded() is called.
|
|
is_negotiation_needed_ = false;
|
|
update_negotiation_needed_on_empty_chain_ = true;
|
|
return false;
|
|
}
|
|
// We must not fire if the signaling state is no longer "stable". If
|
|
// negotiation is still needed when we return to "stable", a new negotiation
|
|
// needed event will be generated, so this one can safely be suppressed.
|
|
if (signaling_state_ != PeerConnectionInterface::kStable) {
|
|
return false;
|
|
}
|
|
// All checks have passed - please fire "negotiationneeded" now!
|
|
return true;
|
|
}
|
|
|
|
rtc::scoped_refptr<StreamCollectionInterface>
|
|
SdpOfferAnswerHandler::local_streams() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(!IsUnifiedPlan()) << "local_streams is not available with Unified "
|
|
"Plan SdpSemantics. Please use GetSenders "
|
|
"instead.";
|
|
return local_streams_;
|
|
}
|
|
|
|
rtc::scoped_refptr<StreamCollectionInterface>
|
|
SdpOfferAnswerHandler::remote_streams() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(!IsUnifiedPlan()) << "remote_streams is not available with Unified "
|
|
"Plan SdpSemantics. Please use GetReceivers "
|
|
"instead.";
|
|
return remote_streams_;
|
|
}
|
|
|
|
bool SdpOfferAnswerHandler::AddStream(MediaStreamInterface* local_stream) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(!IsUnifiedPlan()) << "AddStream is not available with Unified Plan "
|
|
"SdpSemantics. Please use AddTrack instead.";
|
|
if (pc_->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, &SdpOfferAnswerHandler::OnAudioTrackAdded);
|
|
observer->SignalAudioTrackRemoved.connect(
|
|
this, &SdpOfferAnswerHandler::OnAudioTrackRemoved);
|
|
observer->SignalVideoTrackAdded.connect(
|
|
this, &SdpOfferAnswerHandler::OnVideoTrackAdded);
|
|
observer->SignalVideoTrackRemoved.connect(
|
|
this, &SdpOfferAnswerHandler::OnVideoTrackRemoved);
|
|
stream_observers_.push_back(std::unique_ptr<MediaStreamObserver>(observer));
|
|
|
|
for (const auto& track : local_stream->GetAudioTracks()) {
|
|
rtp_manager()->AddAudioTrack(track.get(), local_stream);
|
|
}
|
|
for (const auto& track : local_stream->GetVideoTracks()) {
|
|
rtp_manager()->AddVideoTrack(track.get(), local_stream);
|
|
}
|
|
|
|
pc_->stats()->AddStream(local_stream);
|
|
UpdateNegotiationNeeded();
|
|
return true;
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::RemoveStream(MediaStreamInterface* local_stream) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(!IsUnifiedPlan()) << "RemoveStream is not available with Unified "
|
|
"Plan SdpSemantics. Please use RemoveTrack "
|
|
"instead.";
|
|
TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
|
|
if (!pc_->IsClosed()) {
|
|
for (const auto& track : local_stream->GetAudioTracks()) {
|
|
rtp_manager()->RemoveAudioTrack(track.get(), local_stream);
|
|
}
|
|
for (const auto& track : local_stream->GetVideoTracks()) {
|
|
rtp_manager()->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 (pc_->IsClosed()) {
|
|
return;
|
|
}
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::OnAudioTrackAdded(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (pc_->IsClosed()) {
|
|
return;
|
|
}
|
|
rtp_manager()->AddAudioTrack(track, stream);
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::OnAudioTrackRemoved(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (pc_->IsClosed()) {
|
|
return;
|
|
}
|
|
rtp_manager()->RemoveAudioTrack(track, stream);
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::OnVideoTrackAdded(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (pc_->IsClosed()) {
|
|
return;
|
|
}
|
|
rtp_manager()->AddVideoTrack(track, stream);
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::OnVideoTrackRemoved(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (pc_->IsClosed()) {
|
|
return;
|
|
}
|
|
rtp_manager()->RemoveVideoTrack(track, stream);
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
|
|
RTCError SdpOfferAnswerHandler::Rollback(SdpType desc_type) {
|
|
auto state = signaling_state();
|
|
if (state != PeerConnectionInterface::kHaveLocalOffer &&
|
|
state != PeerConnectionInterface::kHaveRemoteOffer) {
|
|
return RTCError(RTCErrorType::INVALID_STATE,
|
|
"Called in wrong signalingState: " +
|
|
GetSignalingStateString(signaling_state()));
|
|
}
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> all_added_streams;
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> all_removed_streams;
|
|
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> removed_receivers;
|
|
|
|
for (auto&& transceivers_stable_state_pair : transceivers()->StableStates()) {
|
|
auto transceiver = transceivers_stable_state_pair.first;
|
|
auto state = transceivers_stable_state_pair.second;
|
|
|
|
if (state.remote_stream_ids()) {
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> added_streams;
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
|
|
SetAssociatedRemoteStreams(transceiver->internal()->receiver_internal(),
|
|
state.remote_stream_ids().value(),
|
|
&added_streams, &removed_streams);
|
|
all_added_streams.insert(all_added_streams.end(), added_streams.begin(),
|
|
added_streams.end());
|
|
all_removed_streams.insert(all_removed_streams.end(),
|
|
removed_streams.begin(),
|
|
removed_streams.end());
|
|
if (!state.has_m_section() && !state.newly_created()) {
|
|
continue;
|
|
}
|
|
}
|
|
|
|
RTC_DCHECK(transceiver->internal()->mid().has_value());
|
|
DestroyTransceiverChannel(transceiver);
|
|
|
|
if (signaling_state() == PeerConnectionInterface::kHaveRemoteOffer &&
|
|
transceiver->receiver()) {
|
|
removed_receivers.push_back(transceiver->receiver());
|
|
}
|
|
if (state.newly_created()) {
|
|
if (transceiver->internal()->reused_for_addtrack()) {
|
|
transceiver->internal()->set_created_by_addtrack(true);
|
|
} else {
|
|
transceivers()->Remove(transceiver);
|
|
}
|
|
}
|
|
if (state.init_send_encodings()) {
|
|
transceiver->internal()->sender_internal()->set_init_send_encodings(
|
|
state.init_send_encodings().value());
|
|
}
|
|
transceiver->internal()->sender_internal()->set_transport(nullptr);
|
|
transceiver->internal()->receiver_internal()->set_transport(nullptr);
|
|
transceiver->internal()->set_mid(state.mid());
|
|
transceiver->internal()->set_mline_index(state.mline_index());
|
|
}
|
|
transport_controller()->RollbackTransports();
|
|
transceivers()->DiscardStableStates();
|
|
pending_local_description_.reset();
|
|
pending_remote_description_.reset();
|
|
ChangeSignalingState(PeerConnectionInterface::kStable);
|
|
|
|
// Once all processing has finished, fire off callbacks.
|
|
for (const auto& receiver : removed_receivers) {
|
|
pc_->Observer()->OnRemoveTrack(receiver);
|
|
}
|
|
for (const auto& stream : all_added_streams) {
|
|
pc_->Observer()->OnAddStream(stream);
|
|
}
|
|
for (const auto& stream : all_removed_streams) {
|
|
pc_->Observer()->OnRemoveStream(stream);
|
|
}
|
|
|
|
// The assumption is that in case of implicit rollback UpdateNegotiationNeeded
|
|
// gets called in SetRemoteDescription.
|
|
if (desc_type == SdpType::kRollback) {
|
|
UpdateNegotiationNeeded();
|
|
if (is_negotiation_needed_) {
|
|
// Legacy version.
|
|
pc_->Observer()->OnRenegotiationNeeded();
|
|
// Spec-compliant version; the event may get invalidated before firing.
|
|
GenerateNegotiationNeededEvent();
|
|
}
|
|
}
|
|
return RTCError::OK();
|
|
}
|
|
|
|
bool SdpOfferAnswerHandler::IsUnifiedPlan() const {
|
|
return pc_->IsUnifiedPlan();
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::OnOperationsChainEmpty() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (pc_->IsClosed() || !update_negotiation_needed_on_empty_chain_)
|
|
return;
|
|
update_negotiation_needed_on_empty_chain_ = false;
|
|
// Firing when chain is empty is only supported in Unified Plan to avoid Plan
|
|
// B regressions. (In Plan B, onnegotiationneeded is already broken anyway, so
|
|
// firing it even more might just be confusing.)
|
|
if (IsUnifiedPlan()) {
|
|
UpdateNegotiationNeeded();
|
|
}
|
|
}
|
|
|
|
absl::optional<bool> SdpOfferAnswerHandler::is_caller() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return is_caller_;
|
|
}
|
|
|
|
bool SdpOfferAnswerHandler::HasNewIceCredentials() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return local_ice_credentials_to_replace_->HasIceCredentials();
|
|
}
|
|
|
|
bool SdpOfferAnswerHandler::IceRestartPending(
|
|
const std::string& content_name) const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return pending_ice_restarts_.find(content_name) !=
|
|
pending_ice_restarts_.end();
|
|
}
|
|
|
|
bool SdpOfferAnswerHandler::NeedsIceRestart(
|
|
const std::string& content_name) const {
|
|
return pc_->NeedsIceRestart(content_name);
|
|
}
|
|
|
|
absl::optional<rtc::SSLRole> SdpOfferAnswerHandler::GetDtlsRole(
|
|
const std::string& mid) const {
|
|
return transport_controller()->GetDtlsRole(mid);
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::UpdateNegotiationNeeded() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (!IsUnifiedPlan()) {
|
|
pc_->Observer()->OnRenegotiationNeeded();
|
|
GenerateNegotiationNeededEvent();
|
|
return;
|
|
}
|
|
|
|
// In the spec, a task is queued here to run the following steps - this is
|
|
// meant to ensure we do not fire onnegotiationneeded prematurely if multiple
|
|
// changes are being made at once. In order to support Chromium's
|
|
// implementation where the JavaScript representation of the PeerConnection
|
|
// lives on a separate thread though, the queuing of a task is instead
|
|
// performed by the PeerConnectionObserver posting from the signaling thread
|
|
// to the JavaScript main thread that negotiation is needed. And because the
|
|
// Operations Chain lives on the WebRTC signaling thread,
|
|
// ShouldFireNegotiationNeededEvent() must be called before firing the event
|
|
// to ensure the Operations Chain is still empty and the event has not been
|
|
// invalidated.
|
|
|
|
// If connection's [[IsClosed]] slot is true, abort these steps.
|
|
if (pc_->IsClosed())
|
|
return;
|
|
|
|
// If connection's signaling state is not "stable", abort these steps.
|
|
if (signaling_state() != PeerConnectionInterface::kStable)
|
|
return;
|
|
|
|
// NOTE
|
|
// The negotiation-needed flag will be updated once the state transitions to
|
|
// "stable", as part of the steps for setting an RTCSessionDescription.
|
|
|
|
// If the result of checking if negotiation is needed is false, clear the
|
|
// negotiation-needed flag by setting connection's [[NegotiationNeeded]] slot
|
|
// to false, and abort these steps.
|
|
bool is_negotiation_needed = CheckIfNegotiationIsNeeded();
|
|
if (!is_negotiation_needed) {
|
|
is_negotiation_needed_ = false;
|
|
// Invalidate any negotiation needed event that may previosuly have been
|
|
// generated.
|
|
++negotiation_needed_event_id_;
|
|
return;
|
|
}
|
|
|
|
// If connection's [[NegotiationNeeded]] slot is already true, abort these
|
|
// steps.
|
|
if (is_negotiation_needed_)
|
|
return;
|
|
|
|
// Set connection's [[NegotiationNeeded]] slot to true.
|
|
is_negotiation_needed_ = true;
|
|
|
|
// Queue a task that runs the following steps:
|
|
// If connection's [[IsClosed]] slot is true, abort these steps.
|
|
// If connection's [[NegotiationNeeded]] slot is false, abort these steps.
|
|
// Fire an event named negotiationneeded at connection.
|
|
pc_->Observer()->OnRenegotiationNeeded();
|
|
// Fire the spec-compliant version; when ShouldFireNegotiationNeededEvent() is
|
|
// used in the task queued by the observer, this event will only fire when the
|
|
// chain is empty.
|
|
GenerateNegotiationNeededEvent();
|
|
}
|
|
|
|
bool SdpOfferAnswerHandler::CheckIfNegotiationIsNeeded() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// 1. If any implementation-specific negotiation is required, as described at
|
|
// the start of this section, return true.
|
|
|
|
// 2. If connection.[[LocalIceCredentialsToReplace]] is not empty, return
|
|
// true.
|
|
if (local_ice_credentials_to_replace_->HasIceCredentials()) {
|
|
return true;
|
|
}
|
|
|
|
// 3. Let description be connection.[[CurrentLocalDescription]].
|
|
const SessionDescriptionInterface* description = current_local_description();
|
|
if (!description)
|
|
return true;
|
|
|
|
// 4. If connection has created any RTCDataChannels, and no m= section in
|
|
// description has been negotiated yet for data, return true.
|
|
if (data_channel_controller()->HasSctpDataChannels()) {
|
|
if (!cricket::GetFirstDataContent(description->description()->contents()))
|
|
return true;
|
|
}
|
|
|
|
// 5. For each transceiver in connection's set of transceivers, perform the
|
|
// following checks:
|
|
for (const auto& transceiver : transceivers()->ListInternal()) {
|
|
const ContentInfo* current_local_msection =
|
|
FindTransceiverMSection(transceiver, description);
|
|
|
|
const ContentInfo* current_remote_msection =
|
|
FindTransceiverMSection(transceiver, current_remote_description());
|
|
|
|
// 5.4 If transceiver is stopped and is associated with an m= section,
|
|
// but the associated m= section is not yet rejected in
|
|
// connection.[[CurrentLocalDescription]] or
|
|
// connection.[[CurrentRemoteDescription]], return true.
|
|
if (transceiver->stopped()) {
|
|
RTC_DCHECK(transceiver->stopping());
|
|
if (current_local_msection && !current_local_msection->rejected &&
|
|
((current_remote_msection && !current_remote_msection->rejected) ||
|
|
!current_remote_msection)) {
|
|
return true;
|
|
}
|
|
continue;
|
|
}
|
|
|
|
// 5.1 If transceiver.[[Stopping]] is true and transceiver.[[Stopped]] is
|
|
// false, return true.
|
|
if (transceiver->stopping() && !transceiver->stopped())
|
|
return true;
|
|
|
|
// 5.2 If transceiver isn't stopped and isn't yet associated with an m=
|
|
// section in description, return true.
|
|
if (!current_local_msection)
|
|
return true;
|
|
|
|
const MediaContentDescription* current_local_media_description =
|
|
current_local_msection->media_description();
|
|
// 5.3 If transceiver isn't stopped and is associated with an m= section
|
|
// in description then perform the following checks:
|
|
|
|
// 5.3.1 If transceiver.[[Direction]] is "sendrecv" or "sendonly", and the
|
|
// associated m= section in description either doesn't contain a single
|
|
// "a=msid" line, or the number of MSIDs from the "a=msid" lines in this
|
|
// m= section, or the MSID values themselves, differ from what is in
|
|
// transceiver.sender.[[AssociatedMediaStreamIds]], return true.
|
|
if (RtpTransceiverDirectionHasSend(transceiver->direction())) {
|
|
if (current_local_media_description->streams().size() == 0)
|
|
return true;
|
|
|
|
std::vector<std::string> msection_msids;
|
|
for (const auto& stream : current_local_media_description->streams()) {
|
|
for (const std::string& msid : stream.stream_ids())
|
|
msection_msids.push_back(msid);
|
|
}
|
|
|
|
std::vector<std::string> transceiver_msids =
|
|
transceiver->sender()->stream_ids();
|
|
if (msection_msids.size() != transceiver_msids.size())
|
|
return true;
|
|
|
|
absl::c_sort(transceiver_msids);
|
|
absl::c_sort(msection_msids);
|
|
if (transceiver_msids != msection_msids)
|
|
return true;
|
|
}
|
|
|
|
// 5.3.2 If description is of type "offer", and the direction of the
|
|
// associated m= section in neither connection.[[CurrentLocalDescription]]
|
|
// nor connection.[[CurrentRemoteDescription]] matches
|
|
// transceiver.[[Direction]], return true.
|
|
if (description->GetType() == SdpType::kOffer) {
|
|
if (!current_remote_description())
|
|
return true;
|
|
|
|
if (!current_remote_msection)
|
|
return true;
|
|
|
|
RtpTransceiverDirection current_local_direction =
|
|
current_local_media_description->direction();
|
|
RtpTransceiverDirection current_remote_direction =
|
|
current_remote_msection->media_description()->direction();
|
|
if (transceiver->direction() != current_local_direction &&
|
|
transceiver->direction() !=
|
|
RtpTransceiverDirectionReversed(current_remote_direction)) {
|
|
return true;
|
|
}
|
|
}
|
|
|
|
// 5.3.3 If description is of type "answer", and the direction of the
|
|
// associated m= section in the description does not match
|
|
// transceiver.[[Direction]] intersected with the offered direction (as
|
|
// described in [JSEP] (section 5.3.1.)), return true.
|
|
if (description->GetType() == SdpType::kAnswer) {
|
|
if (!remote_description())
|
|
return true;
|
|
|
|
const ContentInfo* offered_remote_msection =
|
|
FindTransceiverMSection(transceiver, remote_description());
|
|
|
|
RtpTransceiverDirection offered_direction =
|
|
offered_remote_msection
|
|
? offered_remote_msection->media_description()->direction()
|
|
: RtpTransceiverDirection::kInactive;
|
|
|
|
if (current_local_media_description->direction() !=
|
|
(RtpTransceiverDirectionIntersection(
|
|
transceiver->direction(),
|
|
RtpTransceiverDirectionReversed(offered_direction)))) {
|
|
return true;
|
|
}
|
|
}
|
|
}
|
|
|
|
// If all the preceding checks were performed and true was not returned,
|
|
// nothing remains to be negotiated; return false.
|
|
return false;
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::GenerateNegotiationNeededEvent() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
++negotiation_needed_event_id_;
|
|
pc_->Observer()->OnNegotiationNeededEvent(negotiation_needed_event_id_);
|
|
}
|
|
|
|
RTCError SdpOfferAnswerHandler::ValidateSessionDescription(
|
|
const SessionDescriptionInterface* sdesc,
|
|
cricket::ContentSource source,
|
|
const std::map<std::string, const cricket::ContentGroup*>&
|
|
bundle_groups_by_mid) {
|
|
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 ||
|
|
pc_->dtls_enabled()) {
|
|
RTCError crypto_error = VerifyCrypto(
|
|
sdesc->description(), pc_->dtls_enabled(), bundle_groups_by_mid);
|
|
if (!crypto_error.ok()) {
|
|
return crypto_error;
|
|
}
|
|
}
|
|
|
|
// Verify ice-ufrag and ice-pwd.
|
|
if (!VerifyIceUfragPwdPresent(sdesc->description(), bundle_groups_by_mid)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
kSdpWithoutIceUfragPwd);
|
|
}
|
|
|
|
if (!pc_->ValidateBundleSettings(sdesc->description(),
|
|
bundle_groups_by_mid)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
kBundleWithoutRtcpMux);
|
|
}
|
|
|
|
// TODO(skvlad): When the local rtcp-mux policy is Require, reject any
|
|
// m-lines that do not rtcp-mux enabled.
|
|
|
|
// Verify m-lines in Answer when compared against Offer.
|
|
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
|
|
// With an answer we want to compare the new answer session description with
|
|
// the offer's session description from the current negotiation.
|
|
const cricket::SessionDescription* offer_desc =
|
|
(source == cricket::CS_LOCAL) ? remote_description()->description()
|
|
: local_description()->description();
|
|
if (!MediaSectionsHaveSameCount(*offer_desc, *sdesc->description()) ||
|
|
!MediaSectionsInSameOrder(*offer_desc, nullptr, *sdesc->description(),
|
|
type)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
kMlineMismatchInAnswer);
|
|
}
|
|
} else {
|
|
// The re-offers should respect the order of m= sections in current
|
|
// description. See RFC3264 Section 8 paragraph 4 for more details.
|
|
// With a re-offer, either the current local or current remote descriptions
|
|
// could be the most up to date, so we would like to check against both of
|
|
// them if they exist. It could be the case that one of them has a 0 port
|
|
// for a media section, but the other does not. This is important to check
|
|
// against in the case that we are recycling an m= section.
|
|
const cricket::SessionDescription* current_desc = nullptr;
|
|
const cricket::SessionDescription* secondary_current_desc = nullptr;
|
|
if (local_description()) {
|
|
current_desc = local_description()->description();
|
|
if (remote_description()) {
|
|
secondary_current_desc = remote_description()->description();
|
|
}
|
|
} else if (remote_description()) {
|
|
current_desc = remote_description()->description();
|
|
}
|
|
if (current_desc &&
|
|
!MediaSectionsInSameOrder(*current_desc, secondary_current_desc,
|
|
*sdesc->description(), type)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
kMlineMismatchInSubsequentOffer);
|
|
}
|
|
}
|
|
|
|
if (IsUnifiedPlan()) {
|
|
// Ensure that each audio and video media section has at most one
|
|
// "StreamParams". This will return an error if receiving a session
|
|
// description from a "Plan B" endpoint which adds multiple tracks of the
|
|
// same type. With Unified Plan, there can only be at most one track per
|
|
// media section.
|
|
for (const ContentInfo& content : sdesc->description()->contents()) {
|
|
const MediaContentDescription& desc = *content.media_description();
|
|
if ((desc.type() == cricket::MEDIA_TYPE_AUDIO ||
|
|
desc.type() == cricket::MEDIA_TYPE_VIDEO) &&
|
|
desc.streams().size() > 1u) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"Media section has more than one track specified "
|
|
"with a=ssrc lines which is not supported with "
|
|
"Unified Plan.");
|
|
}
|
|
}
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
RTCError SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels(
|
|
cricket::ContentSource source,
|
|
const SessionDescriptionInterface& new_session,
|
|
const SessionDescriptionInterface* old_local_description,
|
|
const SessionDescriptionInterface* old_remote_description,
|
|
const std::map<std::string, const cricket::ContentGroup*>&
|
|
bundle_groups_by_mid) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
|
|
if (new_session.GetType() == SdpType::kOffer) {
|
|
// If the BUNDLE policy is max-bundle, then we know for sure that all
|
|
// transports will be bundled from the start. Return an error if max-bundle
|
|
// is specified but the session description does not have a BUNDLE group.
|
|
if (pc_->configuration()->bundle_policy ==
|
|
PeerConnectionInterface::kBundlePolicyMaxBundle &&
|
|
bundle_groups_by_mid.empty()) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"max-bundle configured but session description "
|
|
"has no BUNDLE group");
|
|
}
|
|
}
|
|
|
|
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);
|
|
auto it = bundle_groups_by_mid.find(new_content.name);
|
|
const cricket::ContentGroup* bundle_group =
|
|
it != bundle_groups_by_mid.end() ? it->second : nullptr;
|
|
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()) {
|
|
// In the case where a transceiver is rejected locally, we don't
|
|
// expect to find a transceiver, but might find it in the case
|
|
// where state is still "stopping", not "stopped".
|
|
if (new_content.rejected) {
|
|
continue;
|
|
}
|
|
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 (pc_->GetDataMid() && new_content.name != *(pc_->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 if (media_type == cricket::MEDIA_TYPE_UNSUPPORTED) {
|
|
RTC_LOG(LS_INFO) << "Ignoring unsupported media type";
|
|
} else {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Unknown section type.");
|
|
}
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
|
|
SdpOfferAnswerHandler::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 RTC_DCHECK_IS_ON
|
|
// 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), the transceiver
|
|
// should have been removed by RemoveStoppedtransceivers()->
|
|
if (IsMediaSectionBeingRecycled(type, content, old_local_content,
|
|
old_remote_content)) {
|
|
const std::string& old_mid =
|
|
(old_local_content && old_local_content->rejected)
|
|
? old_local_content->name
|
|
: old_remote_content->name;
|
|
auto old_transceiver = transceivers()->FindByMid(old_mid);
|
|
// The transceiver should be disassociated in RemoveStoppedTransceivers()
|
|
RTC_DCHECK(!old_transceiver);
|
|
}
|
|
#endif
|
|
|
|
const MediaContentDescription* media_desc = content.media_description();
|
|
auto transceiver = transceivers()->FindByMid(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 = transceivers()->FindByMLineIndex(mline_index);
|
|
}
|
|
if (!transceiver) {
|
|
// This may happen normally when media sections are rejected.
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"Transceiver not found based on m-line index");
|
|
}
|
|
} else {
|
|
RTC_DCHECK_EQ(source, cricket::CS_REMOTE);
|
|
// If the m= section is sendrecv or recvonly, and there are RtpTransceivers
|
|
// of the same type...
|
|
// When simulcast is requested, a transceiver cannot be associated because
|
|
// AddTrack cannot be called to initialize it.
|
|
if (!transceiver &&
|
|
RtpTransceiverDirectionHasRecv(media_desc->direction()) &&
|
|
!media_desc->HasSimulcast()) {
|
|
transceiver = FindAvailableTransceiverToReceive(media_desc->type());
|
|
}
|
|
// If no RtpTransceiver was found in the previous step, create one with a
|
|
// recvonly direction.
|
|
if (!transceiver) {
|
|
RTC_LOG(LS_INFO) << "Adding "
|
|
<< cricket::MediaTypeToString(media_desc->type())
|
|
<< " transceiver for MID=" << content.name
|
|
<< " at i=" << mline_index
|
|
<< " in response to the remote description.";
|
|
std::string sender_id = rtc::CreateRandomUuid();
|
|
std::vector<RtpEncodingParameters> send_encodings =
|
|
GetSendEncodingsFromRemoteDescription(*media_desc);
|
|
auto sender = rtp_manager()->CreateSender(media_desc->type(), sender_id,
|
|
nullptr, {}, send_encodings);
|
|
std::string receiver_id;
|
|
if (!media_desc->streams().empty()) {
|
|
receiver_id = media_desc->streams()[0].id;
|
|
} else {
|
|
receiver_id = rtc::CreateRandomUuid();
|
|
}
|
|
auto receiver =
|
|
rtp_manager()->CreateReceiver(media_desc->type(), receiver_id);
|
|
transceiver = rtp_manager()->CreateAndAddTransceiver(sender, receiver);
|
|
transceiver->internal()->set_direction(
|
|
RtpTransceiverDirection::kRecvOnly);
|
|
if (type == SdpType::kOffer) {
|
|
transceivers()->StableState(transceiver)->set_newly_created();
|
|
}
|
|
}
|
|
|
|
RTC_DCHECK(transceiver);
|
|
|
|
// Check if the offer indicated simulcast but the answer rejected it.
|
|
// This can happen when simulcast is not supported on the remote party.
|
|
if (SimulcastIsRejected(old_local_content, *media_desc)) {
|
|
RTC_HISTOGRAM_BOOLEAN(kSimulcastDisabled, true);
|
|
RTCError error =
|
|
DisableSimulcastInSender(transceiver->internal()->sender_internal());
|
|
if (!error.ok()) {
|
|
RTC_LOG(LS_ERROR) << "Failed to remove rejected simulcast.";
|
|
return std::move(error);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (transceiver->media_type() != media_desc->type()) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::INVALID_PARAMETER,
|
|
"Transceiver type does not match media description type.");
|
|
}
|
|
|
|
if (media_desc->HasSimulcast()) {
|
|
std::vector<SimulcastLayer> layers =
|
|
source == cricket::CS_LOCAL
|
|
? media_desc->simulcast_description().send_layers().GetAllLayers()
|
|
: media_desc->simulcast_description()
|
|
.receive_layers()
|
|
.GetAllLayers();
|
|
RTCError error = UpdateSimulcastLayerStatusInSender(
|
|
layers, transceiver->internal()->sender_internal());
|
|
if (!error.ok()) {
|
|
RTC_LOG(LS_ERROR) << "Failed updating status for simulcast layers.";
|
|
return std::move(error);
|
|
}
|
|
}
|
|
if (type == SdpType::kOffer) {
|
|
bool state_changes = transceiver->internal()->mid() != content.name ||
|
|
transceiver->internal()->mline_index() != mline_index;
|
|
if (state_changes) {
|
|
transceivers()
|
|
->StableState(transceiver)
|
|
->SetMSectionIfUnset(transceiver->internal()->mid(),
|
|
transceiver->internal()->mline_index());
|
|
}
|
|
}
|
|
// Associate the found or created RtpTransceiver with the m= section by
|
|
// setting the value of the RtpTransceiver's mid property to the MID of the m=
|
|
// section, and establish a mapping between the transceiver and the index of
|
|
// the m= section.
|
|
transceiver->internal()->set_mid(content.name);
|
|
transceiver->internal()->set_mline_index(mline_index);
|
|
return std::move(transceiver);
|
|
}
|
|
|
|
RTCError SdpOfferAnswerHandler::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 SdpOfferAnswerHandler::UpdateDataChannel(
|
|
cricket::ContentSource source,
|
|
const cricket::ContentInfo& content,
|
|
const cricket::ContentGroup* bundle_group) {
|
|
if (content.rejected) {
|
|
RTC_LOG(LS_INFO) << "Rejected data channel, mid=" << content.mid();
|
|
DestroyDataChannelTransport();
|
|
} else {
|
|
if (!data_channel_controller()->data_channel_transport()) {
|
|
RTC_LOG(LS_INFO) << "Creating data channel, mid=" << content.mid();
|
|
if (!CreateDataChannel(content.name)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Failed to create data channel.");
|
|
}
|
|
}
|
|
}
|
|
return RTCError::OK();
|
|
}
|
|
|
|
bool SdpOfferAnswerHandler::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 SdpOfferAnswerHandler::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);
|
|
}
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::FillInMissingRemoteMids(
|
|
cricket::SessionDescription* new_remote_description) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(new_remote_description);
|
|
const cricket::ContentInfos no_infos;
|
|
const cricket::ContentInfos& local_contents =
|
|
(local_description() ? local_description()->description()->contents()
|
|
: no_infos);
|
|
const cricket::ContentInfos& remote_contents =
|
|
(remote_description() ? remote_description()->description()->contents()
|
|
: no_infos);
|
|
for (size_t i = 0; i < new_remote_description->contents().size(); ++i) {
|
|
cricket::ContentInfo& content = new_remote_description->contents()[i];
|
|
if (!content.name.empty()) {
|
|
continue;
|
|
}
|
|
std::string new_mid;
|
|
absl::string_view source_explanation;
|
|
if (IsUnifiedPlan()) {
|
|
if (i < local_contents.size()) {
|
|
new_mid = local_contents[i].name;
|
|
source_explanation = "from the matching local media section";
|
|
} else if (i < remote_contents.size()) {
|
|
new_mid = remote_contents[i].name;
|
|
source_explanation = "from the matching previous remote media section";
|
|
} else {
|
|
new_mid = mid_generator_.GenerateString();
|
|
source_explanation = "generated just now";
|
|
}
|
|
} else {
|
|
new_mid = std::string(
|
|
GetDefaultMidForPlanB(content.media_description()->type()));
|
|
source_explanation = "to match pre-existing behavior";
|
|
}
|
|
RTC_DCHECK(!new_mid.empty());
|
|
content.name = new_mid;
|
|
new_remote_description->transport_infos()[i].content_name = new_mid;
|
|
RTC_LOG(LS_INFO) << "SetRemoteDescription: Remote media section at i=" << i
|
|
<< " is missing an a=mid line. Filling in the value '"
|
|
<< new_mid << "' " << source_explanation << ".";
|
|
}
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
SdpOfferAnswerHandler::FindAvailableTransceiverToReceive(
|
|
cricket::MediaType media_type) const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
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()->List()) {
|
|
if (transceiver->media_type() == media_type &&
|
|
transceiver->internal()->created_by_addtrack() && !transceiver->mid() &&
|
|
!transceiver->stopped()) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
const cricket::ContentInfo*
|
|
SdpOfferAnswerHandler::FindMediaSectionForTransceiver(
|
|
const RtpTransceiver* transceiver,
|
|
const SessionDescriptionInterface* sdesc) const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(transceiver);
|
|
RTC_DCHECK(sdesc);
|
|
if (IsUnifiedPlan()) {
|
|
if (!transceiver->mid()) {
|
|
// This transceiver is not associated with a media section yet.
|
|
return nullptr;
|
|
}
|
|
return sdesc->description()->GetContentByName(*transceiver->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());
|
|
}
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::GetOptionsForOffer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
ExtractSharedMediaSessionOptions(offer_answer_options, session_options);
|
|
|
|
if (IsUnifiedPlan()) {
|
|
GetOptionsForUnifiedPlanOffer(offer_answer_options, session_options);
|
|
} else {
|
|
GetOptionsForPlanBOffer(offer_answer_options, session_options);
|
|
}
|
|
|
|
// Apply ICE restart flag and renomination flag.
|
|
bool ice_restart = offer_answer_options.ice_restart || HasNewIceCredentials();
|
|
for (auto& options : session_options->media_description_options) {
|
|
options.transport_options.ice_restart = ice_restart;
|
|
options.transport_options.enable_ice_renomination =
|
|
pc_->configuration()->enable_ice_renomination;
|
|
}
|
|
|
|
session_options->rtcp_cname = rtcp_cname_;
|
|
session_options->crypto_options = pc_->GetCryptoOptions();
|
|
session_options->pooled_ice_credentials =
|
|
pc_->network_thread()->Invoke<std::vector<cricket::IceParameters>>(
|
|
RTC_FROM_HERE,
|
|
[this] { return port_allocator()->GetPooledIceCredentials(); });
|
|
session_options->offer_extmap_allow_mixed =
|
|
pc_->configuration()->offer_extmap_allow_mixed;
|
|
|
|
// Allow fallback for using obsolete SCTP syntax.
|
|
// Note that the default in |session_options| is true, while
|
|
// the default in |options| is false.
|
|
session_options->use_obsolete_sctp_sdp =
|
|
offer_answer_options.use_obsolete_sctp_sdp;
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::GetOptionsForPlanBOffer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
// Figure out transceiver directional preferences.
|
|
bool send_audio =
|
|
!rtp_manager()->GetAudioTransceiver()->internal()->senders().empty();
|
|
bool send_video =
|
|
!rtp_manager()->GetVideoTransceiver()->internal()->senders().empty();
|
|
|
|
// By default, generate sendrecv/recvonly m= sections.
|
|
bool recv_audio = true;
|
|
bool recv_video = true;
|
|
|
|
// By default, only offer a new m= section if we have media to send with it.
|
|
bool offer_new_audio_description = send_audio;
|
|
bool offer_new_video_description = send_video;
|
|
bool offer_new_data_description =
|
|
data_channel_controller()->HasDataChannels();
|
|
|
|
// The "offer_to_receive_X" options allow those defaults to be overridden.
|
|
if (offer_answer_options.offer_to_receive_audio !=
|
|
PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined) {
|
|
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
|
|
offer_new_audio_description =
|
|
offer_new_audio_description ||
|
|
(offer_answer_options.offer_to_receive_audio > 0);
|
|
}
|
|
if (offer_answer_options.offer_to_receive_video !=
|
|
RTCOfferAnswerOptions::kUndefined) {
|
|
recv_video = (offer_answer_options.offer_to_receive_video > 0);
|
|
offer_new_video_description =
|
|
offer_new_video_description ||
|
|
(offer_answer_options.offer_to_receive_video > 0);
|
|
}
|
|
|
|
absl::optional<size_t> audio_index;
|
|
absl::optional<size_t> video_index;
|
|
absl::optional<size_t> data_index;
|
|
// If a current description exists, generate m= sections in the same order,
|
|
// using the first audio/video/data section that appears and rejecting
|
|
// extraneous ones.
|
|
if (local_description()) {
|
|
GenerateMediaDescriptionOptions(
|
|
local_description(),
|
|
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
|
|
RtpTransceiverDirectionFromSendRecv(send_video, recv_video),
|
|
&audio_index, &video_index, &data_index, session_options);
|
|
}
|
|
|
|
// Add audio/video/data m= sections to the end if needed.
|
|
if (!audio_index && offer_new_audio_description) {
|
|
cricket::MediaDescriptionOptions options(
|
|
cricket::MEDIA_TYPE_AUDIO, cricket::CN_AUDIO,
|
|
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio), false);
|
|
options.header_extensions =
|
|
channel_manager()->GetSupportedAudioRtpHeaderExtensions();
|
|
session_options->media_description_options.push_back(options);
|
|
audio_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
if (!video_index && offer_new_video_description) {
|
|
cricket::MediaDescriptionOptions options(
|
|
cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO,
|
|
RtpTransceiverDirectionFromSendRecv(send_video, recv_video), false);
|
|
options.header_extensions =
|
|
channel_manager()->GetSupportedVideoRtpHeaderExtensions();
|
|
session_options->media_description_options.push_back(options);
|
|
video_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
if (!data_index && offer_new_data_description) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForActiveData(cricket::CN_DATA));
|
|
data_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
|
|
cricket::MediaDescriptionOptions* audio_media_description_options =
|
|
!audio_index ? nullptr
|
|
: &session_options->media_description_options[*audio_index];
|
|
cricket::MediaDescriptionOptions* video_media_description_options =
|
|
!video_index ? nullptr
|
|
: &session_options->media_description_options[*video_index];
|
|
|
|
AddPlanBRtpSenderOptions(rtp_manager()->GetSendersInternal(),
|
|
audio_media_description_options,
|
|
video_media_description_options,
|
|
offer_answer_options.num_simulcast_layers);
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::GetOptionsForUnifiedPlanOffer(
|
|
const RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
// Rules for generating an offer are dictated by JSEP sections 5.2.1 (Initial
|
|
// Offers) and 5.2.2 (Subsequent Offers).
|
|
RTC_DCHECK_EQ(session_options->media_description_options.size(), 0);
|
|
const ContentInfos no_infos;
|
|
const ContentInfos& local_contents =
|
|
(local_description() ? local_description()->description()->contents()
|
|
: no_infos);
|
|
const ContentInfos& remote_contents =
|
|
(remote_description() ? remote_description()->description()->contents()
|
|
: no_infos);
|
|
// The mline indices that can be recycled. New transceivers should reuse these
|
|
// slots first.
|
|
std::queue<size_t> recycleable_mline_indices;
|
|
// First, go through each media section that exists in either the local or
|
|
// remote description and generate a media section in this offer for the
|
|
// associated transceiver. If a media section can be recycled, generate a
|
|
// default, rejected media section here that can be later overwritten.
|
|
for (size_t i = 0;
|
|
i < std::max(local_contents.size(), remote_contents.size()); ++i) {
|
|
// Either |local_content| or |remote_content| is non-null.
|
|
const ContentInfo* local_content =
|
|
(i < local_contents.size() ? &local_contents[i] : nullptr);
|
|
const ContentInfo* current_local_content =
|
|
GetContentByIndex(current_local_description(), i);
|
|
const ContentInfo* remote_content =
|
|
(i < remote_contents.size() ? &remote_contents[i] : nullptr);
|
|
const ContentInfo* current_remote_content =
|
|
GetContentByIndex(current_remote_description(), i);
|
|
bool had_been_rejected =
|
|
(current_local_content && current_local_content->rejected) ||
|
|
(current_remote_content && current_remote_content->rejected);
|
|
const std::string& mid =
|
|
(local_content ? local_content->name : remote_content->name);
|
|
cricket::MediaType media_type =
|
|
(local_content ? local_content->media_description()->type()
|
|
: remote_content->media_description()->type());
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO) {
|
|
// A media section is considered eligible for recycling if it is marked as
|
|
// rejected in either the current local or current remote description.
|
|
auto transceiver = transceivers()->FindByMid(mid);
|
|
if (!transceiver) {
|
|
// No associated transceiver. The media section has been stopped.
|
|
recycleable_mline_indices.push(i);
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(media_type, mid,
|
|
RtpTransceiverDirection::kInactive,
|
|
/*stopped=*/true));
|
|
} else {
|
|
// NOTE: a stopping transceiver should be treated as a stopped one in
|
|
// createOffer as specified in
|
|
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-createoffer.
|
|
if (had_been_rejected && transceiver->stopping()) {
|
|
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->internal(), mid,
|
|
/*is_create_offer=*/true));
|
|
// 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 if (media_type == cricket::MEDIA_TYPE_UNSUPPORTED) {
|
|
RTC_DCHECK(local_content->rejected);
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(media_type, mid,
|
|
RtpTransceiverDirection::kInactive,
|
|
/*stopped=*/true));
|
|
} else {
|
|
RTC_CHECK_EQ(cricket::MEDIA_TYPE_DATA, media_type);
|
|
if (had_been_rejected) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForRejectedData(mid));
|
|
} else {
|
|
RTC_CHECK(pc_->GetDataMid());
|
|
if (mid == *(pc_->GetDataMid())) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForActiveData(mid));
|
|
} else {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForRejectedData(mid));
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// Next, look for transceivers that are newly added (that is, are not stopped
|
|
// and not associated). Reuse media sections marked as recyclable first,
|
|
// otherwise append to the end of the offer. New media sections should be
|
|
// added in the order they were added to the PeerConnection.
|
|
for (const auto& transceiver : transceivers()->ListInternal()) {
|
|
if (transceiver->mid() || transceiver->stopping()) {
|
|
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_.GenerateString(),
|
|
/*is_create_offer=*/true);
|
|
} else {
|
|
mline_index = session_options->media_description_options.size();
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForTransceiver(
|
|
transceiver, mid_generator_.GenerateString(),
|
|
/*is_create_offer=*/true));
|
|
}
|
|
// See comment above for why CreateOffer changes the transceiver's state.
|
|
transceiver->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 (!pc_->GetDataMid() && data_channel_controller()->HasDataChannels()) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForActiveData(
|
|
mid_generator_.GenerateString()));
|
|
}
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::GetOptionsForAnswer(
|
|
const RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
ExtractSharedMediaSessionOptions(offer_answer_options, session_options);
|
|
|
|
if (IsUnifiedPlan()) {
|
|
GetOptionsForUnifiedPlanAnswer(offer_answer_options, session_options);
|
|
} else {
|
|
GetOptionsForPlanBAnswer(offer_answer_options, session_options);
|
|
}
|
|
|
|
// Apply ICE renomination flag.
|
|
for (auto& options : session_options->media_description_options) {
|
|
options.transport_options.enable_ice_renomination =
|
|
pc_->configuration()->enable_ice_renomination;
|
|
}
|
|
|
|
session_options->rtcp_cname = rtcp_cname_;
|
|
session_options->crypto_options = pc_->GetCryptoOptions();
|
|
session_options->pooled_ice_credentials =
|
|
pc_->network_thread()->Invoke<std::vector<cricket::IceParameters>>(
|
|
RTC_FROM_HERE,
|
|
[this] { return port_allocator()->GetPooledIceCredentials(); });
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::GetOptionsForPlanBAnswer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
// Figure out transceiver directional preferences.
|
|
bool send_audio =
|
|
!rtp_manager()->GetAudioTransceiver()->internal()->senders().empty();
|
|
bool send_video =
|
|
!rtp_manager()->GetVideoTransceiver()->internal()->senders().empty();
|
|
|
|
// By default, generate sendrecv/recvonly m= sections. The direction is also
|
|
// restricted by the direction in the offer.
|
|
bool recv_audio = true;
|
|
bool recv_video = true;
|
|
|
|
// The "offer_to_receive_X" options allow those defaults to be overridden.
|
|
if (offer_answer_options.offer_to_receive_audio !=
|
|
RTCOfferAnswerOptions::kUndefined) {
|
|
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
|
|
}
|
|
if (offer_answer_options.offer_to_receive_video !=
|
|
RTCOfferAnswerOptions::kUndefined) {
|
|
recv_video = (offer_answer_options.offer_to_receive_video > 0);
|
|
}
|
|
|
|
absl::optional<size_t> audio_index;
|
|
absl::optional<size_t> video_index;
|
|
absl::optional<size_t> data_index;
|
|
|
|
// Generate m= sections that match those in the offer.
|
|
// Note that mediasession.cc will handle intersection our preferred
|
|
// direction with the offered direction.
|
|
GenerateMediaDescriptionOptions(
|
|
remote_description(),
|
|
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
|
|
RtpTransceiverDirectionFromSendRecv(send_video, recv_video), &audio_index,
|
|
&video_index, &data_index, session_options);
|
|
|
|
cricket::MediaDescriptionOptions* audio_media_description_options =
|
|
!audio_index ? nullptr
|
|
: &session_options->media_description_options[*audio_index];
|
|
cricket::MediaDescriptionOptions* video_media_description_options =
|
|
!video_index ? nullptr
|
|
: &session_options->media_description_options[*video_index];
|
|
|
|
AddPlanBRtpSenderOptions(rtp_manager()->GetSendersInternal(),
|
|
audio_media_description_options,
|
|
video_media_description_options,
|
|
offer_answer_options.num_simulcast_layers);
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::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 = transceivers()->FindByMid(content.name);
|
|
if (transceiver) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForTransceiver(
|
|
transceiver->internal(), content.name,
|
|
/*is_create_offer=*/false));
|
|
} else {
|
|
// This should only happen with rejected transceivers.
|
|
RTC_DCHECK(content.rejected);
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(media_type, content.name,
|
|
RtpTransceiverDirection::kInactive,
|
|
/*stopped=*/true));
|
|
}
|
|
} else if (media_type == cricket::MEDIA_TYPE_UNSUPPORTED) {
|
|
RTC_DCHECK(content.rejected);
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(media_type, content.name,
|
|
RtpTransceiverDirection::kInactive,
|
|
/*stopped=*/true));
|
|
} 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 (content.rejected || content.name != *(pc_->GetDataMid())) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForRejectedData(content.name));
|
|
} else {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForActiveData(content.name));
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
const char* SdpOfferAnswerHandler::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 SdpOfferAnswerHandler::GetSessionErrorMsg() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
rtc::StringBuilder desc;
|
|
desc << kSessionError << SessionErrorToString(session_error()) << ". ";
|
|
desc << kSessionErrorDesc << session_error_desc() << ".";
|
|
return desc.Release();
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::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 SdpOfferAnswerHandler::HandleLegacyOfferOptions(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
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 SdpOfferAnswerHandler::RemoveRecvDirectionFromReceivingTransceiversOfType(
|
|
cricket::MediaType media_type) {
|
|
for (const auto& transceiver : GetReceivingTransceiversOfType(media_type)) {
|
|
RtpTransceiverDirection new_direction =
|
|
RtpTransceiverDirectionWithRecvSet(transceiver->direction(), false);
|
|
if (new_direction != transceiver->direction()) {
|
|
RTC_LOG(LS_INFO) << "Changing " << cricket::MediaTypeToString(media_type)
|
|
<< " transceiver (MID="
|
|
<< transceiver->mid().value_or("<not set>") << ") from "
|
|
<< RtpTransceiverDirectionToString(
|
|
transceiver->direction())
|
|
<< " to "
|
|
<< RtpTransceiverDirectionToString(new_direction)
|
|
<< " since CreateOffer specified offer_to_receive=0";
|
|
transceiver->internal()->set_direction(new_direction);
|
|
}
|
|
}
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::AddUpToOneReceivingTransceiverOfType(
|
|
cricket::MediaType media_type) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (GetReceivingTransceiversOfType(media_type).empty()) {
|
|
RTC_LOG(LS_INFO)
|
|
<< "Adding one recvonly " << cricket::MediaTypeToString(media_type)
|
|
<< " transceiver since CreateOffer specified offer_to_receive=1";
|
|
RtpTransceiverInit init;
|
|
init.direction = RtpTransceiverDirection::kRecvOnly;
|
|
pc_->AddTransceiver(media_type, nullptr, init,
|
|
/*update_negotiation_needed=*/false);
|
|
}
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
|
|
SdpOfferAnswerHandler::GetReceivingTransceiversOfType(
|
|
cricket::MediaType media_type) {
|
|
std::vector<
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
|
|
receiving_transceivers;
|
|
for (const auto& transceiver : transceivers()->List()) {
|
|
if (!transceiver->stopped() && transceiver->media_type() == media_type &&
|
|
RtpTransceiverDirectionHasRecv(transceiver->direction())) {
|
|
receiving_transceivers.push_back(transceiver);
|
|
}
|
|
}
|
|
return receiving_transceivers;
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::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 SdpOfferAnswerHandler::RemoveRemoteStreamsIfEmpty(
|
|
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& remote_streams,
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// TODO(https://crbug.com/webrtc/9480): When we use stream IDs instead of
|
|
// streams, see if the stream was removed by checking if this was the last
|
|
// receiver with that stream ID.
|
|
for (const auto& remote_stream : remote_streams) {
|
|
if (remote_stream->GetAudioTracks().empty() &&
|
|
remote_stream->GetVideoTracks().empty()) {
|
|
remote_streams_->RemoveStream(remote_stream);
|
|
removed_streams->push_back(remote_stream);
|
|
}
|
|
}
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::RemoveSenders(cricket::MediaType media_type) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
UpdateLocalSenders(std::vector<cricket::StreamParams>(), media_type);
|
|
UpdateRemoteSendersList(std::vector<cricket::StreamParams>(), false,
|
|
media_type, nullptr);
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::UpdateLocalSenders(
|
|
const std::vector<cricket::StreamParams>& streams,
|
|
cricket::MediaType media_type) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
std::vector<RtpSenderInfo>* current_senders =
|
|
rtp_manager()->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) {
|
|
rtp_manager()->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 =
|
|
rtp_manager()->FindSenderInfo(*current_senders, stream_id, sender_id);
|
|
if (!sender_info) {
|
|
current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
|
|
rtp_manager()->OnLocalSenderAdded(current_senders->back(), media_type);
|
|
}
|
|
}
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::UpdateRemoteSendersList(
|
|
const cricket::StreamParamsVec& streams,
|
|
bool default_sender_needed,
|
|
cricket::MediaType media_type,
|
|
StreamCollection* new_streams) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(!IsUnifiedPlan());
|
|
|
|
std::vector<RtpSenderInfo>* current_senders =
|
|
rtp_manager()->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 {
|
|
rtp_manager()->OnRemoteSenderRemoved(
|
|
info, remote_streams_->find(info.stream_id), 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 =
|
|
rtp_manager()->FindSenderInfo(*current_senders, stream_id, sender_id);
|
|
if (!sender_info) {
|
|
current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
|
|
rtp_manager()->OnRemoteSenderAdded(current_senders->back(), stream,
|
|
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 = rtp_manager()->FindSenderInfo(
|
|
*current_senders, kDefaultStreamId, default_sender_id);
|
|
if (!default_sender_info) {
|
|
current_senders->push_back(
|
|
RtpSenderInfo(kDefaultStreamId, default_sender_id, /*ssrc=*/0));
|
|
rtp_manager()->OnRemoteSenderAdded(current_senders->back(),
|
|
default_stream, media_type);
|
|
}
|
|
}
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::EnableSending() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
for (const auto& transceiver : transceivers()->ListInternal()) {
|
|
cricket::ChannelInterface* channel = transceiver->channel();
|
|
if (channel) {
|
|
channel->Enable(true);
|
|
}
|
|
}
|
|
}
|
|
|
|
RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
|
|
SdpType type,
|
|
cricket::ContentSource source,
|
|
const std::map<std::string, const cricket::ContentGroup*>&
|
|
bundle_groups_by_mid) {
|
|
const SessionDescriptionInterface* sdesc =
|
|
(source == cricket::CS_LOCAL ? local_description()
|
|
: remote_description());
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(sdesc);
|
|
|
|
if (!UpdatePayloadTypeDemuxingState(source, bundle_groups_by_mid)) {
|
|
// Note that this is never expected to fail, since RtpDemuxer doesn't return
|
|
// an error when changing payload type demux criteria, which is all this
|
|
// does.
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Failed to update payload type demuxing state.");
|
|
}
|
|
|
|
// Push down the new SDP media section for each audio/video transceiver.
|
|
auto rtp_transceivers = transceivers()->ListInternal();
|
|
std::vector<
|
|
std::pair<cricket::ChannelInterface*, const MediaContentDescription*>>
|
|
channels;
|
|
for (const auto& transceiver : rtp_transceivers) {
|
|
const ContentInfo* content_info =
|
|
FindMediaSectionForTransceiver(transceiver, sdesc);
|
|
cricket::ChannelInterface* channel = transceiver->channel();
|
|
if (!channel || !content_info || content_info->rejected) {
|
|
continue;
|
|
}
|
|
const MediaContentDescription* content_desc =
|
|
content_info->media_description();
|
|
if (!content_desc) {
|
|
continue;
|
|
}
|
|
|
|
transceiver->OnNegotiationUpdate(type, content_desc);
|
|
channels.push_back(std::make_pair(channel, content_desc));
|
|
}
|
|
|
|
if (!channels.empty()) {
|
|
RTCError error =
|
|
pc_->worker_thread()->Invoke<RTCError>(RTC_FROM_HERE, [&]() {
|
|
std::string error;
|
|
for (const auto& entry : channels) {
|
|
bool success =
|
|
(source == cricket::CS_LOCAL)
|
|
? entry.first->SetLocalContent(entry.second, type, &error)
|
|
: entry.first->SetRemoteContent(entry.second, type, &error);
|
|
if (!success) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, error);
|
|
}
|
|
}
|
|
return RTCError::OK();
|
|
});
|
|
if (!error.ok()) {
|
|
return 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 (pc_->sctp_mid() && local_description() && remote_description()) {
|
|
auto local_sctp_description = cricket::GetFirstSctpDataContentDescription(
|
|
local_description()->description());
|
|
auto remote_sctp_description = cricket::GetFirstSctpDataContentDescription(
|
|
remote_description()->description());
|
|
if (local_sctp_description && remote_sctp_description) {
|
|
int max_message_size;
|
|
// A remote max message size of zero means "any size supported".
|
|
// We configure the connection with our own max message size.
|
|
if (remote_sctp_description->max_message_size() == 0) {
|
|
max_message_size = local_sctp_description->max_message_size();
|
|
} else {
|
|
max_message_size =
|
|
std::min(local_sctp_description->max_message_size(),
|
|
remote_sctp_description->max_message_size());
|
|
}
|
|
pc_->StartSctpTransport(local_sctp_description->port(),
|
|
remote_sctp_description->port(),
|
|
max_message_size);
|
|
}
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
RTCError SdpOfferAnswerHandler::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());
|
|
}
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::RemoveStoppedTransceivers() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// 3.2.10.1: For each transceiver in the connection's set of transceivers
|
|
// run the following steps:
|
|
if (!IsUnifiedPlan())
|
|
return;
|
|
// Traverse a copy of the transceiver list.
|
|
auto transceiver_list = transceivers()->List();
|
|
for (auto transceiver : transceiver_list) {
|
|
// 3.2.10.1.1: If transceiver is stopped, associated with an m= section
|
|
// and the associated m= section is rejected in
|
|
// connection.[[CurrentLocalDescription]] or
|
|
// connection.[[CurrentRemoteDescription]], remove the
|
|
// transceiver from the connection's set of transceivers.
|
|
if (!transceiver->stopped()) {
|
|
continue;
|
|
}
|
|
const ContentInfo* local_content = FindMediaSectionForTransceiver(
|
|
transceiver->internal(), local_description());
|
|
const ContentInfo* remote_content = FindMediaSectionForTransceiver(
|
|
transceiver->internal(), remote_description());
|
|
if ((local_content && local_content->rejected) ||
|
|
(remote_content && remote_content->rejected)) {
|
|
RTC_LOG(LS_INFO) << "Dissociating transceiver"
|
|
" since the media section is being recycled.";
|
|
transceiver->internal()->set_mid(absl::nullopt);
|
|
transceiver->internal()->set_mline_index(absl::nullopt);
|
|
} else if (!local_content && !remote_content) {
|
|
// TODO(bugs.webrtc.org/11973): Consider if this should be removed already
|
|
// See https://github.com/w3c/webrtc-pc/issues/2576
|
|
RTC_LOG(LS_INFO)
|
|
<< "Dropping stopped transceiver that was never associated";
|
|
}
|
|
transceivers()->Remove(transceiver);
|
|
}
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::RemoveUnusedChannels(
|
|
const SessionDescription* desc) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// 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(rtp_manager()->GetVideoTransceiver());
|
|
}
|
|
|
|
const cricket::ContentInfo* audio_info = cricket::GetFirstAudioContent(desc);
|
|
if (!audio_info || audio_info->rejected) {
|
|
DestroyTransceiverChannel(rtp_manager()->GetAudioTransceiver());
|
|
}
|
|
|
|
const cricket::ContentInfo* data_info = cricket::GetFirstDataContent(desc);
|
|
if (!data_info || data_info->rejected) {
|
|
DestroyDataChannelTransport();
|
|
}
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::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);
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::UpdateEndedRemoteMediaStreams() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
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);
|
|
pc_->Observer()->OnRemoveStream(std::move(stream));
|
|
}
|
|
}
|
|
|
|
bool SdpOfferAnswerHandler::UseCandidatesInSessionDescription(
|
|
const SessionDescriptionInterface* remote_desc) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
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 SdpOfferAnswerHandler::UseCandidate(
|
|
const IceCandidateInterface* candidate) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
|
|
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
|
|
|
|
RTCErrorOr<const cricket::ContentInfo*> result =
|
|
FindContentInfo(remote_description(), candidate);
|
|
if (!result.ok())
|
|
return false;
|
|
|
|
const cricket::Candidate& c = candidate->candidate();
|
|
RTCError error = cricket::VerifyCandidate(c);
|
|
if (!error.ok()) {
|
|
RTC_LOG(LS_WARNING) << "Invalid candidate: " << c.ToString();
|
|
return true;
|
|
}
|
|
|
|
pc_->AddRemoteCandidate(result.value()->name, c);
|
|
|
|
return true;
|
|
}
|
|
|
|
// 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 SdpOfferAnswerHandler::ReadyToUseRemoteCandidate(
|
|
const IceCandidateInterface* candidate,
|
|
const SessionDescriptionInterface* remote_desc,
|
|
bool* valid) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
*valid = true;
|
|
|
|
const SessionDescriptionInterface* current_remote_desc =
|
|
remote_desc ? remote_desc : remote_description();
|
|
|
|
if (!current_remote_desc) {
|
|
return false;
|
|
}
|
|
|
|
RTCErrorOr<const cricket::ContentInfo*> result =
|
|
FindContentInfo(current_remote_desc, candidate);
|
|
if (!result.ok()) {
|
|
RTC_LOG(LS_ERROR) << "ReadyToUseRemoteCandidate: Invalid candidate. "
|
|
<< result.error().message();
|
|
|
|
*valid = false;
|
|
return false;
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
RTCErrorOr<const cricket::ContentInfo*> SdpOfferAnswerHandler::FindContentInfo(
|
|
const SessionDescriptionInterface* description,
|
|
const IceCandidateInterface* candidate) {
|
|
if (!candidate->sdp_mid().empty()) {
|
|
auto& contents = description->description()->contents();
|
|
auto it = absl::c_find_if(
|
|
contents, [candidate](const cricket::ContentInfo& content_info) {
|
|
return content_info.mid() == candidate->sdp_mid();
|
|
});
|
|
if (it == contents.end()) {
|
|
return RTCError(
|
|
RTCErrorType::INVALID_PARAMETER,
|
|
"Mid " + candidate->sdp_mid() +
|
|
" specified but no media section with that mid found.");
|
|
} else {
|
|
return &*it;
|
|
}
|
|
} else if (candidate->sdp_mline_index() >= 0) {
|
|
size_t mediacontent_index =
|
|
static_cast<size_t>(candidate->sdp_mline_index());
|
|
size_t content_size = description->description()->contents().size();
|
|
if (mediacontent_index < content_size) {
|
|
return &description->description()->contents()[mediacontent_index];
|
|
} else {
|
|
return RTCError(RTCErrorType::INVALID_RANGE,
|
|
"Media line index (" +
|
|
rtc::ToString(candidate->sdp_mline_index()) +
|
|
") out of range (number of mlines: " +
|
|
rtc::ToString(content_size) + ").");
|
|
}
|
|
}
|
|
|
|
return RTCError(RTCErrorType::INVALID_PARAMETER,
|
|
"Neither sdp_mline_index nor sdp_mid specified.");
|
|
}
|
|
|
|
RTCError SdpOfferAnswerHandler::CreateChannels(const SessionDescription& desc) {
|
|
// Creating the media channels. Transports should already have been created
|
|
// at this point.
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(&desc);
|
|
if (voice && !voice->rejected &&
|
|
!rtp_manager()->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.");
|
|
}
|
|
rtp_manager()->GetAudioTransceiver()->internal()->SetChannel(voice_channel);
|
|
}
|
|
|
|
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(&desc);
|
|
if (video && !video->rejected &&
|
|
!rtp_manager()->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.");
|
|
}
|
|
rtp_manager()->GetVideoTransceiver()->internal()->SetChannel(video_channel);
|
|
}
|
|
|
|
const cricket::ContentInfo* data = cricket::GetFirstDataContent(&desc);
|
|
if (data && !data->rejected &&
|
|
!data_channel_controller()->data_channel_transport()) {
|
|
if (!CreateDataChannel(data->name)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Failed to create data channel.");
|
|
}
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
|
|
cricket::VoiceChannel* SdpOfferAnswerHandler::CreateVoiceChannel(
|
|
const std::string& mid) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (!channel_manager()->media_engine())
|
|
return nullptr;
|
|
|
|
RtpTransportInternal* rtp_transport = pc_->GetRtpTransport(mid);
|
|
|
|
// TODO(bugs.webrtc.org/11992): CreateVoiceChannel internally switches to the
|
|
// worker thread. We shouldn't be using the |call_ptr_| hack here but simply
|
|
// be on the worker thread and use |call_| (update upstream code).
|
|
return channel_manager()->CreateVoiceChannel(
|
|
pc_->call_ptr(), pc_->configuration()->media_config, rtp_transport,
|
|
signaling_thread(), mid, pc_->SrtpRequired(), pc_->GetCryptoOptions(),
|
|
&ssrc_generator_, audio_options());
|
|
}
|
|
|
|
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
|
|
cricket::VideoChannel* SdpOfferAnswerHandler::CreateVideoChannel(
|
|
const std::string& mid) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (!channel_manager()->media_engine())
|
|
return nullptr;
|
|
|
|
// NOTE: This involves a non-ideal hop (Invoke) over to the network thread.
|
|
RtpTransportInternal* rtp_transport = pc_->GetRtpTransport(mid);
|
|
|
|
// TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to the
|
|
// worker thread. We shouldn't be using the |call_ptr_| hack here but simply
|
|
// be on the worker thread and use |call_| (update upstream code).
|
|
return channel_manager()->CreateVideoChannel(
|
|
pc_->call_ptr(), pc_->configuration()->media_config, rtp_transport,
|
|
signaling_thread(), mid, pc_->SrtpRequired(), pc_->GetCryptoOptions(),
|
|
&ssrc_generator_, video_options(),
|
|
video_bitrate_allocator_factory_.get());
|
|
}
|
|
|
|
bool SdpOfferAnswerHandler::CreateDataChannel(const std::string& mid) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (!pc_->network_thread()->Invoke<bool>(RTC_FROM_HERE, [this, &mid] {
|
|
RTC_DCHECK_RUN_ON(pc_->network_thread());
|
|
return pc_->SetupDataChannelTransport_n(mid);
|
|
})) {
|
|
return false;
|
|
}
|
|
// TODO(tommi): Is this necessary? SetupDataChannelTransport_n() above
|
|
// will have queued up updating the transport name on the signaling thread
|
|
// and could update the mid at the same time. This here is synchronous
|
|
// though, but it changes the state of PeerConnection and makes it be
|
|
// out of sync (transport name not set while the mid is set).
|
|
pc_->SetSctpDataMid(mid);
|
|
return true;
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::DestroyTransceiverChannel(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver) {
|
|
RTC_DCHECK(transceiver);
|
|
RTC_LOG_THREAD_BLOCK_COUNT();
|
|
|
|
// TODO(tommi): We're currently on the signaling thread.
|
|
// There are multiple hops to the worker ahead.
|
|
// Consider if we can make the call to SetChannel() on the worker thread
|
|
// (and require that to be the context it's always called in) and also
|
|
// call DestroyChannelInterface there, since it also needs to hop to the
|
|
// worker.
|
|
|
|
cricket::ChannelInterface* channel = transceiver->internal()->channel();
|
|
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
|
|
if (channel) {
|
|
// TODO(tommi): VideoRtpReceiver::SetMediaChannel blocks and jumps to the
|
|
// worker thread. When being set to nullptr, there are additional
|
|
// blocking calls to e.g. ClearRecordableEncodedFrameCallback which triggers
|
|
// another blocking call or Stop() for video channels.
|
|
// The channel object also needs to be de-initialized on the network thread
|
|
// so if ownership of the channel object lies with the transceiver, we could
|
|
// un-set the channel pointer and uninitialize/destruct the channel object
|
|
// at the same time, rather than in separate steps.
|
|
transceiver->internal()->SetChannel(nullptr);
|
|
// TODO(tommi): All channel objects end up getting deleted on the
|
|
// worker thread (ideally should be on the network thread but the
|
|
// MediaChannel objects are tied to the worker. Can the teardown be done
|
|
// asynchronously across the threads rather than blocking?
|
|
DestroyChannelInterface(channel);
|
|
}
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::DestroyDataChannelTransport() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
const bool has_sctp = pc_->sctp_mid().has_value();
|
|
|
|
if (has_sctp)
|
|
data_channel_controller()->OnTransportChannelClosed();
|
|
|
|
pc_->network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
|
|
RTC_DCHECK_RUN_ON(pc_->network_thread());
|
|
pc_->TeardownDataChannelTransport_n();
|
|
});
|
|
|
|
if (has_sctp)
|
|
pc_->ResetSctpDataMid();
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::DestroyChannelInterface(
|
|
cricket::ChannelInterface* channel) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(channel_manager()->media_engine());
|
|
RTC_DCHECK(channel);
|
|
|
|
// TODO(bugs.webrtc.org/11992): All the below methods should be called on the
|
|
// worker thread. (they switch internally anyway). Change
|
|
// DestroyChannelInterface to either be called on the worker thread, or do
|
|
// this asynchronously on the worker.
|
|
RTC_LOG_THREAD_BLOCK_COUNT();
|
|
|
|
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:
|
|
RTC_NOTREACHED()
|
|
<< "Trying to destroy datachannel through DestroyChannelInterface";
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED() << "Unknown media type: " << channel->media_type();
|
|
break;
|
|
}
|
|
|
|
// TODO(tommi): Figure out why we can get 2 blocking calls when running
|
|
// PeerConnectionCryptoTest.CreateAnswerWithDifferentSslRoles.
|
|
// and 3 when running
|
|
// PeerConnectionCryptoTest.CreateAnswerWithDifferentSslRoles
|
|
// RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(1);
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::DestroyAllChannels() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (!transceivers()) {
|
|
return;
|
|
}
|
|
|
|
RTC_LOG_THREAD_BLOCK_COUNT();
|
|
|
|
// Destroy video channels first since they may have a pointer to a voice
|
|
// channel.
|
|
auto list = transceivers()->List();
|
|
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
|
|
|
|
for (const auto& transceiver : list) {
|
|
if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
|
|
DestroyTransceiverChannel(transceiver);
|
|
}
|
|
}
|
|
for (const auto& transceiver : list) {
|
|
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
|
DestroyTransceiverChannel(transceiver);
|
|
}
|
|
}
|
|
|
|
DestroyDataChannelTransport();
|
|
}
|
|
|
|
void SdpOfferAnswerHandler::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) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
for (const cricket::ContentInfo& content :
|
|
session_desc->description()->contents()) {
|
|
if (IsAudioContent(&content)) {
|
|
// If we already have an audio m= section, reject this extra one.
|
|
if (*audio_index) {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(
|
|
cricket::MEDIA_TYPE_AUDIO, content.name,
|
|
RtpTransceiverDirection::kInactive, /*stopped=*/true));
|
|
} else {
|
|
bool stopped = (audio_direction == RtpTransceiverDirection::kInactive);
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO,
|
|
content.name, audio_direction,
|
|
stopped));
|
|
*audio_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
session_options->media_description_options.back().header_extensions =
|
|
channel_manager()->GetSupportedAudioRtpHeaderExtensions();
|
|
} else if (IsVideoContent(&content)) {
|
|
// If we already have an video m= section, reject this extra one.
|
|
if (*video_index) {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(
|
|
cricket::MEDIA_TYPE_VIDEO, content.name,
|
|
RtpTransceiverDirection::kInactive, /*stopped=*/true));
|
|
} else {
|
|
bool stopped = (video_direction == RtpTransceiverDirection::kInactive);
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO,
|
|
content.name, video_direction,
|
|
stopped));
|
|
*video_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
session_options->media_description_options.back().header_extensions =
|
|
channel_manager()->GetSupportedVideoRtpHeaderExtensions();
|
|
} else if (IsUnsupportedContent(&content)) {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(cricket::MEDIA_TYPE_UNSUPPORTED,
|
|
content.name,
|
|
RtpTransceiverDirection::kInactive,
|
|
/*stopped=*/true));
|
|
} 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
|
|
SdpOfferAnswerHandler::GetMediaDescriptionOptionsForActiveData(
|
|
const std::string& mid) const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Direction for data sections is meaningless, but legacy endpoints might
|
|
// expect sendrecv.
|
|
cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
|
|
RtpTransceiverDirection::kSendRecv,
|
|
/*stopped=*/false);
|
|
return options;
|
|
}
|
|
|
|
cricket::MediaDescriptionOptions
|
|
SdpOfferAnswerHandler::GetMediaDescriptionOptionsForRejectedData(
|
|
const std::string& mid) const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
|
|
RtpTransceiverDirection::kInactive,
|
|
/*stopped=*/true);
|
|
return options;
|
|
}
|
|
|
|
bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState(
|
|
cricket::ContentSource source,
|
|
const std::map<std::string, const cricket::ContentGroup*>&
|
|
bundle_groups_by_mid) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// We may need to delete any created default streams and disable creation of
|
|
// new ones on the basis of payload type. This is needed to avoid SSRC
|
|
// collisions in Call's RtpDemuxer, in the case that a transceiver has
|
|
// created a default stream, and then some other channel gets the SSRC
|
|
// signaled in the corresponding Unified Plan "m=" section. Specifically, we
|
|
// need to disable payload type based demuxing when two bundled "m=" sections
|
|
// are using the same payload type(s). For more context
|
|
// see https://bugs.chromium.org/p/webrtc/issues/detail?id=11477
|
|
const SessionDescriptionInterface* sdesc =
|
|
(source == cricket::CS_LOCAL ? local_description()
|
|
: remote_description());
|
|
struct PayloadTypes {
|
|
std::set<int> audio_payload_types;
|
|
std::set<int> video_payload_types;
|
|
bool pt_demuxing_enabled_audio = true;
|
|
bool pt_demuxing_enabled_video = true;
|
|
};
|
|
std::map<const cricket::ContentGroup*, PayloadTypes> payload_types_by_bundle;
|
|
for (auto& content_info : sdesc->description()->contents()) {
|
|
auto it = bundle_groups_by_mid.find(content_info.name);
|
|
const cricket::ContentGroup* bundle_group =
|
|
it != bundle_groups_by_mid.end() ? it->second : nullptr;
|
|
// If this m= section isn't bundled, it's safe to demux by payload type
|
|
// since other m= sections using the same payload type will also be using
|
|
// different transports.
|
|
if (!bundle_group) {
|
|
continue;
|
|
}
|
|
PayloadTypes* payload_types = &payload_types_by_bundle[bundle_group];
|
|
if (content_info.rejected ||
|
|
(source == cricket::ContentSource::CS_LOCAL &&
|
|
!RtpTransceiverDirectionHasRecv(
|
|
content_info.media_description()->direction())) ||
|
|
(source == cricket::ContentSource::CS_REMOTE &&
|
|
!RtpTransceiverDirectionHasSend(
|
|
content_info.media_description()->direction()))) {
|
|
// Ignore transceivers that are not receiving.
|
|
continue;
|
|
}
|
|
switch (content_info.media_description()->type()) {
|
|
case cricket::MediaType::MEDIA_TYPE_AUDIO: {
|
|
const cricket::AudioContentDescription* audio_desc =
|
|
content_info.media_description()->as_audio();
|
|
for (const cricket::AudioCodec& audio : audio_desc->codecs()) {
|
|
if (payload_types->audio_payload_types.count(audio.id)) {
|
|
// Two m= sections are using the same payload type, thus demuxing
|
|
// by payload type is not possible.
|
|
payload_types->pt_demuxing_enabled_audio = false;
|
|
}
|
|
payload_types->audio_payload_types.insert(audio.id);
|
|
}
|
|
break;
|
|
}
|
|
case cricket::MediaType::MEDIA_TYPE_VIDEO: {
|
|
const cricket::VideoContentDescription* video_desc =
|
|
content_info.media_description()->as_video();
|
|
for (const cricket::VideoCodec& video : video_desc->codecs()) {
|
|
if (payload_types->video_payload_types.count(video.id)) {
|
|
// Two m= sections are using the same payload type, thus demuxing
|
|
// by payload type is not possible.
|
|
payload_types->pt_demuxing_enabled_video = false;
|
|
}
|
|
payload_types->video_payload_types.insert(video.id);
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
// Ignore data channels.
|
|
continue;
|
|
}
|
|
}
|
|
|
|
// Gather all updates ahead of time so that all channels can be updated in a
|
|
// single Invoke; necessary due to thread guards.
|
|
std::vector<std::pair<RtpTransceiverDirection, cricket::ChannelInterface*>>
|
|
channels_to_update;
|
|
for (const auto& transceiver : transceivers()->ListInternal()) {
|
|
cricket::ChannelInterface* channel = transceiver->channel();
|
|
const ContentInfo* content =
|
|
FindMediaSectionForTransceiver(transceiver, sdesc);
|
|
if (!channel || !content) {
|
|
continue;
|
|
}
|
|
RtpTransceiverDirection local_direction =
|
|
content->media_description()->direction();
|
|
if (source == cricket::CS_REMOTE) {
|
|
local_direction = RtpTransceiverDirectionReversed(local_direction);
|
|
}
|
|
channels_to_update.emplace_back(local_direction, transceiver->channel());
|
|
}
|
|
|
|
if (channels_to_update.empty()) {
|
|
return true;
|
|
}
|
|
return pc_->worker_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE,
|
|
[&channels_to_update, &bundle_groups_by_mid, &payload_types_by_bundle]() {
|
|
for (const auto& it : channels_to_update) {
|
|
RtpTransceiverDirection local_direction = it.first;
|
|
cricket::ChannelInterface* channel = it.second;
|
|
cricket::MediaType media_type = channel->media_type();
|
|
auto bundle_it = bundle_groups_by_mid.find(channel->content_name());
|
|
const cricket::ContentGroup* bundle_group =
|
|
bundle_it != bundle_groups_by_mid.end() ? bundle_it->second
|
|
: nullptr;
|
|
if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO) {
|
|
if (!channel->SetPayloadTypeDemuxingEnabled(
|
|
(!bundle_group || payload_types_by_bundle[bundle_group]
|
|
.pt_demuxing_enabled_audio) &&
|
|
RtpTransceiverDirectionHasRecv(local_direction))) {
|
|
return false;
|
|
}
|
|
} else if (media_type == cricket::MediaType::MEDIA_TYPE_VIDEO) {
|
|
if (!channel->SetPayloadTypeDemuxingEnabled(
|
|
(!bundle_group || payload_types_by_bundle[bundle_group]
|
|
.pt_demuxing_enabled_video) &&
|
|
RtpTransceiverDirectionHasRecv(local_direction))) {
|
|
return false;
|
|
}
|
|
}
|
|
}
|
|
return true;
|
|
});
|
|
}
|
|
|
|
} // namespace webrtc
|