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gerrit-public.fairphone.software / platform / external / webrtc / f9381f0e734a64a3a58f19312c3d26825616cc3d / . / pc / peerconnection.cc
blob: 49ab979e409221b38e76a060919fc519bdbca875 [file] [log] [blame]
/*
* Copyright 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/peerconnection.h"
#include <algorithm>
#include <set>
#include <utility>
#include <vector>
#include "api/jsepicecandidate.h"
#include "api/jsepsessiondescription.h"
#include "api/mediaconstraintsinterface.h"
#include "api/mediastreamproxy.h"
#include "api/mediastreamtrackproxy.h"
#include "call/call.h"
#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/sctp/sctptransport.h"
#include "pc/audiotrack.h"
#include "pc/channel.h"
#include "pc/channelmanager.h"
#include "pc/dtmfsender.h"
#include "pc/mediastream.h"
#include "pc/mediastreamobserver.h"
#include "pc/remoteaudiosource.h"
#include "pc/rtpmediautils.h"
#include "pc/rtpreceiver.h"
#include "pc/rtpsender.h"
#include "pc/sctputils.h"
#include "pc/sdputils.h"
#include "pc/streamcollection.h"
#include "pc/videocapturertracksource.h"
#include "pc/videotrack.h"
#include "rtc_base/bind.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/stringencode.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/field_trial.h"
using cricket::ContentInfo;
using cricket::ContentInfos;
using cricket::MediaContentDescription;
using cricket::SessionDescription;
using cricket::TransportInfo;
using cricket::LOCAL_PORT_TYPE;
using cricket::STUN_PORT_TYPE;
using cricket::RELAY_PORT_TYPE;
using cricket::PRFLX_PORT_TYPE;
namespace webrtc {
// Error messages
const char kBundleWithoutRtcpMux[] =
"rtcp-mux must be enabled when BUNDLE "
"is enabled.";
const char kInvalidCandidates[] = "Description contains invalid candidates.";
const char kInvalidSdp[] = "Invalid session description.";
const char kMlineMismatchInAnswer[] =
"The order of m-lines in answer doesn't match order in offer. Rejecting "
"answer.";
const char kMlineMismatchInSubsequentOffer[] =
"The order of m-lines in subsequent offer doesn't match order from "
"previous offer/answer.";
const char kSdpWithoutDtlsFingerprint[] =
"Called with SDP without DTLS fingerprint.";
const char kSdpWithoutSdesCrypto[] = "Called with SDP without SDES crypto.";
const char kSdpWithoutIceUfragPwd[] =
"Called with SDP without ice-ufrag and ice-pwd.";
const char kSessionError[] = "Session error code: ";
const char kSessionErrorDesc[] = "Session error description: ";
const char kDtlsSrtpSetupFailureRtp[] =
"Couldn't set up DTLS-SRTP on RTP channel.";
const char kDtlsSrtpSetupFailureRtcp[] =
"Couldn't set up DTLS-SRTP on RTCP channel.";
const char kEnableBundleFailed[] = "Failed to enable BUNDLE.";
namespace {
static const char kDefaultStreamLabel[] = "default";
static const char kDefaultAudioSenderId[] = "defaulta0";
static const char kDefaultVideoSenderId[] = "defaultv0";
// The length of RTCP CNAMEs.
static const int kRtcpCnameLength = 16;
enum {
MSG_SET_SESSIONDESCRIPTION_SUCCESS = 0,
MSG_SET_SESSIONDESCRIPTION_FAILED,
MSG_CREATE_SESSIONDESCRIPTION_FAILED,
MSG_GETSTATS,
MSG_FREE_DATACHANNELS,
};
struct SetSessionDescriptionMsg : public rtc::MessageData {
explicit SetSessionDescriptionMsg(
webrtc::SetSessionDescriptionObserver* observer)
: observer(observer) {
}
rtc::scoped_refptr<webrtc::SetSessionDescriptionObserver> observer;
std::string error;
};
struct CreateSessionDescriptionMsg : public rtc::MessageData {
explicit CreateSessionDescriptionMsg(
webrtc::CreateSessionDescriptionObserver* observer)
: observer(observer) {}
rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer;
std::string error;
};
struct GetStatsMsg : public rtc::MessageData {
GetStatsMsg(webrtc::StatsObserver* observer,
webrtc::MediaStreamTrackInterface* track)
: observer(observer), track(track) {
}
rtc::scoped_refptr<webrtc::StatsObserver> observer;
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track;
};
// Check if we can send |new_stream| on a PeerConnection.
bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
webrtc::MediaStreamInterface* new_stream) {
if (!new_stream || !current_streams) {
return false;
}
if (current_streams->find(new_stream->label()) != nullptr) {
RTC_LOG(LS_ERROR) << "MediaStream with label " << new_stream->label()
<< " is already added.";
return false;
}
return true;
}
// If the direction is "recvonly" or "inactive", treat the description
// as containing no streams.
// See: https://code.google.com/p/webrtc/issues/detail?id=5054
std::vector<cricket::StreamParams> GetActiveStreams(
const cricket::MediaContentDescription* desc) {
return RtpTransceiverDirectionHasSend(desc->direction())
? desc->streams()
: std::vector<cricket::StreamParams>();
}
bool IsValidOfferToReceiveMedia(int value) {
typedef PeerConnectionInterface::RTCOfferAnswerOptions Options;
return (value >= Options::kUndefined) &&
(value <= Options::kMaxOfferToReceiveMedia);
}
// Add options to |[audio/video]_media_description_options| from |senders|.
void AddRtpSenderOptions(
const std::vector<rtc::scoped_refptr<
RtpSenderProxyWithInternal<RtpSenderInternal>>>& senders,
cricket::MediaDescriptionOptions* audio_media_description_options,
cricket::MediaDescriptionOptions* video_media_description_options) {
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(), 1);
}
}
}
}
// Add options to |session_options| from |rtp_data_channels|.
void AddRtpDataChannelOptions(
const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
rtp_data_channels,
cricket::MediaDescriptionOptions* data_media_description_options) {
if (!data_media_description_options) {
return;
}
// Check for data channels.
for (const auto& kv : rtp_data_channels) {
const DataChannel* channel = kv.second;
if (channel->state() == DataChannel::kConnecting ||
channel->state() == DataChannel::kOpen) {
// Legacy RTP data channels are signaled with the track/stream ID set to
// the data channel's label.
data_media_description_options->AddRtpDataChannel(channel->label(),
channel->label());
}
}
}
uint32_t ConvertIceTransportTypeToCandidateFilter(
PeerConnectionInterface::IceTransportsType type) {
switch (type) {
case PeerConnectionInterface::kNone:
return cricket::CF_NONE;
case PeerConnectionInterface::kRelay:
return cricket::CF_RELAY;
case PeerConnectionInterface::kNoHost:
return (cricket::CF_ALL & ~cricket::CF_HOST);
case PeerConnectionInterface::kAll:
return cricket::CF_ALL;
default:
RTC_NOTREACHED();
}
return cricket::CF_NONE;
}
// Helper to set an error and return from a method.
bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
if (error) {
error->set_type(type);
}
return type == webrtc::RTCErrorType::NONE;
}
bool SafeSetError(webrtc::RTCError error, webrtc::RTCError* error_out) {
if (error_out) {
*error_out = std::move(error);
}
return error.ok();
}
std::string GetSignalingStateString(
PeerConnectionInterface::SignalingState state) {
switch (state) {
case PeerConnectionInterface::kStable:
return "kStable";
case PeerConnectionInterface::kHaveLocalOffer:
return "kHaveLocalOffer";
case PeerConnectionInterface::kHaveLocalPrAnswer:
return "kHavePrAnswer";
case PeerConnectionInterface::kHaveRemoteOffer:
return "kHaveRemoteOffer";
case PeerConnectionInterface::kHaveRemotePrAnswer:
return "kHaveRemotePrAnswer";
case PeerConnectionInterface::kClosed:
return "kClosed";
}
RTC_NOTREACHED();
return "";
}
IceCandidatePairType GetIceCandidatePairCounter(
const cricket::Candidate& local,
const cricket::Candidate& remote) {
const auto& l = local.type();
const auto& r = remote.type();
const auto& host = LOCAL_PORT_TYPE;
const auto& srflx = STUN_PORT_TYPE;
const auto& relay = RELAY_PORT_TYPE;
const auto& prflx = PRFLX_PORT_TYPE;
if (l == host && r == host) {
bool local_private = IPIsPrivate(local.address().ipaddr());
bool remote_private = IPIsPrivate(remote.address().ipaddr());
if (local_private) {
if (remote_private) {
return kIceCandidatePairHostPrivateHostPrivate;
} else {
return kIceCandidatePairHostPrivateHostPublic;
}
} else {
if (remote_private) {
return kIceCandidatePairHostPublicHostPrivate;
} else {
return kIceCandidatePairHostPublicHostPublic;
}
}
}
if (l == host && r == srflx)
return kIceCandidatePairHostSrflx;
if (l == host && r == relay)
return kIceCandidatePairHostRelay;
if (l == host && r == prflx)
return kIceCandidatePairHostPrflx;
if (l == srflx && r == host)
return kIceCandidatePairSrflxHost;
if (l == srflx && r == srflx)
return kIceCandidatePairSrflxSrflx;
if (l == srflx && r == relay)
return kIceCandidatePairSrflxRelay;
if (l == srflx && r == prflx)
return kIceCandidatePairSrflxPrflx;
if (l == relay && r == host)
return kIceCandidatePairRelayHost;
if (l == relay && r == srflx)
return kIceCandidatePairRelaySrflx;
if (l == relay && r == relay)
return kIceCandidatePairRelayRelay;
if (l == relay && r == prflx)
return kIceCandidatePairRelayPrflx;
if (l == prflx && r == host)
return kIceCandidatePairPrflxHost;
if (l == prflx && r == srflx)
return kIceCandidatePairPrflxSrflx;
if (l == prflx && r == relay)
return kIceCandidatePairPrflxRelay;
return kIceCandidatePairMax;
}
// Verify that the order of media sections in |new_desc| matches
// |existing_desc|. The number of m= sections in |new_desc| should be no less
// than |existing_desc|.
bool MediaSectionsInSameOrder(const SessionDescription* existing_desc,
const SessionDescription* new_desc) {
if (!existing_desc || !new_desc) {
return false;
}
if (existing_desc->contents().size() > new_desc->contents().size()) {
return false;
}
for (size_t i = 0; i < existing_desc->contents().size(); ++i) {
if (new_desc->contents()[i].name != existing_desc->contents()[i].name) {
return false;
}
const MediaContentDescription* new_desc_mdesc =
static_cast<const MediaContentDescription*>(
new_desc->contents()[i].description);
const MediaContentDescription* existing_desc_mdesc =
static_cast<const MediaContentDescription*>(
existing_desc->contents()[i].description);
if (new_desc_mdesc->type() != existing_desc_mdesc->type()) {
return false;
}
}
return true;
}
bool MediaSectionsHaveSameCount(const SessionDescription* desc1,
const SessionDescription* desc2) {
if (!desc1 || !desc2) {
return false;
}
return desc1->contents().size() == desc2->contents().size();
}
// Checks that each non-rejected content has SDES crypto keys or a DTLS
// fingerprint, unless it's in a BUNDLE group, in which case only the
// BUNDLE-tag section (first media section/description in the BUNDLE group)
// needs a ufrag and pwd. Mismatches, such as replying with a DTLS fingerprint
// to SDES keys, will be caught in JsepTransport negotiation, and backstopped
// by Channel's |srtp_required| check.
RTCError VerifyCrypto(const SessionDescription* desc, bool dtls_enabled) {
const cricket::ContentGroup* bundle =
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
for (const cricket::ContentInfo& content_info : desc->contents()) {
if (content_info.rejected) {
continue;
}
const std::string& mid = content_info.name;
if (bundle && bundle->HasContentName(mid) &&
mid != *(bundle->FirstContentName())) {
// This isn't the first media section in the BUNDLE group, so it's not
// required to have crypto attributes, since only the crypto attributes
// from the first section actually get used.
continue;
}
// If the content isn't rejected or bundled into another m= section, crypto
// must be present.
const MediaContentDescription* media =
static_cast<const MediaContentDescription*>(content_info.description);
const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
if (!media || !tinfo) {
// Something is not right.
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
}
if (dtls_enabled) {
if (!tinfo->description.identity_fingerprint) {
RTC_LOG(LS_WARNING)
<< "Session description must have DTLS fingerprint if "
"DTLS enabled.";
return RTCError(RTCErrorType::INVALID_PARAMETER,
kSdpWithoutDtlsFingerprint);
}
} else {
if (media->cryptos().empty()) {
RTC_LOG(LS_WARNING)
<< "Session description must have SDES when DTLS disabled.";
return RTCError(RTCErrorType::INVALID_PARAMETER, kSdpWithoutSdesCrypto);
}
}
}
return RTCError::OK();
}
// Checks that each non-rejected content has ice-ufrag and ice-pwd set, unless
// it's in a BUNDLE group, in which case only the BUNDLE-tag section (first
// media section/description in the BUNDLE group) needs a ufrag and pwd.
bool VerifyIceUfragPwdPresent(const SessionDescription* desc) {
const cricket::ContentGroup* bundle =
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
for (const cricket::ContentInfo& content_info : desc->contents()) {
if (content_info.rejected) {
continue;
}
const std::string& mid = content_info.name;
if (bundle && bundle->HasContentName(mid) &&
mid != *(bundle->FirstContentName())) {
// This isn't the first media section in the BUNDLE group, so it's not
// required to have ufrag/password, since only the ufrag/password from
// the first section actually get used.
continue;
}
// If the content isn't rejected or bundled into another m= section,
// ice-ufrag and ice-pwd must be present.
const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
if (!tinfo) {
// Something is not right.
RTC_LOG(LS_ERROR) << kInvalidSdp;
return false;
}
if (tinfo->description.ice_ufrag.empty() ||
tinfo->description.ice_pwd.empty()) {
RTC_LOG(LS_ERROR) << "Session description must have ice ufrag and pwd.";
return false;
}
}
return true;
}
bool GetTrackIdBySsrc(const SessionDescription* session_description,
uint32_t ssrc,
std::string* track_id) {
RTC_DCHECK(track_id != NULL);
const cricket::ContentInfo* audio_info =
cricket::GetFirstAudioContent(session_description);
if (audio_info) {
const cricket::MediaContentDescription* audio_content =
static_cast<const cricket::MediaContentDescription*>(
audio_info->description);
const auto* found =
cricket::GetStreamBySsrc(audio_content->streams(), ssrc);
if (found) {
*track_id = found->id;
return true;
}
}
const cricket::ContentInfo* video_info =
cricket::GetFirstVideoContent(session_description);
if (video_info) {
const cricket::MediaContentDescription* video_content =
static_cast<const cricket::MediaContentDescription*>(
video_info->description);
const auto* found =
cricket::GetStreamBySsrc(video_content->streams(), ssrc);
if (found) {
*track_id = found->id;
return true;
}
}
return false;
}
// Get the SCTP port out of a SessionDescription.
// Return -1 if not found.
int GetSctpPort(const SessionDescription* session_description) {
const ContentInfo* content_info = GetFirstDataContent(session_description);
RTC_DCHECK(content_info);
if (!content_info) {
return -1;
}
const cricket::DataContentDescription* data =
static_cast<const cricket::DataContentDescription*>(
(content_info->description));
std::string value;
cricket::DataCodec match_pattern(cricket::kGoogleSctpDataCodecPlType,
cricket::kGoogleSctpDataCodecName);
for (const cricket::DataCodec& codec : data->codecs()) {
if (!codec.Matches(match_pattern)) {
continue;
}
if (codec.GetParam(cricket::kCodecParamPort, &value)) {
return rtc::FromString<int>(value);
}
}
return -1;
}
// Returns true if |new_desc| requests an ICE restart (i.e., new ufrag/pwd).
bool CheckForRemoteIceRestart(const SessionDescriptionInterface* old_desc,
const SessionDescriptionInterface* new_desc,
const std::string& content_name) {
if (!old_desc) {
return false;
}
const SessionDescription* new_sd = new_desc->description();
const SessionDescription* old_sd = old_desc->description();
const ContentInfo* cinfo = new_sd->GetContentByName(content_name);
if (!cinfo || cinfo->rejected) {
return false;
}
// If the content isn't rejected, check if ufrag and password has changed.
const cricket::TransportDescription* new_transport_desc =
new_sd->GetTransportDescriptionByName(content_name);
const cricket::TransportDescription* old_transport_desc =
old_sd->GetTransportDescriptionByName(content_name);
if (!new_transport_desc || !old_transport_desc) {
// No transport description exists. This is not an ICE restart.
return false;
}
if (cricket::IceCredentialsChanged(
old_transport_desc->ice_ufrag, old_transport_desc->ice_pwd,
new_transport_desc->ice_ufrag, new_transport_desc->ice_pwd)) {
RTC_LOG(LS_INFO) << "Remote peer requests ICE restart for " << content_name
<< ".";
return true;
}
return false;
}
} // namespace
// Upon completion, posts a task to execute the callback of the
// SetSessionDescriptionObserver asynchronously on the same thread. At this
// point, the state of the peer connection might no longer reflect the effects
// of the SetRemoteDescription operation, as the peer connection could have been
// modified during the post.
// TODO(hbos): Remove this class once we remove the version of
// PeerConnectionInterface::SetRemoteDescription() that takes a
// SetSessionDescriptionObserver as an argument.
class PeerConnection::SetRemoteDescriptionObserverAdapter
: public rtc::RefCountedObject<SetRemoteDescriptionObserverInterface> {
public:
SetRemoteDescriptionObserverAdapter(
rtc::scoped_refptr<PeerConnection> pc,
rtc::scoped_refptr<SetSessionDescriptionObserver> wrapper)
: pc_(std::move(pc)), wrapper_(std::move(wrapper)) {}
// SetRemoteDescriptionObserverInterface implementation.
void OnSetRemoteDescriptionComplete(RTCError error) override {
if (error.ok())
pc_->PostSetSessionDescriptionSuccess(wrapper_);
else
pc_->PostSetSessionDescriptionFailure(wrapper_, error.message());
}
private:
rtc::scoped_refptr<PeerConnection> pc_;
rtc::scoped_refptr<SetSessionDescriptionObserver> wrapper_;
};
bool PeerConnectionInterface::RTCConfiguration::operator==(
const PeerConnectionInterface::RTCConfiguration& o) const {
// This static_assert prevents us from accidentally breaking operator==.
// Note: Order matters! Fields must be ordered the same as RTCConfiguration.
struct stuff_being_tested_for_equality {
IceServers servers;
IceTransportsType type;
BundlePolicy bundle_policy;
RtcpMuxPolicy rtcp_mux_policy;
std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
int ice_candidate_pool_size;
bool disable_ipv6;
bool disable_ipv6_on_wifi;
int max_ipv6_networks;
bool enable_rtp_data_channel;
rtc::Optional<int> screencast_min_bitrate;
rtc::Optional<bool> combined_audio_video_bwe;
rtc::Optional<bool> enable_dtls_srtp;
TcpCandidatePolicy tcp_candidate_policy;
CandidateNetworkPolicy candidate_network_policy;
int audio_jitter_buffer_max_packets;
bool audio_jitter_buffer_fast_accelerate;
int ice_connection_receiving_timeout;
int ice_backup_candidate_pair_ping_interval;
ContinualGatheringPolicy continual_gathering_policy;
bool prioritize_most_likely_ice_candidate_pairs;
struct cricket::MediaConfig media_config;
bool prune_turn_ports;
bool presume_writable_when_fully_relayed;
bool enable_ice_renomination;
bool redetermine_role_on_ice_restart;
rtc::Optional<int> ice_check_min_interval;
rtc::Optional<rtc::IntervalRange> ice_regather_interval_range;
webrtc::TurnCustomizer* turn_customizer;
SdpSemantics sdp_semantics;
};
static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
"Did you add something to RTCConfiguration and forget to "
"update operator==?");
return type == o.type && servers == o.servers &&
bundle_policy == o.bundle_policy &&
rtcp_mux_policy == o.rtcp_mux_policy &&
tcp_candidate_policy == o.tcp_candidate_policy &&
candidate_network_policy == o.candidate_network_policy &&
audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
audio_jitter_buffer_fast_accelerate ==
o.audio_jitter_buffer_fast_accelerate &&
ice_connection_receiving_timeout ==
o.ice_connection_receiving_timeout &&
ice_backup_candidate_pair_ping_interval ==
o.ice_backup_candidate_pair_ping_interval &&
continual_gathering_policy == o.continual_gathering_policy &&
certificates == o.certificates &&
prioritize_most_likely_ice_candidate_pairs ==
o.prioritize_most_likely_ice_candidate_pairs &&
media_config == o.media_config && disable_ipv6 == o.disable_ipv6 &&
disable_ipv6_on_wifi == o.disable_ipv6_on_wifi &&
max_ipv6_networks == o.max_ipv6_networks &&
enable_rtp_data_channel == o.enable_rtp_data_channel &&
screencast_min_bitrate == o.screencast_min_bitrate &&
combined_audio_video_bwe == o.combined_audio_video_bwe &&
enable_dtls_srtp == o.enable_dtls_srtp &&
ice_candidate_pool_size == o.ice_candidate_pool_size &&
prune_turn_ports == o.prune_turn_ports &&
presume_writable_when_fully_relayed ==
o.presume_writable_when_fully_relayed &&
enable_ice_renomination == o.enable_ice_renomination &&
redetermine_role_on_ice_restart == o.redetermine_role_on_ice_restart &&
ice_check_min_interval == o.ice_check_min_interval &&
ice_regather_interval_range == o.ice_regather_interval_range &&
turn_customizer == o.turn_customizer &&
sdp_semantics == o.sdp_semantics;
}
bool PeerConnectionInterface::RTCConfiguration::operator!=(
const PeerConnectionInterface::RTCConfiguration& o) const {
return !(*this == o);
}
// Generate a RTCP CNAME when a PeerConnection is created.
std::string GenerateRtcpCname() {
std::string cname;
if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
RTC_LOG(LS_ERROR) << "Failed to generate CNAME.";
RTC_NOTREACHED();
}
return cname;
}
bool ValidateOfferAnswerOptions(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options) {
return IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_audio) &&
IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_video);
}
// From |rtc_options|, fill parts of |session_options| shared by all generated
// m= sections (in other words, nothing that involves a map/array).
void ExtractSharedMediaSessionOptions(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options) {
session_options->vad_enabled = rtc_options.voice_activity_detection;
session_options->bundle_enabled = rtc_options.use_rtp_mux;
}
bool ConvertConstraintsToOfferAnswerOptions(
const MediaConstraintsInterface* constraints,
PeerConnectionInterface::RTCOfferAnswerOptions* offer_answer_options) {
if (!constraints) {
return true;
}
bool value = false;
size_t mandatory_constraints_satisfied = 0;
if (FindConstraint(constraints,
MediaConstraintsInterface::kOfferToReceiveAudio, &value,
&mandatory_constraints_satisfied)) {
offer_answer_options->offer_to_receive_audio =
value ? PeerConnectionInterface::RTCOfferAnswerOptions::
kOfferToReceiveMediaTrue
: 0;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kOfferToReceiveVideo, &value,
&mandatory_constraints_satisfied)) {
offer_answer_options->offer_to_receive_video =
value ? PeerConnectionInterface::RTCOfferAnswerOptions::
kOfferToReceiveMediaTrue
: 0;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kVoiceActivityDetection, &value,
&mandatory_constraints_satisfied)) {
offer_answer_options->voice_activity_detection = value;
}
if (FindConstraint(constraints, MediaConstraintsInterface::kUseRtpMux, &value,
&mandatory_constraints_satisfied)) {
offer_answer_options->use_rtp_mux = value;
}
if (FindConstraint(constraints, MediaConstraintsInterface::kIceRestart,
&value, &mandatory_constraints_satisfied)) {
offer_answer_options->ice_restart = value;
}
return mandatory_constraints_satisfied == constraints->GetMandatory().size();
}
PeerConnection::PeerConnection(PeerConnectionFactory* factory,
std::unique_ptr<RtcEventLog> event_log,
std::unique_ptr<Call> call)
: factory_(factory),
event_log_(std::move(event_log)),
rtcp_cname_(GenerateRtcpCname()),
local_streams_(StreamCollection::Create()),
remote_streams_(StreamCollection::Create()),
call_(std::move(call)) {}
PeerConnection::~PeerConnection() {
TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
RTC_DCHECK_RUN_ON(signaling_thread());
StopAndDestroyChannels();
// Destroy stats after stopping all transceivers because the senders/receivers
// will update the stats collector before stopping.
stats_.reset(nullptr);
if (stats_collector_) {
stats_collector_->WaitForPendingRequest();
stats_collector_ = nullptr;
}
RTC_LOG(LS_INFO) << "Session: " << session_id() << " is destroyed.";
webrtc_session_desc_factory_.reset();
sctp_invoker_.reset();
sctp_factory_.reset();
transport_controller_.reset();
// port_allocator_ lives on the network thread and should be destroyed there.
network_thread()->Invoke<void>(RTC_FROM_HERE,
[this] { port_allocator_.reset(); });
// call_ and event_log_ must be destroyed on the worker thread.
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
call_.reset();
event_log_.reset();
});
}
void PeerConnection::StopAndDestroyChannels() {
for (auto transceiver : transceivers_) {
transceiver->Stop();
}
// Destroy video channels first since they may have a pointer to a voice
// channel.
for (auto transceiver : transceivers_) {
if (transceiver->internal()->media_type() == cricket::MEDIA_TYPE_VIDEO) {
DestroyTransceiverChannel(transceiver);
}
}
for (auto transceiver : transceivers_) {
if (transceiver->internal()->media_type() == cricket::MEDIA_TYPE_AUDIO) {
DestroyTransceiverChannel(transceiver);
}
}
DestroyDataChannel();
}
bool PeerConnection::Initialize(
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer) {
TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
RTCError config_error = ValidateConfiguration(configuration);
if (!config_error.ok()) {
RTC_LOG(LS_ERROR) << "Invalid configuration: " << config_error.message();
return false;
}
if (!allocator) {
RTC_LOG(LS_ERROR)
<< "PeerConnection initialized without a PortAllocator? "
<< "This shouldn't happen if using PeerConnectionFactory.";
return false;
}
if (!observer) {
// TODO(deadbeef): Why do we do this?
RTC_LOG(LS_ERROR) << "PeerConnection initialized without a "
<< "PeerConnectionObserver";
return false;
}
observer_ = observer;
port_allocator_ = std::move(allocator);
// The port allocator lives on the network thread and should be initialized
// there.
if (!network_thread()->Invoke<bool>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::InitializePortAllocator_n,
this, configuration))) {
return false;
}
// RFC 3264: The numeric value of the session id and version in the
// o line MUST be representable with a "64 bit signed integer".
// Due to this constraint session id |session_id_| is max limited to
// LLONG_MAX.
session_id_ = rtc::ToString(rtc::CreateRandomId64() & LLONG_MAX);
transport_controller_.reset(factory_->CreateTransportController(
port_allocator_.get(), configuration.redetermine_role_on_ice_restart));
transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLED);
transport_controller_->SignalConnectionState.connect(
this, &PeerConnection::OnTransportControllerConnectionState);
transport_controller_->SignalGatheringState.connect(
this, &PeerConnection::OnTransportControllerGatheringState);
transport_controller_->SignalCandidatesGathered.connect(
this, &PeerConnection::OnTransportControllerCandidatesGathered);
transport_controller_->SignalCandidatesRemoved.connect(
this, &PeerConnection::OnTransportControllerCandidatesRemoved);
transport_controller_->SignalDtlsHandshakeError.connect(
this, &PeerConnection::OnTransportControllerDtlsHandshakeError);
sctp_factory_ = factory_->CreateSctpTransportInternalFactory();
stats_.reset(new StatsCollector(this));
stats_collector_ = RTCStatsCollector::Create(this);
configuration_ = configuration;
const PeerConnectionFactoryInterface::Options& options = factory_->options();
transport_controller_->SetSslMaxProtocolVersion(options.ssl_max_version);
// Obtain a certificate from RTCConfiguration if any were provided (optional).
rtc::scoped_refptr<rtc::RTCCertificate> certificate;
if (!configuration.certificates.empty()) {
// TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of
// just picking the first one. The decision should be made based on the DTLS
// handshake. The DTLS negotiations need to know about all certificates.
certificate = configuration.certificates[0];
}
transport_controller_->SetIceConfig(ParseIceConfig(configuration));
if (options.disable_encryption) {
dtls_enabled_ = false;
} else {
// Enable DTLS by default if we have an identity store or a certificate.
dtls_enabled_ = (cert_generator || certificate);
// |configuration| can override the default |dtls_enabled_| value.
if (configuration.enable_dtls_srtp) {
dtls_enabled_ = *(configuration.enable_dtls_srtp);
}
}
// Enable creation of RTP data channels if the kEnableRtpDataChannels is set.
// It takes precendence over the disable_sctp_data_channels
// PeerConnectionFactoryInterface::Options.
if (configuration.enable_rtp_data_channel) {
data_channel_type_ = cricket::DCT_RTP;
} else {
// DTLS has to be enabled to use SCTP.
if (!options.disable_sctp_data_channels && dtls_enabled_) {
data_channel_type_ = cricket::DCT_SCTP;
}
}
video_options_.screencast_min_bitrate_kbps =
configuration.screencast_min_bitrate;
audio_options_.combined_audio_video_bwe =
configuration.combined_audio_video_bwe;
audio_options_.audio_jitter_buffer_max_packets =
configuration.audio_jitter_buffer_max_packets;
audio_options_.audio_jitter_buffer_fast_accelerate =
configuration.audio_jitter_buffer_fast_accelerate;
// Whether the certificate generator/certificate is null or not determines
// what PeerConnectionDescriptionFactory will do, so make sure that we give it
// the right instructions by clearing the variables if needed.
if (!dtls_enabled_) {
cert_generator.reset();
certificate = nullptr;
} else if (certificate) {
// Favor generated certificate over the certificate generator.
cert_generator.reset();
}
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
signaling_thread(), channel_manager(), this, session_id(),
std::move(cert_generator), certificate));
webrtc_session_desc_factory_->SignalCertificateReady.connect(
this, &PeerConnection::OnCertificateReady);
if (options.disable_encryption) {
webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
}
webrtc_session_desc_factory_->set_enable_encrypted_rtp_header_extensions(
options.crypto_options.enable_encrypted_rtp_header_extensions);
// Add default audio/video transceivers for Plan B SDP.
if (!IsUnifiedPlan()) {
transceivers_.push_back(
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_AUDIO)));
transceivers_.push_back(
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_VIDEO)));
}
return true;
}
RTCError PeerConnection::ValidateConfiguration(
const RTCConfiguration& config) const {
if (config.ice_regather_interval_range &&
config.continual_gathering_policy == GATHER_ONCE) {
return RTCError(RTCErrorType::INVALID_PARAMETER,
"ice_regather_interval_range specified but continual "
"gathering policy is GATHER_ONCE");
}
return RTCError::OK();
}
rtc::scoped_refptr<StreamCollectionInterface>
PeerConnection::local_streams() {
return local_streams_;
}
rtc::scoped_refptr<StreamCollectionInterface>
PeerConnection::remote_streams() {
return remote_streams_;
}
bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
TRACE_EVENT0("webrtc", "PeerConnection::AddStream");
if (IsClosed()) {
return false;
}
if (!CanAddLocalMediaStream(local_streams_, local_stream)) {
return false;
}
local_streams_->AddStream(local_stream);
MediaStreamObserver* observer = new MediaStreamObserver(local_stream);
observer->SignalAudioTrackAdded.connect(this,
&PeerConnection::OnAudioTrackAdded);
observer->SignalAudioTrackRemoved.connect(
this, &PeerConnection::OnAudioTrackRemoved);
observer->SignalVideoTrackAdded.connect(this,
&PeerConnection::OnVideoTrackAdded);
observer->SignalVideoTrackRemoved.connect(
this, &PeerConnection::OnVideoTrackRemoved);
stream_observers_.push_back(std::unique_ptr<MediaStreamObserver>(observer));
for (const auto& track : local_stream->GetAudioTracks()) {
AddAudioTrack(track.get(), local_stream);
}
for (const auto& track : local_stream->GetVideoTracks()) {
AddVideoTrack(track.get(), local_stream);
}
stats_->AddStream(local_stream);
observer_->OnRenegotiationNeeded();
return true;
}
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
if (!IsClosed()) {
for (const auto& track : local_stream->GetAudioTracks()) {
RemoveAudioTrack(track.get(), local_stream);
}
for (const auto& track : local_stream->GetVideoTracks()) {
RemoveVideoTrack(track.get(), local_stream);
}
}
local_streams_->RemoveStream(local_stream);
stream_observers_.erase(
std::remove_if(
stream_observers_.begin(), stream_observers_.end(),
[local_stream](const std::unique_ptr<MediaStreamObserver>& observer) {
return observer->stream()->label().compare(local_stream->label()) ==
0;
}),
stream_observers_.end());
if (IsClosed()) {
return;
}
observer_->OnRenegotiationNeeded();
}
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::AddTrack(
MediaStreamTrackInterface* track,
std::vector<MediaStreamInterface*> streams) {
TRACE_EVENT0("webrtc", "PeerConnection::AddTrack");
std::vector<std::string> stream_labels;
for (auto* stream : streams) {
if (!stream) {
RTC_LOG(LS_ERROR) << "Stream list has null element.";
return nullptr;
}
stream_labels.push_back(stream->label());
}
auto sender_or_error = AddTrackWithStreamLabels(track, stream_labels);
if (!sender_or_error.ok()) {
return nullptr;
}
return sender_or_error.MoveValue();
}
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
PeerConnection::AddTrackWithStreamLabels(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_labels) {
TRACE_EVENT0("webrtc", "PeerConnection::AddTrackWithStreamLabels");
if (!track) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Track is null.");
}
if (!(track->kind() == MediaStreamTrackInterface::kAudioKind ||
track->kind() == MediaStreamTrackInterface::kVideoKind)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Track has invalid kind: " + track->kind());
}
// TODO(bugs.webrtc.org/7932): Support adding a track to multiple streams.
if (stream_labels.size() > 1u) {
LOG_AND_RETURN_ERROR(
RTCErrorType::UNSUPPORTED_OPERATION,
"AddTrack with more than one stream is not currently supported.");
}
if (IsClosed()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"PeerConnection is closed.");
}
if (FindSenderForTrack(track)) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Sender already exists for track " + track->id() + ".");
}
// TODO(bugs.webrtc.org/7933): MediaSession expects the sender to have exactly
// one stream. AddTrackInternal will return an error if there is more than one
// stream, but if the caller specifies none then we need to generate a random
// stream label.
std::vector<std::string> adjusted_stream_labels = stream_labels;
if (stream_labels.empty()) {
adjusted_stream_labels.push_back(rtc::CreateRandomUuid());
}
RTC_DCHECK_EQ(1, adjusted_stream_labels.size());
auto sender_or_error =
(IsUnifiedPlan() ? AddTrackUnifiedPlan(track, adjusted_stream_labels)
: AddTrackPlanB(track, adjusted_stream_labels));
if (sender_or_error.ok()) {
observer_->OnRenegotiationNeeded();
}
return sender_or_error;
}
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
PeerConnection::AddTrackPlanB(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_labels) {
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
auto new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(),
new AudioRtpSender(static_cast<AudioTrackInterface*>(track.get()),
voice_channel(), stats_.get()));
GetAudioTransceiver()->internal()->AddSender(new_sender);
new_sender->internal()->set_stream_ids(stream_labels);
const RtpSenderInfo* sender_info =
FindSenderInfo(local_audio_sender_infos_,
new_sender->internal()->stream_id(), track->id());
if (sender_info) {
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
}
return rtc::scoped_refptr<RtpSenderInterface>(new_sender);
} else {
RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
auto new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(),
new VideoRtpSender(static_cast<VideoTrackInterface*>(track.get()),
video_channel()));
GetVideoTransceiver()->internal()->AddSender(new_sender);
new_sender->internal()->set_stream_ids(stream_labels);
const RtpSenderInfo* sender_info =
FindSenderInfo(local_video_sender_infos_,
new_sender->internal()->stream_id(), track->id());
if (sender_info) {
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
}
return rtc::scoped_refptr<RtpSenderInterface>(new_sender);
}
}
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
PeerConnection::AddTrackUnifiedPlan(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_labels) {
auto transceiver = FindFirstTransceiverForAddedTrack(track);
if (transceiver) {
if (transceiver->direction() == RtpTransceiverDirection::kRecvOnly) {
transceiver->SetDirection(RtpTransceiverDirection::kSendRecv);
} else if (transceiver->direction() == RtpTransceiverDirection::kInactive) {
transceiver->SetDirection(RtpTransceiverDirection::kSendOnly);
}
} else {
cricket::MediaType media_type =
(track->kind() == MediaStreamTrackInterface::kAudioKind
? cricket::MEDIA_TYPE_AUDIO
: cricket::MEDIA_TYPE_VIDEO);
transceiver = CreateTransceiver(media_type);
transceiver->internal()->set_created_by_addtrack(true);
transceiver->SetDirection(RtpTransceiverDirection::kSendRecv);
}
transceiver->sender()->SetTrack(track);
transceiver->internal()->sender_internal()->set_stream_ids(stream_labels);
return transceiver->sender();
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::FindFirstTransceiverForAddedTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
RTC_DCHECK(track);
for (auto transceiver : transceivers_) {
if (!transceiver->sender()->track() &&
cricket::MediaTypeToString(transceiver->internal()->media_type()) ==
track->kind() &&
!transceiver->internal()->has_ever_been_used_to_send()) {
return transceiver;
}
}
return nullptr;
}
bool PeerConnection::RemoveTrack(RtpSenderInterface* sender) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveTrack");
return RemoveTrackInternal(sender).ok();
}
RTCError PeerConnection::RemoveTrackInternal(
rtc::scoped_refptr<RtpSenderInterface> sender) {
if (!sender) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Sender is null.");
}
if (IsClosed()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"PeerConnection is closed.");
}
if (IsUnifiedPlan()) {
auto transceiver = FindTransceiverBySender(sender);
if (!transceiver || !sender->track()) {
return RTCError::OK();
}
sender->SetTrack(nullptr);
if (transceiver->direction() == RtpTransceiverDirection::kSendRecv) {
transceiver->internal()->SetDirection(RtpTransceiverDirection::kRecvOnly);
} else if (transceiver->direction() == RtpTransceiverDirection::kSendOnly) {
transceiver->internal()->SetDirection(RtpTransceiverDirection::kInactive);
}
} else {
bool removed;
if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
removed = GetAudioTransceiver()->internal()->RemoveSender(sender);
} else {
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, sender->media_type());
removed = GetVideoTransceiver()->internal()->RemoveSender(sender);
}
if (!removed) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Couldn't find sender " + sender->id() + " to remove.");
}
}
observer_->OnRenegotiationNeeded();
return RTCError::OK();
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::FindTransceiverBySender(
rtc::scoped_refptr<RtpSenderInterface> sender) {
for (auto transceiver : transceivers_) {
if (transceiver->sender() == sender) {
return transceiver;
}
}
return nullptr;
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
return AddTransceiver(track, RtpTransceiverInit());
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init) {
if (!IsUnifiedPlan()) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"AddTransceiver only supported when Unified Plan is enabled.");
}
if (!track) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "track is null");
}
cricket::MediaType media_type;
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
media_type = cricket::MEDIA_TYPE_AUDIO;
} else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
media_type = cricket::MEDIA_TYPE_VIDEO;
} else {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Track kind is not audio or video");
}
return AddTransceiver(media_type, track, init);
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(cricket::MediaType media_type) {
return AddTransceiver(media_type, RtpTransceiverInit());
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(cricket::MediaType media_type,
const RtpTransceiverInit& init) {
if (!IsUnifiedPlan()) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"AddTransceiver only supported when Unified Plan is enabled.");
}
if (!(media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"media type is not audio or video");
}
return AddTransceiver(media_type, nullptr, init);
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
cricket::MediaType media_type,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init) {
RTC_DCHECK((media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO));
if (track) {
RTC_DCHECK_EQ(media_type,
(track->kind() == MediaStreamTrackInterface::kAudioKind
? cricket::MEDIA_TYPE_AUDIO
: cricket::MEDIA_TYPE_VIDEO));
}
// TODO(bugs.webrtc.org/7600): Verify init.
auto transceiver = CreateTransceiver(media_type);
transceiver->SetDirection(init.direction);
if (track) {
transceiver->sender()->SetTrack(track);
}
observer_->OnRenegotiationNeeded();
return rtc::scoped_refptr<RtpTransceiverInterface>(transceiver);
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::CreateTransceiver(cricket::MediaType media_type) {
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender;
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver;
std::string receiver_id = rtc::CreateRandomUuid();
// TODO(bugs.webrtc.org/7600): Initializing the sender/receiver with a null
// channel prevents users from calling SetParameters on them, which is needed
// to be in compliance with the spec.
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), new AudioRtpSender(nullptr, stats_.get()));
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), new AudioRtpReceiver(receiver_id, {}, 0, nullptr));
} else {
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, media_type);
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), new VideoRtpSender(nullptr));
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(),
new VideoRtpReceiver(receiver_id, {}, worker_thread(), 0, nullptr));
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
transceiver = RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(), new RtpTransceiver(sender, receiver));
transceivers_.push_back(transceiver);
return transceiver;
}
rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
AudioTrackInterface* track) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateDtmfSender");
if (IsClosed()) {
return nullptr;
}
if (!track) {
RTC_LOG(LS_ERROR) << "CreateDtmfSender - track is NULL.";
return nullptr;
}
auto track_sender = FindSenderForTrack(track);
if (!track_sender) {
RTC_LOG(LS_ERROR) << "CreateDtmfSender called with a non-added track.";
return nullptr;
}
return track_sender->GetDtmfSender();
}
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
const std::string& kind,
const std::string& stream_id) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
if (IsClosed()) {
return nullptr;
}
// TODO(steveanton): Move construction of the RtpSenders to RtpTransceiver.
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
if (kind == MediaStreamTrackInterface::kAudioKind) {
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), new AudioRtpSender(voice_channel(), stats_.get()));
GetAudioTransceiver()->internal()->AddSender(new_sender);
} else if (kind == MediaStreamTrackInterface::kVideoKind) {
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), new VideoRtpSender(video_channel()));
GetVideoTransceiver()->internal()->AddSender(new_sender);
} else {
RTC_LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
return nullptr;
}
if (!stream_id.empty()) {
new_sender->internal()->set_stream_id(stream_id);
}
return new_sender;
}
std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
const {
std::vector<rtc::scoped_refptr<RtpSenderInterface>> ret;
for (auto sender : GetSendersInternal()) {
ret.push_back(sender);
}
return ret;
}
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
PeerConnection::GetSendersInternal() const {
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
all_senders;
for (auto transceiver : transceivers_) {
auto senders = transceiver->internal()->senders();
all_senders.insert(all_senders.end(), senders.begin(), senders.end());
}
return all_senders;
}
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
PeerConnection::GetReceivers() const {
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> ret;
for (const auto& receiver : GetReceiversInternal()) {
ret.push_back(receiver);
}
return ret;
}
std::vector<
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
PeerConnection::GetReceiversInternal() const {
std::vector<
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
all_receivers;
for (auto transceiver : transceivers_) {
auto receivers = transceiver->internal()->receivers();
all_receivers.insert(all_receivers.end(), receivers.begin(),
receivers.end());
}
return all_receivers;
}
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::GetTransceivers() const {
RTC_DCHECK(IsUnifiedPlan());
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers;
for (auto transceiver : transceivers_) {
all_transceivers.push_back(transceiver);
}
return all_transceivers;
}
bool PeerConnection::GetStats(StatsObserver* observer,
MediaStreamTrackInterface* track,
StatsOutputLevel level) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK(signaling_thread()->IsCurrent());
if (!observer) {
RTC_LOG(LS_ERROR) << "GetStats - observer is NULL.";
return false;
}
stats_->UpdateStats(level);
// The StatsCollector is used to tell if a track is valid because it may
// remember tracks that the PeerConnection previously removed.
if (track && !stats_->IsValidTrack(track->id())) {
RTC_LOG(LS_WARNING) << "GetStats is called with an invalid track: "
<< track->id();
return false;
}
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_GETSTATS,
new GetStatsMsg(observer, track));
return true;
}
void PeerConnection::GetStats(RTCStatsCollectorCallback* callback) {
RTC_DCHECK(stats_collector_);
stats_collector_->GetStatsReport(callback);
}
PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
return signaling_state_;
}
PeerConnectionInterface::IceConnectionState
PeerConnection::ice_connection_state() {
return ice_connection_state_;
}
PeerConnectionInterface::IceGatheringState
PeerConnection::ice_gathering_state() {
return ice_gathering_state_;
}
rtc::scoped_refptr<DataChannelInterface>
PeerConnection::CreateDataChannel(
const std::string& label,
const DataChannelInit* config) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
bool first_datachannel = !HasDataChannels();
std::unique_ptr<InternalDataChannelInit> internal_config;
if (config) {
internal_config.reset(new InternalDataChannelInit(*config));
}
rtc::scoped_refptr<DataChannelInterface> channel(
InternalCreateDataChannel(label, internal_config.get()));
if (!channel.get()) {
return nullptr;
}
// Trigger the onRenegotiationNeeded event for every new RTP DataChannel, or
// the first SCTP DataChannel.
if (data_channel_type() == cricket::DCT_RTP || first_datachannel) {
observer_->OnRenegotiationNeeded();
}
return DataChannelProxy::Create(signaling_thread(), channel.get());
}
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options;
// Always create an offer even if |ConvertConstraintsToOfferAnswerOptions|
// returns false for now. Because |ConvertConstraintsToOfferAnswerOptions|
// compares the mandatory fields parsed with the mandatory fields added in the
// |constraints| and some downstream applications might create offers with
// mandatory fields which would not be parsed in the helper method. For
// example, in Chromium/remoting, |kEnableDtlsSrtp| is added to the
// |constraints| as a mandatory field but it is not parsed.
ConvertConstraintsToOfferAnswerOptions(constraints, &offer_answer_options);
CreateOffer(observer, offer_answer_options);
}
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
if (!observer) {
RTC_LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
return;
}
if (IsClosed()) {
std::string error = "CreateOffer called when PeerConnection is closed.";
RTC_LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(observer, error);
return;
}
if (!ValidateOfferAnswerOptions(options)) {
std::string error = "CreateOffer called with invalid options.";
RTC_LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(observer, error);
return;
}
cricket::MediaSessionOptions session_options;
GetOptionsForOffer(options, &session_options);
webrtc_session_desc_factory_->CreateOffer(observer, options, session_options);
}
void PeerConnection::CreateAnswer(
CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
if (!observer) {
RTC_LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
return;
}
PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options;
if (!ConvertConstraintsToOfferAnswerOptions(constraints,
&offer_answer_options)) {
std::string error = "CreateAnswer called with invalid constraints.";
RTC_LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(observer, error);
return;
}
CreateAnswer(observer, offer_answer_options);
}
void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
if (!observer) {
RTC_LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
return;
}
if (IsClosed()) {
std::string error = "CreateAnswer called when PeerConnection is closed.";
RTC_LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(observer, error);
return;
}
if (remote_description() &&
remote_description()->GetType() != SdpType::kOffer) {
std::string error = "CreateAnswer called without remote offer.";
RTC_LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(observer, error);
return;
}
cricket::MediaSessionOptions session_options;
GetOptionsForAnswer(options, &session_options);
webrtc_session_desc_factory_->CreateAnswer(observer, session_options);
}
void PeerConnection::SetLocalDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {
TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
if (!observer) {
RTC_LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
return;
}
if (!desc) {
PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
return;
}
SdpType type = desc->GetType();
RTCError error = ApplyLocalDescription(rtc::WrapUnique(desc));
// |desc| may be destroyed at this point.
if (!error.ok()) {
std::ostringstream oss;
oss << "Failed to set local " << SdpTypeToString(type)
<< " sdp: " << error.message();
std::string error_message = oss.str();
RTC_LOG(LS_ERROR) << error_message << " (" << error.type() << ")";
PostSetSessionDescriptionFailure(observer, std::move(error_message));
return;
}
RTC_DCHECK(local_description());
PostSetSessionDescriptionSuccess(observer);
// According to JSEP, after setLocalDescription, changing the candidate pool
// size is not allowed, and changing the set of ICE servers will not result
// in new candidates being gathered.
port_allocator_->FreezeCandidatePool();
// MaybeStartGathering needs to be called after posting
// MSG_SET_SESSIONDESCRIPTION_SUCCESS, so that we don't signal any candidates
// before signaling that SetLocalDescription completed.
transport_controller_->MaybeStartGathering();
if (local_description()->GetType() == SdpType::kAnswer) {
// TODO(deadbeef): We already had to hop to the network thread for
// MaybeStartGathering...
network_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
port_allocator_.get()));
}
}
RTCError PeerConnection::ApplyLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(desc);
RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_LOCAL);
if (!error.ok()) {
return error;
}
// Update stats here so that we have the most recent stats for tracks and
// streams that might be removed by updating the session description.
stats_->UpdateStats(kStatsOutputLevelStandard);
// Update the initial_offerer flag if this session is the initial_offerer.
SdpType type = desc->GetType();
if (!initial_offerer_.has_value()) {
initial_offerer_.emplace(type == SdpType::kOffer);
if (*initial_offerer_) {
transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLING);
} else {
transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLED);
}
}
if (type == SdpType::kAnswer) {
current_local_description_ = std::move(desc);
pending_local_description_ = nullptr;
current_remote_description_ = std::move(pending_remote_description_);
} else {
pending_local_description_ = std::move(desc);
}
// The session description to apply now must be accessed by
// |local_description()|.
RTC_DCHECK(local_description());
// Transport and Media channels will be created only when offer is set.
if (type == SdpType::kOffer) {
// TODO(mallinath) - Handle CreateChannel failure, as new local description
// is applied. Restore back to old description.
RTCError error = CreateChannels(local_description()->description());
if (!error.ok()) {
return error;
}
}
// Remove unused channels if MediaContentDescription is rejected.
RemoveUnusedChannels(local_description()->description());
error = UpdateSessionState(type, cricket::CS_LOCAL);
if (!error.ok()) {
return error;
}
if (remote_description()) {
// Now that we have a local description, we can push down remote candidates.
UseCandidatesInSessionDescription(remote_description());
}
pending_ice_restarts_.clear();
if (session_error() != SessionError::kNone) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
}
// If setting the description decided our SSL role, allocate any necessary
// SCTP sids.
rtc::SSLRole role;
if (data_channel_type() == cricket::DCT_SCTP && GetSctpSslRole(&role)) {
AllocateSctpSids(role);
}
// Update state and SSRC of local MediaStreams and DataChannels based on the
// local session description.
const cricket::ContentInfo* audio_content =
GetFirstAudioContent(local_description()->description());
if (audio_content) {
if (audio_content->rejected) {
RemoveSenders(cricket::MEDIA_TYPE_AUDIO);
} else {
const cricket::AudioContentDescription* audio_desc =
static_cast<const cricket::AudioContentDescription*>(
audio_content->description);
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 =
static_cast<const cricket::VideoContentDescription*>(
video_content->description);
UpdateLocalSenders(video_desc->streams(), video_desc->type());
}
}
const cricket::ContentInfo* data_content =
GetFirstDataContent(local_description()->description());
if (data_content) {
const cricket::DataContentDescription* data_desc =
static_cast<const cricket::DataContentDescription*>(
data_content->description);
if (rtc::starts_with(data_desc->protocol().data(),
cricket::kMediaProtocolRtpPrefix)) {
UpdateLocalRtpDataChannels(data_desc->streams());
}
}
return RTCError::OK();
}
void PeerConnection::SetRemoteDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {
SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface>(desc),
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>(
new SetRemoteDescriptionObserverAdapter(this, observer)));
}
void PeerConnection::SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");
if (!observer) {
RTC_LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
return;
}
if (!desc) {
observer->OnSetRemoteDescriptionComplete(RTCError(
RTCErrorType::INVALID_PARAMETER, "SessionDescription is NULL."));
return;
}
const SdpType type = desc->GetType();
RTCError error = ApplyRemoteDescription(std::move(desc));
// |desc| may be destroyed at this point.
if (!error.ok()) {
std::ostringstream oss;
oss << "Failed to set remote " << SdpTypeToString(type)
<< " sdp: " << error.message();
std::string error_message = oss.str();
RTC_LOG(LS_ERROR) << error_message << " (" << error.type() << ")";
observer->OnSetRemoteDescriptionComplete(
RTCError(error.type(), std::move(error_message)));
return;
}
if (remote_description()->GetType() == SdpType::kAnswer) {
// TODO(deadbeef): We already had to hop to the network thread for
// MaybeStartGathering...
network_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
port_allocator_.get()));
}
observer->OnSetRemoteDescriptionComplete(RTCError::OK());
}
RTCError PeerConnection::ApplyRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(desc);
RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_REMOTE);
if (!error.ok()) {
return error;
}
// Update stats here so that we have the most recent stats for tracks and
// streams that might be removed by updating the session description.
stats_->UpdateStats(kStatsOutputLevelStandard);
// Takes the ownership of |desc|. On success, remote_description() is updated
// to reflect the description that was passed in.
const SessionDescriptionInterface* old_remote_description =
remote_description();
// Grab ownership of the description being replaced for the remainder of this
// method, since it's used below as |old_remote_description|.
std::unique_ptr<SessionDescriptionInterface> replaced_remote_description;
SdpType type = desc->GetType();
if (type == SdpType::kAnswer) {
replaced_remote_description = pending_remote_description_
? std::move(pending_remote_description_)
: std::move(current_remote_description_);
current_remote_description_ = std::move(desc);
pending_remote_description_ = nullptr;
current_local_description_ = std::move(pending_local_description_);
} else {
replaced_remote_description = std::move(pending_remote_description_);
pending_remote_description_ = std::move(desc);
}
// The session description to apply now must be accessed by
// |remote_description()|.
RTC_DCHECK(remote_description());
// Transport and Media channels will be created only when offer is set.
if (type == SdpType::kOffer) {
// TODO(mallinath) - Handle CreateChannel failure, as new local description
// is applied. Restore back to old description.
RTCError error = CreateChannels(remote_description()->description());
if (!error.ok()) {
return error;
}
}
// Remove unused channels if MediaContentDescription is rejected.
RemoveUnusedChannels(remote_description()->description());
// NOTE: Candidates allocation will be initiated only when SetLocalDescription
// is called.
error = UpdateSessionState(type, cricket::CS_REMOTE);
if (!error.ok()) {
return error;
}
if (local_description() &&
!UseCandidatesInSessionDescription(remote_description())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidCandidates);
}
if (old_remote_description) {
for (const cricket::ContentInfo& content :
old_remote_description->description()->contents()) {
// Check if this new SessionDescription contains new ICE ufrag and
// password that indicates the remote peer requests an ICE restart.
// TODO(deadbeef): When we start storing both the current and pending
// remote description, this should reset pending_ice_restarts and compare
// against the current description.
if (CheckForRemoteIceRestart(old_remote_description, remote_description(),
content.name)) {
if (type == SdpType::kOffer) {
pending_ice_restarts_.insert(content.name);
}
} else {
// We retain all received candidates only if ICE is not restarted.
// When ICE is restarted, all previous candidates belong to an old
// generation and should not be kept.
// TODO(deadbeef): This goes against the W3C spec which says the remote
// description should only contain candidates from the last set remote
// description plus any candidates added since then. We should remove
// this once we're sure it won't break anything.
WebRtcSessionDescriptionFactory::CopyCandidatesFromSessionDescription(
old_remote_description, content.name, mutable_remote_description());
}
}
}
if (session_error() != SessionError::kNone) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
}
// Set the the ICE connection state to connecting since the connection may
// become writable with peer reflexive candidates before any remote candidate
// is signaled.
// TODO(pthatcher): This is a short-term solution for crbug/446908. A real fix
// is to have a new signal the indicates a change in checking state from the
// transport and expose a new checking() member from transport that can be
// read to determine the current checking state. The existing SignalConnecting
// actually means "gathering candidates", so cannot be be used here.
if (remote_description()->GetType() != SdpType::kOffer &&
ice_connection_state() == PeerConnectionInterface::kIceConnectionNew) {
SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
}
// If setting the description decided our SSL role, allocate any necessary
// SCTP sids.
rtc::SSLRole role;
if (data_channel_type() == cricket::DCT_SCTP && GetSctpSslRole(&role)) {
AllocateSctpSids(role);
}
const cricket::ContentInfo* audio_content =
GetFirstAudioContent(remote_description()->description());
const cricket::ContentInfo* video_content =
GetFirstVideoContent(remote_description()->description());
const cricket::AudioContentDescription* audio_desc =
GetFirstAudioContentDescription(remote_description()->description());
const cricket::VideoContentDescription* video_desc =
GetFirstVideoContentDescription(remote_description()->description());
const cricket::DataContentDescription* data_desc =
GetFirstDataContentDescription(remote_description()->description());
// Check if the descriptions include streams, just in case the peer supports
// MSID, but doesn't indicate so with "a=msid-semantic".
if (remote_description()->description()->msid_supported() ||
(audio_desc && !audio_desc->streams().empty()) ||
(video_desc && !video_desc->streams().empty())) {
remote_peer_supports_msid_ = true;
}
// We wait to signal new streams until we finish processing the description,
// since only at that point will new streams have all their tracks.
rtc::scoped_refptr<StreamCollection> new_streams(StreamCollection::Create());
// TODO(steveanton): When removing RTP senders/receivers in response to a
// rejected media section, there is some cleanup logic that expects the voice/
// video channel to still be set. But in this method the voice/video channel
// would have been destroyed by the SetRemoteDescription caller above so the
// cleanup that relies on them fails to run. The RemoveSenders calls should be
// moved to right before the DestroyChannel calls to fix this.
// Find all audio rtp streams and create corresponding remote AudioTracks
// and MediaStreams.
if (audio_content) {
if (audio_content->rejected) {
RemoveSenders(cricket::MEDIA_TYPE_AUDIO);
} else {
bool default_audio_track_needed =
!remote_peer_supports_msid_ &&
RtpTransceiverDirectionHasSend(audio_desc->direction());
UpdateRemoteSendersList(GetActiveStreams(audio_desc),
default_audio_track_needed, audio_desc->type(),
new_streams);
}
}
// Find all video rtp streams and create corresponding remote VideoTracks
// and MediaStreams.
if (video_content) {
if (video_content->rejected) {
RemoveSenders(cricket::MEDIA_TYPE_VIDEO);
} else {
bool default_video_track_needed =
!remote_peer_supports_msid_ &&
RtpTransceiverDirectionHasSend(video_desc->direction());
UpdateRemoteSendersList(GetActiveStreams(video_desc),
default_video_track_needed, video_desc->type(),
new_streams);
}
}
// Update the DataChannels with the information from the remote peer.
if (data_desc) {
if (rtc::starts_with(data_desc->protocol().data(),
cricket::kMediaProtocolRtpPrefix)) {
UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc));
}
}
// Iterate new_streams and notify the observer about new MediaStreams.
for (size_t i = 0; i < new_streams->count(); ++i) {
MediaStreamInterface* new_stream = new_streams->at(i);
stats_->AddStream(new_stream);
observer_->OnAddStream(
rtc::scoped_refptr<MediaStreamInterface>(new_stream));
}
UpdateEndedRemoteMediaStreams();
return RTCError::OK();
}
const cricket::ContentInfo* PeerConnection::FindMediaSectionForTransceiver(
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
transceiver,
const SessionDescriptionInterface* sdesc) const {
RTC_DCHECK(transceiver);
RTC_DCHECK(sdesc);
if (IsUnifiedPlan()) {
if (!transceiver->internal()->mid()) {
// This transceiver is not associated with a media section yet.
return nullptr;
}
return sdesc->description()->GetContentByName(
*transceiver->internal()->mid());
} else {
// Plan B only allows at most one audio and one video section, so use the
// first media section of that type.
return cricket::GetFirstMediaContent(sdesc->description()->contents(),
transceiver->internal()->media_type());
}
}
PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
return configuration_;
}
bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
RTCError* error) {
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
if (local_description() && configuration.ice_candidate_pool_size !=
configuration_.ice_candidate_pool_size) {
RTC_LOG(LS_ERROR) << "Can't change candidate pool size after calling "
"SetLocalDescription.";
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
}
// The simplest (and most future-compatible) way to tell if the config was
// modified in an invalid way is to copy each property we do support
// modifying, then use operator==. There are far more properties we don't
// support modifying than those we do, and more could be added.
RTCConfiguration modified_config = configuration_;
modified_config.servers = configuration.servers;
modified_config.type = configuration.type;
modified_config.ice_candidate_pool_size =
configuration.ice_candidate_pool_size;
modified_config.prune_turn_ports = configuration.prune_turn_ports;
modified_config.ice_check_min_interval = configuration.ice_check_min_interval;
modified_config.turn_customizer = configuration.turn_customizer;
if (configuration != modified_config) {
RTC_LOG(LS_ERROR) << "Modifying the configuration in an unsupported way.";
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
}
// Validate the modified configuration.
RTCError validate_error = ValidateConfiguration(modified_config);
if (!validate_error.ok()) {
return SafeSetError(std::move(validate_error), error);
}
// Note that this isn't possible through chromium, since it's an unsigned
// short in WebIDL.
if (configuration.ice_candidate_pool_size < 0 ||
configuration.ice_candidate_pool_size > UINT16_MAX) {
return SafeSetError(RTCErrorType::INVALID_RANGE, error);
}
// Parse ICE servers before hopping to network thread.
cricket::ServerAddresses stun_servers;
std::vector<cricket::RelayServerConfig> turn_servers;
RTCErrorType parse_error =
ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
if (parse_error != RTCErrorType::NONE) {
return SafeSetError(parse_error, error);
}
// In theory this shouldn't fail.
if (!network_thread()->Invoke<bool>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::ReconfigurePortAllocator_n, this,
stun_servers, turn_servers, modified_config.type,
modified_config.ice_candidate_pool_size,
modified_config.prune_turn_ports,
modified_config.turn_customizer))) {
RTC_LOG(LS_ERROR) << "Failed to apply configuration to PortAllocator.";
return SafeSetError(RTCErrorType::INTERNAL_ERROR, error);
}
// As described in JSEP, calling setConfiguration with new ICE servers or
// candidate policy must set a "needs-ice-restart" bit so that the next offer
// triggers an ICE restart which will pick up the changes.
if (modified_config.servers != configuration_.servers ||
modified_config.type != configuration_.type ||
modified_config.prune_turn_ports != configuration_.prune_turn_ports) {
transport_controller_->SetNeedsIceRestartFlag();
}
if (modified_config.ice_check_min_interval !=
configuration_.ice_check_min_interval) {
transport_controller_->SetIceConfig(ParseIceConfig(modified_config));
}
configuration_ = modified_config;
return SafeSetError(RTCErrorType::NONE, error);
}
bool PeerConnection::AddIceCandidate(
const IceCandidateInterface* ice_candidate) {
TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
if (IsClosed()) {
return false;
}
if (!remote_description()) {
RTC_LOG(LS_ERROR) << "ProcessIceMessage: ICE candidates can't be added "
<< "without any remote session description.";
return false;
}
if (!ice_candidate) {
RTC_LOG(LS_ERROR) << "ProcessIceMessage: Candidate is NULL.";
return false;
}
bool valid = false;
bool ready = ReadyToUseRemoteCandidate(ice_candidate, nullptr, &valid);
if (!valid) {
return false;
}
// Add this candidate to the remote session description.
if (!mutable_remote_description()->AddCandidate(ice_candidate)) {
RTC_LOG(LS_ERROR) << "ProcessIceMessage: Candidate cannot be used.";
return false;
}
if (ready) {
return UseCandidate(ice_candidate);
} else {
RTC_LOG(LS_INFO) << "ProcessIceMessage: Not ready to use candidate.";
return true;
}
}
bool PeerConnection::RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
if (!remote_description()) {
RTC_LOG(LS_ERROR) << "RemoveRemoteIceCandidates: ICE candidates can't be "
<< "removed without any remote session description.";
return false;
}
if (candidates.empty()) {
RTC_LOG(LS_ERROR) << "RemoveRemoteIceCandidates: candidates are empty.";
return false;
}
size_t number_removed =
mutable_remote_description()->RemoveCandidates(candidates);
if (number_removed != candidates.size()) {
RTC_LOG(LS_ERROR)
<< "RemoveRemoteIceCandidates: Failed to remove candidates. "
<< "Requested " << candidates.size() << " but only " << number_removed
<< " are removed.";
}
// Remove the candidates from the transport controller.
std::string error;
bool res = transport_controller_->RemoveRemoteCandidates(candidates, &error);
if (!res && !error.empty()) {
RTC_LOG(LS_ERROR) << "Error when removing remote candidates: " << error;
}
return true;
}
void PeerConnection::RegisterUMAObserver(UMAObserver* observer) {
TRACE_EVENT0("webrtc", "PeerConnection::RegisterUmaObserver");
uma_observer_ = observer;
if (transport_controller()) {
transport_controller()->SetMetricsObserver(uma_observer_);
}
// Send information about IPv4/IPv6 status.
if (uma_observer_) {
port_allocator_->SetMetricsObserver(uma_observer_);
if (port_allocator_->flags() & cricket::PORTALLOCATOR_ENABLE_IPV6) {
uma_observer_->IncrementEnumCounter(
kEnumCounterAddressFamily, kPeerConnection_IPv6,
kPeerConnectionAddressFamilyCounter_Max);
} else {
uma_observer_->IncrementEnumCounter(
kEnumCounterAddressFamily, kPeerConnection_IPv4,
kPeerConnectionAddressFamilyCounter_Max);
}
}
}
RTCError PeerConnection::SetBitrate(const BitrateParameters& bitrate) {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->Invoke<RTCError>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::SetBitrate, this, bitrate));
}
const bool has_min = static_cast<bool>(bitrate.min_bitrate_bps);
const bool has_current = static_cast<bool>(bitrate.current_bitrate_bps);
const bool has_max = static_cast<bool>(bitrate.max_bitrate_bps);
if (has_min && *bitrate.min_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"min_bitrate_bps <= 0");
}
if (has_current) {
if (has_min && *bitrate.current_bitrate_bps < *bitrate.min_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"current_bitrate_bps < min_bitrate_bps");
} else if (*bitrate.current_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"curent_bitrate_bps < 0");
}
}
if (has_max) {
if (has_current &&
*bitrate.max_bitrate_bps < *bitrate.current_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < current_bitrate_bps");
} else if (has_min && *bitrate.max_bitrate_bps < *bitrate.min_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < min_bitrate_bps");
} else if (*bitrate.max_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < 0");
}
}
Call::Config::BitrateConfigMask mask;
mask.min_bitrate_bps = bitrate.min_bitrate_bps;
mask.start_bitrate_bps = bitrate.current_bitrate_bps;
mask.max_bitrate_bps = bitrate.max_bitrate_bps;
RTC_DCHECK(call_.get());
call_->SetBitrateConfigMask(mask);
return RTCError::OK();
}
void PeerConnection::SetBitrateAllocationStrategy(
std::unique_ptr<rtc::BitrateAllocationStrategy>
bitrate_allocation_strategy) {
rtc::Thread* worker_thread = factory_->worker_thread();
if (!worker_thread->IsCurrent()) {
rtc::BitrateAllocationStrategy* strategy_raw =
bitrate_allocation_strategy.release();
auto functor = [this, strategy_raw]() {
call_->SetBitrateAllocationStrategy(
rtc::WrapUnique<rtc::BitrateAllocationStrategy>(strategy_raw));
};
worker_thread->Invoke<void>(RTC_FROM_HERE, functor);
return;
}
RTC_DCHECK(call_.get());
call_->SetBitrateAllocationStrategy(std::move(bitrate_allocation_strategy));
}
void PeerConnection::SetAudioPlayout(bool playout) {
if (!worker_thread()->IsCurrent()) {
worker_thread()->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::SetAudioPlayout, this, playout));
return;
}
auto audio_state =
factory_->channel_manager()->media_engine()->GetAudioState();
audio_state->SetPlayout(playout);
}
void PeerConnection::SetAudioRecording(bool recording) {
if (!worker_thread()->IsCurrent()) {
worker_thread()->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::SetAudioRecording, this, recording));
return;
}
auto audio_state =
factory_->channel_manager()->media_engine()->GetAudioState();
audio_state->SetRecording(recording);
}
std::unique_ptr<rtc::SSLCertificate>
PeerConnection::GetRemoteAudioSSLCertificate() {
if (!voice_channel()) {
return nullptr;
}
return GetRemoteSSLCertificate(voice_channel()->transport_name());
}
bool PeerConnection::StartRtcEventLog(rtc::PlatformFile file,
int64_t max_size_bytes) {
// TODO(eladalon): It would be better to not allow negative values into PC.
const size_t max_size = (max_size_bytes < 0)
? RtcEventLog::kUnlimitedOutput
: rtc::saturated_cast<size_t>(max_size_bytes);
return StartRtcEventLog(
rtc::MakeUnique<RtcEventLogOutputFile>(file, max_size),
webrtc::RtcEventLog::kImmediateOutput);
}
bool PeerConnection::StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) {
// TODO(eladalon): In C++14, this can be done with a lambda.
struct Functor {
bool operator()() {
return pc->StartRtcEventLog_w(std::move(output), output_period_ms);
}
PeerConnection* const pc;
std::unique_ptr<RtcEventLogOutput> output;
const int64_t output_period_ms;
};
return worker_thread()->Invoke<bool>(
RTC_FROM_HERE, Functor{this, std::move(output), output_period_ms});
}
void PeerConnection::StopRtcEventLog() {
worker_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::StopRtcEventLog_w, this));
}
const SessionDescriptionInterface* PeerConnection::local_description() const {
return pending_local_description_ ? pending_local_description_.get()
: current_local_description_.get();
}
const SessionDescriptionInterface* PeerConnection::remote_description() const {
return pending_remote_description_ ? pending_remote_description_.get()
: current_remote_description_.get();
}
const SessionDescriptionInterface* PeerConnection::current_local_description()
const {
return current_local_description_.get();
}
const SessionDescriptionInterface* PeerConnection::current_remote_description()
const {
return current_remote_description_.get();
}
const SessionDescriptionInterface* PeerConnection::pending_local_description()
const {
return pending_local_description_.get();
}
const SessionDescriptionInterface* PeerConnection::pending_remote_description()
const {
return pending_remote_description_.get();
}
void PeerConnection::Close() {
TRACE_EVENT0("webrtc", "PeerConnection::Close");
// Update stats here so that we have the most recent stats for tracks and
// streams before the channels are closed.
stats_->UpdateStats(kStatsOutputLevelStandard);
ChangeSignalingState(PeerConnectionInterface::kClosed);
StopAndDestroyChannels();
network_thread()->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
port_allocator_.get()));
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
call_.reset();
// The event log must outlive call (and any other object that uses it).
event_log_.reset();
});
}
void PeerConnection::OnMessage(rtc::Message* msg) {
switch (msg->message_id) {
case MSG_SET_SESSIONDESCRIPTION_SUCCESS: {
SetSessionDescriptionMsg* param =
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
param->observer->OnSuccess();
delete param;
break;
}
case MSG_SET_SESSIONDESCRIPTION_FAILED: {
SetSessionDescriptionMsg* param =
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
param->observer->OnFailure(param->error);
delete param;
break;
}
case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
CreateSessionDescriptionMsg* param =
static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
param->observer->OnFailure(param->error);
delete param;
break;
}
case MSG_GETSTATS: {
GetStatsMsg* param = static_cast<GetStatsMsg*>(msg->pdata);
StatsReports reports;
stats_->GetStats(param->track, &reports);
param->observer->OnComplete(reports);
delete param;
break;
}
case MSG_FREE_DATACHANNELS: {
sctp_data_channels_to_free_.clear();
break;
}
default:
RTC_NOTREACHED() << "Not implemented";
break;
}
}
void PeerConnection::CreateAudioReceiver(
MediaStreamInterface* stream,
const RtpSenderInfo& remote_sender_info) {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(),
new AudioRtpReceiver(remote_sender_info.sender_id, streams,
remote_sender_info.first_ssrc, voice_channel()));
stream->AddTrack(
static_cast<AudioTrackInterface*>(receiver->internal()->track().get()));
GetAudioTransceiver()->internal()->AddReceiver(receiver);
observer_->OnAddTrack(receiver, std::move(streams));
}
void PeerConnection::CreateVideoReceiver(
MediaStreamInterface* stream,
const RtpSenderInfo& remote_sender_info) {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(),
new VideoRtpReceiver(remote_sender_info.sender_id, streams,
worker_thread(), remote_sender_info.first_ssrc,
video_channel()));
stream->AddTrack(
static_cast<VideoTrackInterface*>(receiver->internal()->track().get()));
GetVideoTransceiver()->internal()->AddReceiver(receiver);
observer_->OnAddTrack(receiver, std::move(streams));
}
// TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
// description.
rtc::scoped_refptr<RtpReceiverInterface> PeerConnection::RemoveAndStopReceiver(
const RtpSenderInfo& remote_sender_info) {
auto receiver = FindReceiverById(remote_sender_info.sender_id);
if (!receiver) {
RTC_LOG(LS_WARNING) << "RtpReceiver for track with id "
<< remote_sender_info.sender_id << " doesn't exist.";
return nullptr;
}
if (receiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
GetAudioTransceiver()->internal()->RemoveReceiver(receiver);
} else {
GetVideoTransceiver()->internal()->RemoveReceiver(receiver);
}
return receiver;
}
void PeerConnection::AddAudioTrack(AudioTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
auto sender = FindSenderForTrack(track);
if (sender) {
// We already have a sender for this track, so just change the stream_id
// so that it's correct in the next call to CreateOffer.
sender->internal()->set_stream_id(stream->label());
return;
}
// Normal case; we've never seen this track before.
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender =
RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(),
new AudioRtpSender(track, {stream->label()}, voice_channel(),
stats_.get()));
GetAudioTransceiver()->internal()->AddSender(new_sender);
// If the sender has already been configured in SDP, we call SetSsrc,
// which will connect the sender to the underlying transport. This can
// occur if a local session description that contains the ID of the sender
// is set before AddStream is called. It can also occur if the local
// session description is not changed and RemoveStream is called, and
// later AddStream is called again with the same stream.
const RtpSenderInfo* sender_info =
FindSenderInfo(local_audio_sender_infos_, stream->label(), track->id());
if (sender_info) {
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
}
}
// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
// indefinitely, when we have unified plan SDP.
void PeerConnection::RemoveAudioTrack(AudioTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
auto sender = FindSenderForTrack(track);
if (!sender) {
RTC_LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
<< " doesn't exist.";
return;
}
GetAudioTransceiver()->internal()->RemoveSender(sender);
}
void PeerConnection::AddVideoTrack(VideoTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
auto sender = FindSenderForTrack(track);
if (sender) {
// We already have a sender for this track, so just change the stream_id
// so that it's correct in the next call to CreateOffer.
sender->internal()->set_stream_id(stream->label());
return;
}
// Normal case; we've never seen this track before.
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender =
RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(),
new VideoRtpSender(track, {stream->label()}, video_channel()));
GetVideoTransceiver()->internal()->AddSender(new_sender);
const RtpSenderInfo* sender_info =
FindSenderInfo(local_video_sender_infos_, stream->label(), track->id());
if (sender_info) {
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
}
}
void PeerConnection::RemoveVideoTrack(VideoTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
auto sender = FindSenderForTrack(track);
if (!sender) {
RTC_LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
<< " doesn't exist.";
return;
}
GetVideoTransceiver()->internal()->RemoveSender(sender);
}
void PeerConnection::SetIceConnectionState(IceConnectionState new_state) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (ice_connection_state_ == new_state) {
return;
}
// After transitioning to "closed", ignore any additional states from
// TransportController (such as "disconnected").
if (IsClosed()) {
return;
}
RTC_LOG(LS_INFO) << "Changing IceConnectionState " << ice_connection_state_
<< " => " << new_state;
RTC_DCHECK(ice_connection_state_ !=
PeerConnectionInterface::kIceConnectionClosed);
ice_connection_state_ = new_state;
observer_->OnIceConnectionChange(ice_connection_state_);
}
void PeerConnection::OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (IsClosed()) {
return;
}
ice_gathering_state_ = new_state;
observer_->OnIceGatheringChange(ice_gathering_state_);
}
void PeerConnection::OnIceCandidate(
std::unique_ptr<IceCandidateInterface> candidate) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (IsClosed()) {
return;
}
observer_->OnIceCandidate(candidate.get());
}
void PeerConnection::OnIceCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (IsClosed()) {
return;
}
observer_->OnIceCandidatesRemoved(candidates);
}
void PeerConnection::ChangeSignalingState(
PeerConnectionInterface::SignalingState signaling_state) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (signaling_state_ == signaling_state) {
return;
}
RTC_LOG(LS_INFO) << "Session: " << session_id() << " Old state: "
<< GetSignalingStateString(signaling_state_)
<< " New state: "
<< GetSignalingStateString(signaling_state);
signaling_state_ = signaling_state;
if (signaling_state == kClosed) {
ice_connection_state_ = kIceConnectionClosed;
observer_->OnIceConnectionChange(ice_connection_state_);
if (ice_gathering_state_ != kIceGatheringComplete) {
ice_gathering_state_ = kIceGatheringComplete;
observer_->OnIceGatheringChange(ice_gathering_state_);
}
}
observer_->OnSignalingChange(signaling_state_);
}
void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
AddAudioTrack(track, stream);
observer_->OnRenegotiationNeeded();
}
void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
RemoveAudioTrack(track, stream);
observer_->OnRenegotiationNeeded();
}
void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
AddVideoTrack(track, stream);
observer_->OnRenegotiationNeeded();
}
void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
RemoveVideoTrack(track, stream);
observer_->OnRenegotiationNeeded();
}
void PeerConnection::PostSetSessionDescriptionSuccess(
SetSessionDescriptionObserver* observer) {
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
signaling_thread()->Post(RTC_FROM_HERE, this,
MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
}
void PeerConnection::PostSetSessionDescriptionFailure(
SetSessionDescriptionObserver* observer,
const std::string& error) {
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
msg->error = error;
signaling_thread()->Post(RTC_FROM_HERE, this,
MSG_SET_SESSIONDESCRIPTION_FAILED, msg);
}
void PeerConnection::PostCreateSessionDescriptionFailure(
CreateSessionDescriptionObserver* observer,
const std::string& error) {
CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
msg->error = error;
signaling_thread()->Post(RTC_FROM_HERE, this,
MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
}
void PeerConnection::GetOptionsForOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options) {
ExtractSharedMediaSessionOptions(rtc_options, session_options);
// Figure out transceiver directional preferences.
bool send_audio = HasRtpSender(cricket::MEDIA_TYPE_AUDIO);
bool send_video = HasRtpSender(cricket::MEDIA_TYPE_VIDEO);
// By default, generate sendrecv/recvonly m= sections.
bool recv_audio = true;
bool recv_video = true;
// By default, only offer a new m= section if we have media to send with it.
bool offer_new_audio_description = send_audio;
bool offer_new_video_description = send_video;
bool offer_new_data_description = HasDataChannels();
// The "offer_to_receive_X" options allow those defaults to be overridden.
if (rtc_options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
recv_audio = (rtc_options.offer_to_receive_audio > 0);
offer_new_audio_description =
offer_new_audio_description || (rtc_options.offer_to_receive_audio > 0);
}
if (rtc_options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
recv_video = (rtc_options.offer_to_receive_video > 0);
offer_new_video_description =
offer_new_video_description || (rtc_options.offer_to_receive_video > 0);
}
rtc::Optional<size_t> audio_index;
rtc::Optional<size_t> video_index;
rtc::Optional<size_t> data_index;
// If a current description exists, generate m= sections in the same order,
// using the first audio/video/data section that appears and rejecting
// extraneous ones.
if (local_description()) {
GenerateMediaDescriptionOptions(
local_description(),
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
RtpTransceiverDirectionFromSendRecv(send_video, recv_video),
&audio_index, &video_index, &data_index, session_options);
}
// Add audio/video/data m= sections to the end if needed.
if (!audio_index && offer_new_audio_description) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_AUDIO, cricket::CN_AUDIO,
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
false));
audio_index = session_options->media_description_options.size() - 1;
}
if (!video_index && offer_new_video_description) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO,
RtpTransceiverDirectionFromSendRecv(send_video, recv_video),
false));
video_index = session_options->media_description_options.size() - 1;
}
if (!data_index && offer_new_data_description) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_DATA, cricket::CN_DATA,
RtpTransceiverDirection::kSendRecv, false));
data_index = session_options->media_description_options.size() - 1;
}
cricket::MediaDescriptionOptions* audio_media_description_options =
!audio_index ? nullptr
: &session_options->media_description_options[*audio_index];
cricket::MediaDescriptionOptions* video_media_description_options =
!video_index ? nullptr
: &session_options->media_description_options[*video_index];
cricket::MediaDescriptionOptions* data_media_description_options =
!data_index ? nullptr
: &session_options->media_description_options[*data_index];
// Apply ICE restart flag and renomination flag.
for (auto& options : session_options->media_description_options) {
options.transport_options.ice_restart = rtc_options.ice_restart;
options.transport_options.enable_ice_renomination =
configuration_.enable_ice_renomination;
}
AddRtpSenderOptions(GetSendersInternal(), audio_media_description_options,
video_media_description_options);
AddRtpDataChannelOptions(rtp_data_channels_, data_media_description_options);
// Intentionally unset the data channel type for RTP data channel with the
// second condition. Otherwise the RTP data channels would be successfully
// negotiated by default and the unit tests in WebRtcDataBrowserTest will fail
// when building with chromium. We want to leave RTP data channels broken, so
// people won't try to use them.
if (!rtp_data_channels_.empty() || data_channel_type() != cricket::DCT_RTP) {
session_options->data_channel_type = data_channel_type();
}
session_options->rtcp_cname = rtcp_cname_;
session_options->crypto_options = factory_->options().crypto_options;
}
void PeerConnection::GetOptionsForAnswer(
const RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options) {
ExtractSharedMediaSessionOptions(rtc_options, session_options);
// Figure out transceiver directional preferences.
bool send_audio = HasRtpSender(cricket::MEDIA_TYPE_AUDIO);
bool send_video = HasRtpSender(cricket::MEDIA_TYPE_VIDEO);
// By default, generate sendrecv/recvonly m= sections. The direction is also
// restricted by the direction in the offer.
bool recv_audio = true;
bool recv_video = true;
// The "offer_to_receive_X" options allow those defaults to be overridden.
if (rtc_options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
recv_audio = (rtc_options.offer_to_receive_audio > 0);
}
if (rtc_options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
recv_video = (rtc_options.offer_to_receive_video > 0);
}
rtc::Optional<size_t> audio_index;
rtc::Optional<size_t> video_index;
rtc::Optional<size_t> data_index;
if (remote_description()) {
// The pending remote description should be an offer.
RTC_DCHECK(remote_description()->GetType() == SdpType::kOffer);
// Generate m= sections that match those in the offer.
// Note that mediasession.cc will handle intersection our preferred
// direction with the offered direction.
GenerateMediaDescriptionOptions(
remote_description(),
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
RtpTransceiverDirectionFromSendRecv(send_video, recv_video),
&audio_index, &video_index, &data_index, session_options);
}
cricket::MediaDescriptionOptions* audio_media_description_options =
!audio_index ? nullptr
: &session_options->media_description_options[*audio_index];
cricket::MediaDescriptionOptions* video_media_description_options =
!video_index ? nullptr
: &session_options->media_description_options[*video_index];
cricket::MediaDescriptionOptions* data_media_description_options =
!data_index ? nullptr
: &session_options->media_description_options[*data_index];
// Apply ICE renomination flag.
for (auto& options : session_options->media_description_options) {
options.transport_options.enable_ice_renomination =
configuration_.enable_ice_renomination;
}
AddRtpSenderOptions(GetSendersInternal(), audio_media_description_options,
video_media_description_options);
AddRtpDataChannelOptions(rtp_data_channels_, data_media_description_options);
// Intentionally unset the data channel type for RTP data channel. Otherwise
// the RTP data channels would be successfully negotiated by default and the
// unit tests in WebRtcDataBrowserTest will fail when building with chromium.
// We want to leave RTP data channels broken, so people won't try to use them.
if (!rtp_data_channels_.empty() || data_channel_type() != cricket::DCT_RTP) {
session_options->data_channel_type = data_channel_type();
}
session_options->rtcp_cname = rtcp_cname_;
session_options->crypto_options = factory_->options().crypto_options;
}
void PeerConnection::GenerateMediaDescriptionOptions(
const SessionDescriptionInterface* session_desc,
RtpTransceiverDirection audio_direction,
RtpTransceiverDirection video_direction,
rtc::Optional<size_t>* audio_index,
rtc::Optional<size_t>* video_index,
rtc::Optional<size_t>* data_index,
cricket::MediaSessionOptions* session_options) {
for (const cricket::ContentInfo& content :
session_desc->description()->contents()) {
if (IsAudioContent(&content)) {
// If we already have an audio m= section, reject this extra one.
if (*audio_index) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_AUDIO, content.name,
RtpTransceiverDirection::kInactive, true));
} else {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_AUDIO, content.name, audio_direction,
audio_direction == RtpTransceiverDirection::kInactive));
*audio_index = session_options->media_description_options.size() - 1;
}
} else if (IsVideoContent(&content)) {
// If we already have an video m= section, reject this extra one.
if (*video_index) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_VIDEO, content.name,
RtpTransceiverDirection::kInactive, true));
} else {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_VIDEO, content.name, video_direction,
video_direction == RtpTransceiverDirection::kInactive));
*video_index = session_options->media_description_options.size() - 1;
}
} else {
RTC_DCHECK(IsDataContent(&content));
// If we already have an data m= section, reject this extra one.
if (*data_index) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_DATA, content.name,
RtpTransceiverDirection::kInactive, true));
} else {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_DATA, content.name,
// Direction for data sections is meaningless, but legacy
// endpoints might expect sendrecv.
RtpTransceiverDirection::kSendRecv, false));
*data_index = session_options->media_description_options.size() - 1;
}
}
}
}
void PeerConnection::RemoveSenders(cricket::MediaType media_type) {
UpdateLocalSenders(std::vector<cricket::StreamParams>(), media_type);
UpdateRemoteSendersList(std::vector<cricket::StreamParams>(), false,
media_type, nullptr);
}
void PeerConnection::UpdateRemoteSendersList(
const cricket::StreamParamsVec& streams,
bool default_sender_needed,
cricket::MediaType media_type,
StreamCollection* new_streams) {
std::vector<RtpSenderInfo>* current_senders =
GetRemoteSenderInfos(media_type);
// Find removed senders. I.e., senders where the sender id or ssrc don't match
// the new StreamParam.
for (auto sender_it = current_senders->begin();
sender_it != current_senders->end();
/* incremented manually */) {
const RtpSenderInfo& info = *sender_it;
const cricket::StreamParams* params =
cricket::GetStreamBySsrc(streams, info.first_ssrc);
bool sender_exists = params && params->id == info.sender_id;
// If this is a default track, and we still need it, don't remove it.
if ((info.stream_label == kDefaultStreamLabel && default_sender_needed) ||
sender_exists) {
++sender_it;
} else {
OnRemoteSenderRemoved(info, media_type);
sender_it = current_senders->erase(sender_it);
}
}
// Find new and active senders.
for (const cricket::StreamParams& params : streams) {
// The sync_label is the MediaStream label and the |stream.id| is the
// sender id.
const std::string& stream_label = params.sync_label;
const std::string& sender_id = params.id;
uint32_t ssrc = params.first_ssrc();
rtc::scoped_refptr<MediaStreamInterface> stream =
remote_streams_->find(stream_label);
if (!stream) {
// This is a new MediaStream. Create a new remote MediaStream.
stream = MediaStreamProxy::Create(rtc::Thread::Current(),
MediaStream::Create(stream_label));
remote_streams_->AddStream(stream);
new_streams->AddStream(stream);
}
const RtpSenderInfo* sender_info =
FindSenderInfo(*current_senders, stream_label, sender_id);
if (!sender_info) {
current_senders->push_back(RtpSenderInfo(stream_label, sender_id, ssrc));
OnRemoteSenderAdded(current_senders->back(), media_type);
}
}
// Add default sender if necessary.
if (default_sender_needed) {
rtc::scoped_refptr<MediaStreamInterface> default_stream =
remote_streams_->find(kDefaultStreamLabel);
if (!default_stream) {
// Create the new default MediaStream.
default_stream = MediaStreamProxy::Create(
rtc::Thread::Current(), MediaStream::Create(kDefaultStreamLabel));
remote_streams_->AddStream(default_stream);
new_streams->AddStream(default_stream);
}
std::string default_sender_id = (media_type == cricket::MEDIA_TYPE_AUDIO)
? kDefaultAudioSenderId
: kDefaultVideoSenderId;
const RtpSenderInfo* default_sender_info = FindSenderInfo(
*current_senders, kDefaultStreamLabel, default_sender_id);
if (!default_sender_info) {
current_senders->push_back(
RtpSenderInfo(kDefaultStreamLabel, default_sender_id, 0));
OnRemoteSenderAdded(current_senders->back(), media_type);
}
}
}
void PeerConnection::OnRemoteSenderAdded(const RtpSenderInfo& sender_info,
cricket::MediaType media_type) {
MediaStreamInterface* stream =
remote_streams_->find(sender_info.stream_label);
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
CreateAudioReceiver(stream, sender_info);
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
CreateVideoReceiver(stream, sender_info);
} else {
RTC_NOTREACHED() << "Invalid media type";
}
}
void PeerConnection::OnRemoteSenderRemoved(const RtpSenderInfo& sender_info,
cricket::MediaType media_type) {
MediaStreamInterface* stream =
remote_streams_->find(sender_info.stream_label);
rtc::scoped_refptr<RtpReceiverInterface> receiver;
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
// When the MediaEngine audio channel is destroyed, the RemoteAudioSource
// will be notified which will end the AudioRtpReceiver::track().
receiver = RemoveAndStopReceiver(sender_info);
rtc::scoped_refptr<AudioTrackInterface> audio_track =
stream->FindAudioTrack(sender_info.sender_id);
if (audio_track) {
stream->RemoveTrack(audio_track);
}
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
// Stopping or destroying a VideoRtpReceiver will end the
// VideoRtpReceiver::track().
receiver = RemoveAndStopReceiver(sender_info);
rtc::scoped_refptr<VideoTrackInterface> video_track =
stream->FindVideoTrack(sender_info.sender_id);
if (video_track) {
// There's no guarantee the track is still available, e.g. the track may
// have been removed from the stream by an application.
stream->RemoveTrack(video_track);
}
} else {
RTC_NOTREACHED() << "Invalid media type";
}
if (receiver) {
observer_->OnRemoveTrack(receiver);
}
}
void PeerConnection::UpdateEndedRemoteMediaStreams() {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_to_remove;
for (size_t i = 0; i < remote_streams_->count(); ++i) {
MediaStreamInterface* stream = remote_streams_->at(i);
if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
streams_to_remove.push_back(stream);
}
}
for (auto& stream : streams_to_remove) {
remote_streams_->RemoveStream(stream);
observer_->OnRemoveStream(std::move(stream));
}
}
void PeerConnection::UpdateLocalSenders(
const std::vector<cricket::StreamParams>& streams,
cricket::MediaType media_type) {
std::vector<RtpSenderInfo>* current_senders = GetLocalSenderInfos(media_type);
// Find removed tracks. I.e., tracks where the track id, stream label or ssrc
// don't match the new StreamParam.
for (auto sender_it = current_senders->begin();
sender_it != current_senders->end();
/* incremented manually */) {
const RtpSenderInfo& info = *sender_it;
const cricket::StreamParams* params =
cricket::GetStreamBySsrc(streams, info.first_ssrc);
if (!params || params->id != info.sender_id ||
params->sync_label != info.stream_label) {
OnLocalSenderRemoved(info, media_type);
sender_it = current_senders->erase(sender_it);
} else {
++sender_it;
}
}
// Find new and active senders.
for (const cricket::StreamParams& params : streams) {
// The sync_label is the MediaStream label and the |stream.id| is the
// sender id.
const std::string& stream_label = params.sync_label;
const std::string& sender_id = params.id;
uint32_t ssrc = params.first_ssrc();
const RtpSenderInfo* sender_info =
FindSenderInfo(*current_senders, stream_label, sender_id);
if (!sender_info) {
current_senders->push_back(RtpSenderInfo(stream_label, sender_id, ssrc));
OnLocalSenderAdded(current_senders->back(), media_type);
}
}
}
void PeerConnection::OnLocalSenderAdded(const RtpSenderInfo& sender_info,
cricket::MediaType media_type) {
auto sender = FindSenderById(sender_info.sender_id);
if (!sender) {
RTC_LOG(LS_WARNING) << "An unknown RtpSender with id "
<< sender_info.sender_id
<< " has been configured in the local description.";
return;
}
if (sender->media_type() != media_type) {
RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
<< " description with an unexpected media type.";
return;
}
sender->internal()->set_stream_id(sender_info.stream_label);
sender->internal()->SetSsrc(sender_info.first_ssrc);
}
void PeerConnection::OnLocalSenderRemoved(const RtpSenderInfo& sender_info,
cricket::MediaType media_type) {
auto sender = FindSenderById(sender_info.sender_id);
if (!sender) {
// This is the normal case. I.e., RemoveStream has been called and the
// SessionDescriptions has been renegotiated.
return;
}
// A sender has been removed from the SessionDescription but it's still
// associated with the PeerConnection. This only occurs if the SDP doesn't
// match with the calls to CreateSender, AddStream and RemoveStream.
if (sender->media_type() != media_type) {
RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
<< " description with an unexpected media type.";
return;
}
sender->internal()->SetSsrc(0);
}
void PeerConnection::UpdateLocalRtpDataChannels(
const cricket::StreamParamsVec& streams) {
std::vector<std::string> existing_channels;
// Find new and active data channels.
for (const cricket::StreamParams& params : streams) {
// |it->sync_label| is actually the data channel label. The reason is that
// we use the same naming of data channels as we do for
// MediaStreams and Tracks.
// For MediaStreams, the sync_label is the MediaStream label and the
// track label is the same as |streamid|.
const std::string& channel_label = params.sync_label;
auto data_channel_it = rtp_data_channels_.find(channel_label);
if (data_channel_it == rtp_data_channels_.end()) {
RTC_LOG(LS_ERROR) << "channel label not found";
continue;
}
// Set the SSRC the data channel should use for sending.
data_channel_it->second->SetSendSsrc(params.first_ssrc());
existing_channels.push_back(data_channel_it->first);
}
UpdateClosingRtpDataChannels(existing_channels, true);
}
void PeerConnection::UpdateRemoteRtpDataChannels(
const cricket::StreamParamsVec& streams) {
std::vector<std::string> existing_channels;
// Find new and active data channels.
for (const cricket::StreamParams& params : streams) {
// The data channel label is either the mslabel or the SSRC if the mslabel
// does not exist. Ex a=ssrc:444330170 mslabel:test1.
std::string label = params.sync_label.empty()
? rtc::ToString(params.first_ssrc())
: params.sync_label;
auto data_channel_it = rtp_data_channels_.find(label);
if (data_channel_it == rtp_data_channels_.end()) {
// This is a new data channel.
CreateRemoteRtpDataChannel(label, params.first_ssrc());
} else {
data_channel_it->second->SetReceiveSsrc(params.first_ssrc());
}
existing_channels.push_back(label);
}
UpdateClosingRtpDataChannels(existing_channels, false);
}
void PeerConnection::UpdateClosingRtpDataChannels(
const std::vector<std::string>& active_channels,
bool is_local_update) {
auto it = rtp_data_channels_.begin();
while (it != rtp_data_channels_.end()) {
DataChannel* data_channel = it->second;
if (std::find(active_channels.begin(), active_channels.end(),
data_channel->label()) != active_channels.end()) {
++it;
continue;
}
if (is_local_update) {
data_channel->SetSendSsrc(0);
} else {
data_channel->RemotePeerRequestClose();
}
if (data_channel->state() == DataChannel::kClosed) {
rtp_data_channels_.erase(it);
it = rtp_data_channels_.begin();
} else {
++it;
}
}
}
void PeerConnection::CreateRemoteRtpDataChannel(const std::string& label,
uint32_t remote_ssrc) {
rtc::scoped_refptr<DataChannel> channel(
InternalCreateDataChannel(label, nullptr));
if (!channel.get()) {
RTC_LOG(LS_WARNING) << "Remote peer requested a DataChannel but"
<< "CreateDataChannel failed.";
return;
}
channel->SetReceiveSsrc(remote_ssrc);
rtc::scoped_refptr<DataChannelInterface> proxy_channel =
DataChannelProxy::Create(signaling_thread(), channel);
observer_->OnDataChannel(std::move(proxy_channel));
}
rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
const std::string& label,
const InternalDataChannelInit* config) {
if (IsClosed()) {
return nullptr;
}
if (data_channel_type() == cricket::DCT_NONE) {
RTC_LOG(LS_ERROR)
<< "InternalCreateDataChannel: Data is not supported in this call.";
return nullptr;
}
InternalDataChannelInit new_config =
config ? (*config) : InternalDataChannelInit();
if (data_channel_type() == cricket::DCT_SCTP) {
if (new_config.id < 0) {
rtc::SSLRole role;
if ((GetSctpSslRole(&role)) &&
!sid_allocator_.AllocateSid(role, &new_config.id)) {
RTC_LOG(LS_ERROR)
<< "No id can be allocated for the SCTP data channel.";
return nullptr;
}
} else if (!sid_allocator_.ReserveSid(new_config.id)) {
RTC_LOG(LS_ERROR) << "Failed to create a SCTP data channel "
<< "because the id is already in use or out of range.";
return nullptr;
}
}
rtc::scoped_refptr<DataChannel> channel(
DataChannel::Create(this, data_channel_type(), label, new_config));
if (!channel) {
sid_allocator_.ReleaseSid(new_config.id);
return nullptr;
}
if (channel->data_channel_type() == cricket::DCT_RTP) {
if (rtp_data_channels_.find(channel->label()) != rtp_data_channels_.end()) {
RTC_LOG(LS_ERROR) << "DataChannel with label " << channel->label()
<< " already exists.";
return nullptr;
}
rtp_data_channels_[channel->label()] = channel;
} else {
RTC_DCHECK(channel->data_channel_type() == cricket::DCT_SCTP);
sctp_data_channels_.push_back(channel);
channel->SignalClosed.connect(this,
&PeerConnection::OnSctpDataChannelClosed);
}
SignalDataChannelCreated(channel.get());
return channel;
}
bool PeerConnection::HasDataChannels() const {
return !rtp_data_channels_.empty() || !sctp_data_channels_.empty();
}
void PeerConnection::AllocateSctpSids(rtc::SSLRole role) {
for (const auto& channel : sctp_data_channels_) {
if (channel->id() < 0) {
int sid;
if (!sid_allocator_.AllocateSid(role, &sid)) {
RTC_LOG(LS_ERROR) << "Failed to allocate SCTP sid.";
continue;
}
channel->SetSctpSid(sid);
}
}
}
void PeerConnection::OnSctpDataChannelClosed(DataChannel* channel) {
RTC_DCHECK(signaling_thread()->IsCurrent());
for (auto it = sctp_data_channels_.begin(); it != sctp_data_channels_.end();
++it) {
if (it->get() == channel) {
if (channel->id() >= 0) {
sid_allocator_.ReleaseSid(channel->id());
}
// Since this method is triggered by a signal from the DataChannel,
// we can't free it directly here; we need to free it asynchronously.
sctp_data_channels_to_free_.push_back(*it);
sctp_data_channels_.erase(it);
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FREE_DATACHANNELS,
nullptr);
return;
}
}
}
void PeerConnection::OnDataChannelDestroyed() {
// Use a temporary copy of the RTP/SCTP DataChannel list because the
// DataChannel may callback to us and try to modify the list.
std::map<std::string, rtc::scoped_refptr<DataChannel>> temp_rtp_dcs;
temp_rtp_dcs.swap(rtp_data_channels_);
for (const auto& kv : temp_rtp_dcs) {
kv.second->OnTransportChannelDestroyed();
}
std::vector<rtc::scoped_refptr<DataChannel>> temp_sctp_dcs;
temp_sctp_dcs.swap(sctp_data_channels_);
for (const auto& channel : temp_sctp_dcs) {
channel->OnTransportChannelDestroyed();
}
}
void PeerConnection::OnDataChannelOpenMessage(
const std::string& label,
const InternalDataChannelInit& config) {
rtc::scoped_refptr<DataChannel> channel(
InternalCreateDataChannel(label, &config));
if (!channel.get()) {
RTC_LOG(LS_ERROR) << "Failed to create DataChannel from the OPEN message.";
return;
}
rtc::scoped_refptr<DataChannelInterface> proxy_channel =
DataChannelProxy::Create(signaling_thread(), channel);
observer_->OnDataChannel(std::move(proxy_channel));
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::GetAudioTransceiver() const {
// This method only works with Plan B SDP, where there is a single
// audio/video transceiver.
RTC_DCHECK(!IsUnifiedPlan());
for (auto transceiver : transceivers_) {
if (transceiver->internal()->media_type() == cricket::MEDIA_TYPE_AUDIO) {
return transceiver;
}
}
RTC_NOTREACHED();
return nullptr;
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::GetVideoTransceiver() const {
// This method only works with Plan B SDP, where there is a single
// audio/video transceiver.
RTC_DCHECK(!IsUnifiedPlan());
for (auto transceiver : transceivers_) {
if (transceiver->internal()->media_type() == cricket::MEDIA_TYPE_VIDEO) {
return transceiver;
}
}
RTC_NOTREACHED();
return nullptr;
}
// TODO(bugs.webrtc.org/7600): Remove this when multiple transceivers with
// individual transceiver directions are supported.
bool PeerConnection::HasRtpSender(cricket::MediaType type) const {
switch (type) {
case cricket::MEDIA_TYPE_AUDIO:
return !GetAudioTransceiver()->internal()->senders().empty();
case cricket::MEDIA_TYPE_VIDEO:
return !GetVideoTransceiver()->internal()->senders().empty();
case cricket::MEDIA_TYPE_DATA:
return false;
}
RTC_NOTREACHED();
return false;
}
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) const {
for (auto transceiver : transceivers_) {
for (auto sender : transceiver->internal()->senders()) {
if (sender->track() == track) {
return sender;
}
}
}
return nullptr;
}
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
PeerConnection::FindSenderById(const std::string& sender_id) const {
for (auto transceiver : transceivers_) {
for (auto sender : transceiver->internal()->senders()) {
if (sender->id() == sender_id) {
return sender;
}
}
}
return nullptr;
}
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
PeerConnection::FindReceiverById(const std::string& receiver_id) const {
for (auto transceiver : transceivers_) {
for (auto receiver : transceiver->internal()->receivers()) {
if (receiver->id() == receiver_id) {
return receiver;
}
}
}
return nullptr;
}
std::vector<PeerConnection::RtpSenderInfo>*
PeerConnection::GetRemoteSenderInfos(cricket::MediaType media_type) {
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO);
return (media_type == cricket::MEDIA_TYPE_AUDIO)
? &remote_audio_sender_infos_
: &remote_video_sender_infos_;
}
std::vector<PeerConnection::RtpSenderInfo>* PeerConnection::GetLocalSenderInfos(
cricket::MediaType media_type) {
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO);
return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &local_audio_sender_infos_
: &local_video_sender_infos_;
}
const PeerConnection::RtpSenderInfo* PeerConnection::FindSenderInfo(
const std::vector<PeerConnection::RtpSenderInfo>& infos,
const std::string& stream_label,
const std::string sender_id) const {
for (const RtpSenderInfo& sender_info : infos) {
if (sender_info.stream_label == stream_label &&
sender_info.sender_id == sender_id) {
return &sender_info;
}
}
return nullptr;
}
DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
for (const auto& channel : sctp_data_channels_) {
if (channel->id() == sid) {
return channel;
}
}
return nullptr;
}
bool PeerConnection::InitializePortAllocator_n(
const RTCConfiguration& configuration) {
cricket::ServerAddresses stun_servers;
std::vector<cricket::RelayServerConfig> turn_servers;
if (ParseIceServers(configuration.servers, &stun_servers, &turn_servers) !=
RTCErrorType::NONE) {
return false;
}
port_allocator_->Initialize();
// To handle both internal and externally created port allocator, we will
// enable BUNDLE here.
int portallocator_flags = port_allocator_->flags();
portallocator_flags |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
cricket::PORTALLOCATOR_ENABLE_IPV6 |
cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI;
// If the disable-IPv6 flag was specified, we'll not override it
// by experiment.
if (configuration.disable_ipv6) {
portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
} else if (webrtc::field_trial::FindFullName("WebRTC-IPv6Default")
.find("Disabled") == 0) {
portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
}
if (configuration.disable_ipv6_on_wifi) {
portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
RTC_LOG(LS_INFO) << "IPv6 candidates on Wi-Fi are disabled.";
}
if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
RTC_LOG(LS_INFO) << "TCP candidates are disabled.";
}
if (configuration.candidate_network_policy ==
kCandidateNetworkPolicyLowCost) {
portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS;
RTC_LOG(LS_INFO) << "Do not gather candidates on high-cost networks";
}
port_allocator_->set_flags(portallocator_flags);
// No step delay is used while allocating ports.
port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
port_allocator_->set_candidate_filter(
ConvertIceTransportTypeToCandidateFilter(configuration.type));
port_allocator_->set_max_ipv6_networks(configuration.max_ipv6_networks);
// Call this last since it may create pooled allocator sessions using the
// properties set above.
port_allocator_->SetConfiguration(stun_servers, turn_servers,
configuration.ice_candidate_pool_size,
configuration.prune_turn_ports,
configuration.turn_customizer);
return true;
}
bool PeerConnection::ReconfigurePortAllocator_n(
const cricket::ServerAddresses& stun_servers,
const std::vector<cricket::RelayServerConfig>& turn_servers,
IceTransportsType type,
int candidate_pool_size,
bool prune_turn_ports,
webrtc::TurnCustomizer* turn_customizer) {
port_allocator_->set_candidate_filter(
ConvertIceTransportTypeToCandidateFilter(type));
// Call this last since it may create pooled allocator sessions using the
// candidate filter set above.
return port_allocator_->SetConfiguration(
stun_servers, turn_servers, candidate_pool_size, prune_turn_ports,
turn_customizer);
}
cricket::ChannelManager* PeerConnection::channel_manager() const {
return factory_->channel_manager();
}
MetricsObserverInterface* PeerConnection::metrics_observer() const {
return uma_observer_;
}
bool PeerConnection::StartRtcEventLog_w(
std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) {
if (!event_log_) {
return false;
}
return event_log_->StartLogging(std::move(output), output_period_ms);
}
void PeerConnection::StopRtcEventLog_w() {
if (event_log_) {
event_log_->StopLogging();
}
}
cricket::BaseChannel* PeerConnection::GetChannel(
const std::string& content_name) {
if (voice_channel() && voice_channel()->content_name() == content_name) {
return voice_channel();
}
if (video_channel() && video_channel()->content_name() == content_name) {
return video_channel();
}
if (rtp_data_channel() &&
rtp_data_channel()->content_name() == content_name) {
return rtp_data_channel();
}
return nullptr;
}
bool PeerConnection::GetSctpSslRole(rtc::SSLRole* role) {
if (!local_description() || !remote_description()) {
RTC_LOG(LS_INFO)
<< "Local and Remote descriptions must be applied to get the "
<< "SSL Role of the SCTP transport.";
return false;
}
if (!sctp_transport_) {
RTC_LOG(LS_INFO) << "Non-rejected SCTP m= section is needed to get the "
<< "SSL Role of the SCTP transport.";
return false;
}
return transport_controller_->GetSslRole(*sctp_transport_name_, role);
}
bool PeerConnection::GetSslRole(const std::string& content_name,
rtc::SSLRole* role) {
if (!local_description() || !remote_description()) {
RTC_LOG(LS_INFO)
<< "Local and Remote descriptions must be applied to get the "
<< "SSL Role of the session.";
return false;
}
return transport_controller_->GetSslRole(GetTransportName(content_name),
role);
}
// TODO(steveanton): Eventually it'd be nice to store the channels as a single
// vector of BaseChannel pointers instead of separate voice and video channel
// vectors. At that point, this will become a simple getter.
std::vector<cricket::BaseChannel*> PeerConnection::Channels() const {
std::vector<cricket::BaseChannel*> channels;
if (voice_channel()) {
channels.push_back(voice_channel());
}
if (video_channel()) {
channels.push_back(video_channel());
}
if (rtp_data_channel_) {
channels.push_back(rtp_data_channel_);
}
return channels;
}
void PeerConnection::SetSessionError(SessionError error,
const std::string& error_desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (error != session_error_) {
session_error_ = error;
session_error_desc_ = error_desc;
}
}
RTCError PeerConnection::UpdateSessionState(SdpType type,
cricket::ContentSource source) {
RTC_DCHECK_RUN_ON(signaling_thread());
// If there's already a pending error then no state transition should happen.
// But all call-sites should be verifying this before calling us!
RTC_DCHECK(session_error() == SessionError::kNone);
// If this is an answer then we know whether to BUNDLE or not. If both the
// local and remote side have agreed to BUNDLE, go ahead and enable it.
if (type == SdpType::kAnswer) {
const cricket::ContentGroup* local_bundle =
local_description()->description()->GetGroupByName(
cricket::GROUP_TYPE_BUNDLE);
const cricket::ContentGroup* remote_bundle =
remote_description()->description()->GetGroupByName(
cricket::GROUP_TYPE_BUNDLE);
if (local_bundle && remote_bundle) {
// The answerer decides the transport to bundle on.
const cricket::ContentGroup* answer_bundle =
(source == cricket::CS_LOCAL ? local_bundle : remote_bundle);
if (!EnableBundle(*answer_bundle)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
kEnableBundleFailed);
}
}
}
// Only push down the transport description after potentially enabling BUNDLE;
// we don't want to push down a description on a transport about to be
// destroyed.
RTCError error = PushdownTransportDescription(source, type);
if (!error.ok()) {
return error;
}
// If this is answer-ish we're ready to let media flow.
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
EnableSending();
}
// Update the signaling state according to the specified state machine (see
// https://w3c.github.io/webrtc-pc/#rtcsignalingstate-enum).
if (type == SdpType::kOffer) {
ChangeSignalingState(source == cricket::CS_LOCAL
? PeerConnectionInterface::kHaveLocalOffer
: PeerConnectionInterface::kHaveRemoteOffer);
} else if (type == SdpType::kPrAnswer) {
ChangeSignalingState(source == cricket::CS_LOCAL
? PeerConnectionInterface::kHaveLocalPrAnswer
: PeerConnectionInterface::kHaveRemotePrAnswer);
} else {
RTC_DCHECK(type == SdpType::kAnswer);
ChangeSignalingState(PeerConnectionInterface::kStable);
}
// Update internal objects according to the session description's media
// descriptions.
error = PushdownMediaDescription(type, source);
if (!error.ok()) {
SetSessionError(SessionError::kContent, error.message());
}
if (session_error() != SessionError::kNone) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
}
return RTCError::OK();
}
RTCError PeerConnection::PushdownMediaDescription(
SdpType type,
cricket::ContentSource source) {
const SessionDescriptionInterface* sdesc =
(source == cricket::CS_LOCAL ? local_description()
: remote_description());
RTC_DCHECK(sdesc);
// Push down the new SDP media section for each audio/video transceiver.
for (auto transceiver : transceivers_) {
const ContentInfo* content_info =
FindMediaSectionForTransceiver(transceiver, sdesc);
cricket::BaseChannel* channel = transceiver->internal()->channel();
if (!channel || !content_info || content_info->rejected) {
continue;
}
const MediaContentDescription* content_desc =
static_cast<const MediaContentDescription*>(content_info->description);
if (!content_desc) {
continue;
}
std::string error;
bool success =
(source == cricket::CS_LOCAL)
? channel->SetLocalContent(content_desc, type, &error)
: channel->SetRemoteContent(content_desc, type, &error);
if (!success) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, std::move(error));
}
}
// If using the RtpDataChannel, push down the new SDP section for it too.
if (rtp_data_channel_) {
const ContentInfo* data_content =
cricket::GetFirstDataContent(sdesc->description());
if (data_content && !data_content->rejected) {
const MediaContentDescription* data_desc =
static_cast<const MediaContentDescription*>(
data_content->description);
if (data_desc) {
std::string error;
bool success =
(source == cricket::CS_LOCAL)
? rtp_data_channel_->SetLocalContent(data_desc, type, &error)
: rtp_data_channel_->SetRemoteContent(data_desc, type,
&error);
if (!success) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
std::move(error));
}
}
}
}
// Need complete offer/answer with an SCTP m= section before starting SCTP,
// according to https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-19
if (sctp_transport_ && local_description() && remote_description() &&
cricket::GetFirstDataContent(local_description()->description()) &&
cricket::GetFirstDataContent(remote_description()->description())) {
bool success = network_thread()->Invoke<bool>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::PushdownSctpParameters_n, this, source));
if (!success) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to push down SCTP parameters.");
}
}
return RTCError::OK();
}
bool PeerConnection::PushdownSctpParameters_n(cricket::ContentSource source) {
RTC_DCHECK(network_thread()->IsCurrent());
RTC_DCHECK(local_description());
RTC_DCHECK(remote_description());
// Apply the SCTP port (which is hidden inside a DataCodec structure...)
// When we support "max-message-size", that would also be pushed down here.
return sctp_transport_->Start(
GetSctpPort(local_description()->description()),
GetSctpPort(remote_description()->description()));
}
RTCError PeerConnection::PushdownTransportDescription(
cricket::ContentSource source,
SdpType type) {
RTC_DCHECK_RUN_ON(signaling_thread());
const SessionDescriptionInterface* sdesc =
(source == cricket::CS_LOCAL ? local_description()
: remote_description());
RTC_DCHECK(sdesc);
for (const cricket::TransportInfo& tinfo :
sdesc->description()->transport_infos()) {
std::string error;
bool success;
if (source == cricket::CS_LOCAL) {
success = transport_controller_->SetLocalTransportDescription(
tinfo.content_name, tinfo.description, type, &error);
} else {
success = transport_controller_->SetRemoteTransportDescription(
tinfo.content_name, tinfo.description, type, &error);
}
if (!success) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Failed to push down transport description: " + error);
}
}
return RTCError::OK();
}
bool PeerConnection::GetTransportDescription(
const SessionDescription* description,
const std::string& content_name,
cricket::TransportDescription* tdesc) {
if (!description || !tdesc) {
return false;
}
const TransportInfo* transport_info =
description->GetTransportInfoByName(content_name);
if (!transport_info) {
return false;
}
*tdesc = transport_info->description;
return true;
}
bool PeerConnection::EnableBundle(const cricket::ContentGroup& bundle) {
const std::string* first_content_name = bundle.FirstContentName();
if (!first_content_name) {
RTC_LOG(LS_WARNING) << "Tried to BUNDLE with no contents.";
return false;
}
const std::string& transport_name = *first_content_name;
auto maybe_set_transport = [this, bundle,
transport_name](cricket::BaseChannel* ch) {
if (!ch || !bundle.HasContentName(ch->content_name())) {
return;
}
std::string old_transport_name = ch->transport_name();
if (old_transport_name == transport_name) {
RTC_LOG(LS_INFO) << "BUNDLE already enabled for " << ch->content_name()
<< " on " << transport_name << ".";
return;
}
cricket::DtlsTransportInternal* rtp_dtls_transport =
transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
bool need_rtcp = (ch->rtcp_dtls_transport() != nullptr);
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
if (need_rtcp) {
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
ch->SetTransports(rtp_dtls_transport, rtcp_dtls_transport);
RTC_LOG(LS_INFO) << "Enabled BUNDLE for " << ch->content_name() << " on "
<< transport_name << ".";
transport_controller_->DestroyDtlsTransport(
old_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
// If the channel needs rtcp, it means that the channel used to have a
// rtcp transport which needs to be deleted now.
if (need_rtcp) {
transport_controller_->DestroyDtlsTransport(
old_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
};
for (auto transceiver : transceivers_) {
maybe_set_transport(transceiver->internal()->channel());
}
maybe_set_transport(rtp_data_channel_);
// For SCTP, transport creation/deletion happens here instead of in the
// object itself.
if (sctp_transport_) {
RTC_DCHECK(sctp_transport_name_);
RTC_DCHECK(sctp_content_name_);
if (transport_name != *sctp_transport_name_ &&
bundle.HasContentName(*sctp_content_name_)) {
network_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::ChangeSctpTransport_n, this,
transport_name));
}
}
return true;
}
cricket::IceConfig PeerConnection::ParseIceConfig(
const PeerConnectionInterface::RTCConfiguration& config) const {
cricket::ContinualGatheringPolicy gathering_policy;
// TODO(honghaiz): Add the third continual gathering policy in
// PeerConnectionInterface and map it to GATHER_CONTINUALLY_AND_RECOVER.
switch (config.continual_gathering_policy) {
case PeerConnectionInterface::GATHER_ONCE:
gathering_policy = cricket::GATHER_ONCE;
break;
case PeerConnectionInterface::GATHER_CONTINUALLY:
gathering_policy = cricket::GATHER_CONTINUALLY;
break;
default:
RTC_NOTREACHED();
gathering_policy = cricket::GATHER_ONCE;
}
cricket::IceConfig ice_config;
ice_config.receiving_timeout = config.ice_connection_receiving_timeout;
ice_config.prioritize_most_likely_candidate_pairs =
config.prioritize_most_likely_ice_candidate_pairs;
ice_config.backup_connection_ping_interval =
config.ice_backup_candidate_pair_ping_interval;
ice_config.continual_gathering_policy = gathering_policy;
ice_config.presume_writable_when_fully_relayed =
config.presume_writable_when_fully_relayed;
ice_config.ice_check_min_interval = config.ice_check_min_interval;
ice_config.regather_all_networks_interval_range =
config.ice_regather_interval_range;
return ice_config;
}
bool PeerConnection::GetLocalTrackIdBySsrc(uint32_t ssrc,
std::string* track_id) {
if (!local_description()) {
return false;
}
return webrtc::GetTrackIdBySsrc(local_description()->description(), ssrc,
track_id);
}
bool PeerConnection::GetRemoteTrackIdBySsrc(uint32_t ssrc,
std::string* track_id) {
if (!remote_description()) {
return false;
}
return webrtc::GetTrackIdBySsrc(remote_description()->description(), ssrc,
track_id);
}
bool PeerConnection::SendData(const cricket::SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
cricket::SendDataResult* result) {
if (!rtp_data_channel_ && !sctp_transport_) {
RTC_LOG(LS_ERROR) << "SendData called when rtp_data_channel_ "
<< "and sctp_transport_ are NULL.";
return false;
}
return rtp_data_channel_
? rtp_data_channel_->SendData(params, payload, result)
: network_thread()->Invoke<bool>(
RTC_FROM_HERE,
Bind(&cricket::SctpTransportInternal::SendData,
sctp_transport_.get(), params, payload, result));
}
bool PeerConnection::ConnectDataChannel(DataChannel* webrtc_data_channel) {
if (!rtp_data_channel_ && !sctp_transport_) {
// Don't log an error here, because DataChannels are expected to call
// ConnectDataChannel in this state. It's the only way to initially tell
// whether or not the underlying transport is ready.
return false;
}
if (rtp_data_channel_) {
rtp_data_channel_->SignalReadyToSendData.connect(
webrtc_data_channel, &DataChannel::OnChannelReady);
rtp_data_channel_->SignalDataReceived.connect(webrtc_data_channel,
&DataChannel::OnDataReceived);
} else {
SignalSctpReadyToSendData.connect(webrtc_data_channel,
&DataChannel::OnChannelReady);
SignalSctpDataReceived.connect(webrtc_data_channel,
&DataChannel::OnDataReceived);
SignalSctpStreamClosedRemotely.connect(
webrtc_data_channel, &DataChannel::OnStreamClosedRemotely);
}
return true;
}
void PeerConnection::DisconnectDataChannel(DataChannel* webrtc_data_channel) {
if (!rtp_data_channel_ && !sctp_transport_) {
RTC_LOG(LS_ERROR)
<< "DisconnectDataChannel called when rtp_data_channel_ and "
"sctp_transport_ are NULL.";
return;
}
if (rtp_data_channel_) {
rtp_data_channel_->SignalReadyToSendData.disconnect(webrtc_data_channel);
rtp_data_channel_->SignalDataReceived.disconnect(webrtc_data_channel);
} else {
SignalSctpReadyToSendData.disconnect(webrtc_data_channel);
SignalSctpDataReceived.disconnect(webrtc_data_channel);
SignalSctpStreamClosedRemotely.disconnect(webrtc_data_channel);
}
}
void PeerConnection::AddSctpDataStream(int sid) {
if (!sctp_transport_) {
RTC_LOG(LS_ERROR)
<< "AddSctpDataStream called when sctp_transport_ is NULL.";
return;
}
network_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::OpenStream,
sctp_transport_.get(), sid));
}
void PeerConnection::RemoveSctpDataStream(int sid) {
if (!sctp_transport_) {
RTC_LOG(LS_ERROR) << "RemoveSctpDataStream called when sctp_transport_ is "
<< "NULL.";
return;
}
network_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::ResetStream,
sctp_transport_.get(), sid));
}
bool PeerConnection::ReadyToSendData() const {
return (rtp_data_channel_ && rtp_data_channel_->ready_to_send_data()) ||
sctp_ready_to_send_data_;
}
std::unique_ptr<SessionStats> PeerConnection::GetSessionStats_s() {
RTC_DCHECK(signaling_thread()->IsCurrent());
ChannelNamePairs channel_name_pairs;
if (voice_channel()) {
channel_name_pairs.voice = ChannelNamePair(
voice_channel()->content_name(), voice_channel()->transport_name());
}
if (video_channel()) {
channel_name_pairs.video = ChannelNamePair(
video_channel()->content_name(), video_channel()->transport_name());
}
if (rtp_data_channel()) {
channel_name_pairs.data =
ChannelNamePair(rtp_data_channel()->content_name(),
rtp_data_channel()->transport_name());
}
if (sctp_transport_) {
RTC_DCHECK(sctp_content_name_);
RTC_DCHECK(sctp_transport_name_);
channel_name_pairs.data =
ChannelNamePair(*sctp_content_name_, *sctp_transport_name_);
}
return GetSessionStats(channel_name_pairs);
}
std::unique_ptr<SessionStats> PeerConnection::GetSessionStats(
const ChannelNamePairs& channel_name_pairs) {
if (network_thread()->IsCurrent()) {
return GetSessionStats_n(channel_name_pairs);
}
return network_thread()->Invoke<std::unique_ptr<SessionStats>>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::GetSessionStats_n, this, channel_name_pairs));
}
bool PeerConnection::GetLocalCertificate(
const std::string& transport_name,
rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
return transport_controller_->GetLocalCertificate(transport_name,
certificate);
}
std::unique_ptr<rtc::SSLCertificate> PeerConnection::GetRemoteSSLCertificate(
const std::string& transport_name) {
return transport_controller_->GetRemoteSSLCertificate(transport_name);
}
cricket::DataChannelType PeerConnection::data_channel_type() const {
return data_channel_type_;
}
bool PeerConnection::IceRestartPending(const std::string& content_name) const {
return pending_ice_restarts_.find(content_name) !=
pending_ice_restarts_.end();
}
bool PeerConnection::NeedsIceRestart(const std::string& content_name) const {
return transport_controller_->NeedsIceRestart(content_name);
}
void PeerConnection::OnCertificateReady(
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
transport_controller_->SetLocalCertificate(certificate);
}
void PeerConnection::OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp) {
SetSessionError(SessionError::kTransport,
rtcp ? kDtlsSrtpSetupFailureRtcp : kDtlsSrtpSetupFailureRtp);
}
void PeerConnection::OnTransportControllerConnectionState(
cricket::IceConnectionState state) {
switch (state) {
case cricket::kIceConnectionConnecting:
// If the current state is Connected or Completed, then there were
// writable channels but now there are not, so the next state must
// be Disconnected.
// kIceConnectionConnecting is currently used as the default,
// un-connected state by the TransportController, so its only use is
// detecting disconnections.
if (ice_connection_state_ ==
PeerConnectionInterface::kIceConnectionConnected ||
ice_connection_state_ ==
PeerConnectionInterface::kIceConnectionCompleted) {
SetIceConnectionState(
PeerConnectionInterface::kIceConnectionDisconnected);
}
break;
case cricket::kIceConnectionFailed:
SetIceConnectionState(PeerConnectionInterface::kIceConnectionFailed);
break;
case cricket::kIceConnectionConnected:
RTC_LOG(LS_INFO) << "Changing to ICE connected state because "
<< "all transports are writable.";
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
break;
case cricket::kIceConnectionCompleted:
RTC_LOG(LS_INFO) << "Changing to ICE completed state because "
<< "all transports are complete.";
if (ice_connection_state_ !=
PeerConnectionInterface::kIceConnectionConnected) {
// If jumping directly from "checking" to "connected",
// signal "connected" first.
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
}
SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
if (metrics_observer()) {
ReportTransportStats();
}
break;
default:
RTC_NOTREACHED();
}
}
void PeerConnection::OnTransportControllerCandidatesGathered(
const std::string& transport_name,
const cricket::Candidates& candidates) {
RTC_DCHECK(signaling_thread()->IsCurrent());
int sdp_mline_index;
if (!GetLocalCandidateMediaIndex(transport_name, &sdp_mline_index)) {
RTC_LOG(LS_ERROR)
<< "OnTransportControllerCandidatesGathered: content name "
<< transport_name << " not found";
return;
}
for (cricket::Candidates::const_iterator citer = candidates.begin();
citer != candidates.end(); ++citer) {
// Use transport_name as the candidate media id.
std::unique_ptr<JsepIceCandidate> candidate(
new JsepIceCandidate(transport_name, sdp_mline_index, *citer));
if (local_description()) {
mutable_local_description()->AddCandidate(candidate.get());
}
OnIceCandidate(std::move(candidate));
}
}
void PeerConnection::OnTransportControllerCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {
RTC_DCHECK(signaling_thread()->IsCurrent());
// Sanity check.
for (const cricket::Candidate& candidate : candidates) {
if (candidate.transport_name().empty()) {
RTC_LOG(LS_ERROR) << "OnTransportControllerCandidatesRemoved: "
<< "empty content name in candidate "
<< candidate.ToString();
return;
}
}
if (local_description()) {
mutable_local_description()->RemoveCandidates(candidates);
}
OnIceCandidatesRemoved(candidates);
}
void PeerConnection::OnTransportControllerDtlsHandshakeError(
rtc::SSLHandshakeError error) {
if (metrics_observer()) {
metrics_observer()->IncrementEnumCounter(
webrtc::kEnumCounterDtlsHandshakeError, static_cast<int>(error),
static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
}
}
void PeerConnection::EnableSending() {
for (auto transceiver : transceivers_) {
cricket::BaseChannel* channel = transceiver->internal()->channel();
if (channel && !channel->enabled()) {
channel->Enable(true);
}
}
if (rtp_data_channel_ && !rtp_data_channel_->enabled()) {
rtp_data_channel_->Enable(true);
}
}
// Returns the media index for a local ice candidate given the content name.
bool PeerConnection::GetLocalCandidateMediaIndex(
const std::string& content_name,
int* sdp_mline_index) {
if (!local_description() || !sdp_mline_index) {
return false;
}
bool content_found = false;
const ContentInfos& contents = local_description()->description()->contents();
for (size_t index = 0; index < contents.size(); ++index) {
if (contents[index].name == content_name) {
*sdp_mline_index = static_cast<int>(index);
content_found = true;
break;
}
}
return content_found;
}
bool PeerConnection::UseCandidatesInSessionDescription(
const SessionDescriptionInterface* remote_desc) {
if (!remote_desc) {
return true;
}
bool ret = true;
for (size_t m = 0; m < remote_desc->number_of_mediasections(); ++m) {
const IceCandidateCollection* candidates = remote_desc->candidates(m);
for (size_t n = 0; n < candidates->count(); ++n) {
const IceCandidateInterface* candidate = candidates->at(n);
bool valid = false;
if (!ReadyToUseRemoteCandidate(candidate, remote_desc, &valid)) {
if (valid) {
RTC_LOG(LS_INFO)
<< "UseCandidatesInSessionDescription: Not ready to use "
<< "candidate.";
}
continue;
}
ret = UseCandidate(candidate);
if (!ret) {
break;
}
}
}
return ret;
}
bool PeerConnection::UseCandidate(const IceCandidateInterface* candidate) {
size_t mediacontent_index = static_cast<size_t>(candidate->sdp_mline_index());
size_t remote_content_size =
remote_description()->description()->contents().size();
if (mediacontent_index >= remote_content_size) {
RTC_LOG(LS_ERROR) << "UseCandidate: Invalid candidate media index.";
return false;
}
cricket::ContentInfo content =
remote_description()->description()->contents()[mediacontent_index];
std::vector<cricket::Candidate> candidates;
candidates.push_back(candidate->candidate());
// Invoking BaseSession method to handle remote candidates.
std::string error;
if (transport_controller_->AddRemoteCandidates(content.name, candidates,
&error)) {
// Candidates successfully submitted for checking.
if (ice_connection_state_ == PeerConnectionInterface::kIceConnectionNew ||
ice_connection_state_ ==
PeerConnectionInterface::kIceConnectionDisconnected) {
// If state is New, then the session has just gotten its first remote ICE
// candidates, so go to Checking.
// If state is Disconnected, the session is re-using old candidates or
// receiving additional ones, so go to Checking.
// If state is Connected, stay Connected.
// TODO(bemasc): If state is Connected, and the new candidates are for a
// newly added transport, then the state actually _should_ move to
// checking. Add a way to distinguish that case.
SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
}
// TODO(bemasc): If state is Completed, go back to Connected.
} else {
if (!error.empty()) {
RTC_LOG(LS_WARNING) << error;
}
}
return true;
}
void PeerConnection::RemoveUnusedChannels(const SessionDescription* desc) {
// Destroy video channel first since it may have a pointer to the
// voice channel.
const cricket::ContentInfo* video_info = cricket::GetFirstVideoContent(desc);
if (!video_info || video_info->rejected) {
DestroyTransceiverChannel(GetVideoTransceiver());
}
const cricket::ContentInfo* audio_info = cricket::GetFirstAudioContent(desc);
if (!audio_info || audio_info->rejected) {
DestroyTransceiverChannel(GetAudioTransceiver());
}
const cricket::ContentInfo* data_info = cricket::GetFirstDataContent(desc);
if (!data_info || data_info->rejected) {
DestroyDataChannel();
}
}
std::string PeerConnection::GetTransportNameForMediaSection(
const std::string& mid,
const cricket::ContentGroup* bundle_group) const {
if (!bundle_group) {
return mid;
}
const std::string* first_content_name = bundle_group->FirstContentName();
if (!first_content_name) {
RTC_LOG(LS_WARNING) << "Tried to BUNDLE with no contents.";
return mid;
}
if (!bundle_group->HasContentName(mid)) {
RTC_LOG(LS_WARNING) << mid << " is not part of any bundle group";
return mid;
}
RTC_LOG(LS_INFO) << "Bundling " << mid << " on " << *first_content_name;
return *first_content_name;
}
RTCError PeerConnection::CreateChannels(const SessionDescription* desc) {
RTC_DCHECK(desc);
const cricket::ContentGroup* bundle_group = nullptr;
if (configuration_.bundle_policy ==
PeerConnectionInterface::kBundlePolicyMaxBundle) {
bundle_group = desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
if (!bundle_group) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max-bundle configured but session description "
"has no BUNDLE group");
}
}
// Creating the media channels and transport proxies.
const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(desc);
if (voice && !voice->rejected &&
!GetAudioTransceiver()->internal()->channel()) {
cricket::VoiceChannel* voice_channel = CreateVoiceChannel(
voice->name,
GetTransportNameForMediaSection(voice->name, bundle_group));
if (!voice_channel) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create voice channel.");
}
GetAudioTransceiver()->internal()->SetChannel(voice_channel);
}
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(desc);
if (video && !video->rejected &&
!GetVideoTransceiver()->internal()->channel()) {
cricket::VideoChannel* video_channel = CreateVideoChannel(
video->name,
GetTransportNameForMediaSection(video->name, bundle_group));
if (!video_channel) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create video channel.");
}
GetVideoTransceiver()->internal()->SetChannel(video_channel);
}
const cricket::ContentInfo* data = cricket::GetFirstDataContent(desc);
if (data_channel_type_ != cricket::DCT_NONE && data && !data->rejected &&
!rtp_data_channel_ && !sctp_transport_) {
if (!CreateDataChannel(data->name, GetTransportNameForMediaSection(
data->name, bundle_group))) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create data channel.");
}
}
return RTCError::OK();
}
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
cricket::VoiceChannel* PeerConnection::CreateVoiceChannel(
const std::string& mid,
const std::string& transport_name) {
cricket::DtlsTransportInternal* rtp_dtls_transport =
transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
if (configuration_.rtcp_mux_policy !=
PeerConnectionInterface::kRtcpMuxPolicyRequire) {
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
cricket::VoiceChannel* voice_channel = channel_manager()->CreateVoiceChannel(
call_.get(), configuration_.media_config, rtp_dtls_transport,
rtcp_dtls_transport, signaling_thread(), mid, SrtpRequired(),
audio_options_);
if (!voice_channel) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
if (rtcp_dtls_transport) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
return nullptr;
}
voice_channel->SignalRtcpMuxFullyActive.connect(
this, &PeerConnection::DestroyRtcpTransport_n);
voice_channel->SignalDtlsSrtpSetupFailure.connect(
this, &PeerConnection::OnDtlsSrtpSetupFailure);
voice_channel->SignalSentPacket.connect(this,
&PeerConnection::OnSentPacket_w);
return voice_channel;
}
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
cricket::VideoChannel* PeerConnection::CreateVideoChannel(
const std::string& mid,
const std::string& transport_name) {
cricket::DtlsTransportInternal* rtp_dtls_transport =
transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
if (configuration_.rtcp_mux_policy !=
PeerConnectionInterface::kRtcpMuxPolicyRequire) {
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel(
call_.get(), configuration_.media_config, rtp_dtls_transport,
rtcp_dtls_transport, signaling_thread(), mid, SrtpRequired(),
video_options_);
if (!video_channel) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
if (rtcp_dtls_transport) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
return nullptr;
}
video_channel->SignalRtcpMuxFullyActive.connect(
this, &PeerConnection::DestroyRtcpTransport_n);
video_channel->SignalDtlsSrtpSetupFailure.connect(
this, &PeerConnection::OnDtlsSrtpSetupFailure);
video_channel->SignalSentPacket.connect(this,
&PeerConnection::OnSentPacket_w);
return video_channel;
}
bool PeerConnection::CreateDataChannel(const std::string& mid,
const std::string& transport_name) {
bool sctp = (data_channel_type_ == cricket::DCT_SCTP);
if (sctp) {
if (!sctp_factory_) {
RTC_LOG(LS_ERROR)
<< "Trying to create SCTP transport, but didn't compile with "
"SCTP support (HAVE_SCTP)";
return false;
}
if (!network_thread()->Invoke<bool>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::CreateSctpTransport_n,
this, mid, transport_name))) {
return false;
}
for (const auto& channel : sctp_data_channels_) {
channel->OnTransportChannelCreated();
}
} else {
cricket::DtlsTransportInternal* rtp_dtls_transport =
transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
if (configuration_.rtcp_mux_policy !=
PeerConnectionInterface::kRtcpMuxPolicyRequire) {
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
rtp_data_channel_ = channel_manager()->CreateRtpDataChannel(
configuration_.media_config, rtp_dtls_transport, rtcp_dtls_transport,
signaling_thread(), mid, SrtpRequired());
if (!rtp_data_channel_) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
if (rtcp_dtls_transport) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
return false;
}
rtp_data_channel_->SignalRtcpMuxFullyActive.connect(
this, &PeerConnection::DestroyRtcpTransport_n);
rtp_data_channel_->SignalDtlsSrtpSetupFailure.connect(
this, &PeerConnection::OnDtlsSrtpSetupFailure);
rtp_data_channel_->SignalSentPacket.connect(
this, &PeerConnection::OnSentPacket_w);
}
return true;
}
Call::Stats PeerConnection::GetCallStats() {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->Invoke<Call::Stats>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::GetCallStats, this));
}
if (call_) {
return call_->GetStats();
} else {
return Call::Stats();
}
}
std::unique_ptr<SessionStats> PeerConnection::GetSessionStats_n(
const ChannelNamePairs& channel_name_pairs) {
RTC_DCHECK(network_thread()->IsCurrent());
std::unique_ptr<SessionStats> session_stats(new SessionStats());
for (const auto channel_name_pair :
{&channel_name_pairs.voice, &channel_name_pairs.video,
&channel_name_pairs.data}) {
if (*channel_name_pair) {
cricket::TransportStats transport_stats;
if (!transport_controller_->GetStats((*channel_name_pair)->transport_name,
&transport_stats)) {
return nullptr;
}
session_stats->proxy_to_transport[(*channel_name_pair)->content_name] =
(*channel_name_pair)->transport_name;
session_stats->transport_stats[(*channel_name_pair)->transport_name] =
std::move(transport_stats);
}
}
return session_stats;
}
bool PeerConnection::CreateSctpTransport_n(const std::string& content_name,
const std::string& transport_name) {
RTC_DCHECK(network_thread()->IsCurrent());
RTC_DCHECK(sctp_factory_);
cricket::DtlsTransportInternal* tc =
transport_controller_->CreateDtlsTransport_n(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
sctp_transport_ = sctp_factory_->CreateSctpTransport(tc);
RTC_DCHECK(sctp_transport_);
sctp_invoker_.reset(new rtc::AsyncInvoker());
sctp_transport_->SignalReadyToSendData.connect(
this, &PeerConnection::OnSctpTransportReadyToSendData_n);
sctp_transport_->SignalDataReceived.connect(
this, &PeerConnection::OnSctpTransportDataReceived_n);
sctp_transport_->SignalStreamClosedRemotely.connect(
this, &PeerConnection::OnSctpStreamClosedRemotely_n);
sctp_transport_name_ = transport_name;
sctp_content_name_ = content_name;
return true;
}
void PeerConnection::ChangeSctpTransport_n(const std::string& transport_name) {
RTC_DCHECK(network_thread()->IsCurrent());
RTC_DCHECK(sctp_transport_);
RTC_DCHECK(sctp_transport_name_);
std::string old_sctp_transport_name = *sctp_transport_name_;
sctp_transport_name_ = transport_name;
cricket::DtlsTransportInternal* tc =
transport_controller_->CreateDtlsTransport_n(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
sctp_transport_->SetTransportChannel(tc);
transport_controller_->DestroyDtlsTransport_n(
old_sctp_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
}
void PeerConnection::DestroySctpTransport_n() {
RTC_DCHECK(network_thread()->IsCurrent());
sctp_transport_.reset(nullptr);
sctp_content_name_.reset();
sctp_transport_name_.reset();
sctp_invoker_.reset(nullptr);
sctp_ready_to_send_data_ = false;
}
void PeerConnection::OnSctpTransportReadyToSendData_n() {
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
RTC_DCHECK(network_thread()->IsCurrent());
// Note: Cannot use rtc::Bind here because it will grab a reference to
// PeerConnection and potentially cause PeerConnection to live longer than
// expected. It is safe not to grab a reference since the sctp_invoker_ will
// be destroyed before PeerConnection is destroyed, and at that point all
// pending tasks will be cleared.
sctp_invoker_->AsyncInvoke<void>(RTC_FROM_HERE, signaling_thread(), [this] {
OnSctpTransportReadyToSendData_s(true);
});
}
void PeerConnection::OnSctpTransportReadyToSendData_s(bool ready) {
RTC_DCHECK(signaling_thread()->IsCurrent());
sctp_ready_to_send_data_ = ready;
SignalSctpReadyToSendData(ready);
}
void PeerConnection::OnSctpTransportDataReceived_n(
const cricket::ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& payload) {
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
RTC_DCHECK(network_thread()->IsCurrent());
// Note: Cannot use rtc::Bind here because it will grab a reference to
// PeerConnection and potentially cause PeerConnection to live longer than
// expected. It is safe not to grab a reference since the sctp_invoker_ will
// be destroyed before PeerConnection is destroyed, and at that point all
// pending tasks will be cleared.
sctp_invoker_->AsyncInvoke<void>(
RTC_FROM_HERE, signaling_thread(), [this, params, payload] {
OnSctpTransportDataReceived_s(params, payload);
});
}
void PeerConnection::OnSctpTransportDataReceived_s(
const cricket::ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& payload) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (params.type == cricket::DMT_CONTROL && IsOpenMessage(payload)) {
// Received OPEN message; parse and signal that a new data channel should
// be created.
std::string label;
InternalDataChannelInit config;
config.id = params.ssrc;
if (!ParseDataChannelOpenMessage(payload, &label, &config)) {
RTC_LOG(LS_WARNING) << "Failed to parse the OPEN message for sid "
<< params.ssrc;
return;
}
config.open_handshake_role = InternalDataChannelInit::kAcker;
OnDataChannelOpenMessage(label, config);
} else {
// Otherwise just forward the signal.
SignalSctpDataReceived(params, payload);
}
}
void PeerConnection::OnSctpStreamClosedRemotely_n(int sid) {
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
RTC_DCHECK(network_thread()->IsCurrent());
sctp_invoker_->AsyncInvoke<void>(
RTC_FROM_HERE, signaling_thread(),
rtc::Bind(&sigslot::signal1<int>::operator(),
&SignalSctpStreamClosedRemotely, sid));
}
// Returns false if bundle is enabled and rtcp_mux is disabled.
bool PeerConnection::ValidateBundleSettings(const SessionDescription* desc) {
bool bundle_enabled = desc->HasGroup(cricket::GROUP_TYPE_BUNDLE);
if (!bundle_enabled)
return true;
const cricket::ContentGroup* bundle_group =
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
RTC_DCHECK(bundle_group != NULL);
const cricket::ContentInfos& contents = desc->contents();
for (cricket::ContentInfos::const_iterator citer = contents.begin();
citer != contents.end(); ++citer) {
const cricket::ContentInfo* content = (&*citer);
RTC_DCHECK(content != NULL);
if (bundle_group->HasContentName(content->name) && !content->rejected &&
content->type == cricket::NS_JINGLE_RTP) {
if (!HasRtcpMuxEnabled(content))
return false;
}
}
// RTCP-MUX is enabled in all the contents.
return true;
}
bool PeerConnection::HasRtcpMuxEnabled(const cricket::ContentInfo* content) {
const cricket::MediaContentDescription* description =
static_cast<cricket::MediaContentDescription*>(content->description);
return description->rtcp_mux();
}
RTCError PeerConnection::ValidateSessionDescription(
const SessionDescriptionInterface* sdesc,
cricket::ContentSource source) {
if (session_error() != SessionError::kNone) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
}
if (!sdesc || !sdesc->description()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
}
SdpType type = sdesc->GetType();
if ((source == cricket::CS_LOCAL && !ExpectSetLocalDescription(type)) ||
(source == cricket::CS_REMOTE && !ExpectSetRemoteDescription(type))) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Called in wrong state: " + GetSignalingStateString(signaling_state()));
}
// Verify crypto settings.
std::string crypto_error;
if (webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED ||
dtls_enabled_) {
RTCError crypto_error = VerifyCrypto(sdesc->description(), dtls_enabled_);
if (!crypto_error.ok()) {
return crypto_error;
}
}
// Verify ice-ufrag and ice-pwd.
if (!VerifyIceUfragPwdPresent(sdesc->description())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
kSdpWithoutIceUfragPwd);
}
if (!ValidateBundleSettings(sdesc->description())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
kBundleWithoutRtcpMux);
}
// TODO(skvlad): When the local rtcp-mux policy is Require, reject any
// m-lines that do not rtcp-mux enabled.
// Verify m-lines in Answer when compared against Offer.
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
const cricket::SessionDescription* offer_desc =
(source == cricket::CS_LOCAL) ? remote_description()->description()
: local_description()->description();
if (!MediaSectionsHaveSameCount(offer_desc, sdesc->description()) ||
!MediaSectionsInSameOrder(offer_desc, sdesc->description())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
kMlineMismatchInAnswer);
}
} else {
const cricket::SessionDescription* current_desc = nullptr;
if (source == cricket::CS_LOCAL && local_description()) {
current_desc = local_description()->description();
} else if (source == cricket::CS_REMOTE && remote_description()) {
current_desc = remote_description()->description();
}
// The re-offers should respect the order of m= sections in current
// description. See RFC3264 Section 8 paragraph 4 for more details.
if (current_desc &&
!MediaSectionsInSameOrder(current_desc, sdesc->description())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
kMlineMismatchInSubsequentOffer);
}
}
return RTCError::OK();
}
bool PeerConnection::ExpectSetLocalDescription(SdpType type) {
PeerConnectionInterface::SignalingState state = signaling_state();
if (type == SdpType::kOffer) {
return (state == PeerConnectionInterface::kStable) ||
(state == PeerConnectionInterface::kHaveLocalOffer);
} else {
RTC_DCHECK(type == SdpType::kPrAnswer || type == SdpType::kAnswer);
return (state == PeerConnectionInterface::kHaveRemoteOffer) ||
(state == PeerConnectionInterface::kHaveLocalPrAnswer);
}
}
bool PeerConnection::ExpectSetRemoteDescription(SdpType type) {
PeerConnectionInterface::SignalingState state = signaling_state();
if (type == SdpType::kOffer) {
return (state == PeerConnectionInterface::kStable) ||
(state == PeerConnectionInterface::kHaveRemoteOffer);
} else {
RTC_DCHECK(type == SdpType::kPrAnswer || type == SdpType::kAnswer);
return (state == PeerConnectionInterface::kHaveLocalOffer) ||
(state == PeerConnectionInterface::kHaveRemotePrAnswer);
}
}
const char* PeerConnection::SessionErrorToString(SessionError error) const {
switch (error) {
case SessionError::kNone:
return "ERROR_NONE";
case SessionError::kContent:
return "ERROR_CONTENT";
case SessionError::kTransport:
return "ERROR_TRANSPORT";
}
RTC_NOTREACHED();
return "";
}
std::string PeerConnection::GetSessionErrorMsg() {
std::ostringstream desc;
desc << kSessionError << SessionErrorToString(session_error()) << ". ";
desc << kSessionErrorDesc << session_error_desc() << ".";
return desc.str();
}
// We need to check the local/remote description for the Transport instead of
// the session, because a new Transport added during renegotiation may have
// them unset while the session has them set from the previous negotiation.
// Not doing so may trigger the auto generation of transport description and
// mess up DTLS identity information, ICE credential, etc.
bool PeerConnection::ReadyToUseRemoteCandidate(
const IceCandidateInterface* candidate,
const SessionDescriptionInterface* remote_desc,
bool* valid) {
*valid = true;
const SessionDescriptionInterface* current_remote_desc =
remote_desc ? remote_desc : remote_description();
if (!current_remote_desc) {
return false;
}
size_t mediacontent_index = static_cast<size_t>(candidate->sdp_mline_index());
size_t remote_content_size =
current_remote_desc->description()->contents().size();
if (mediacontent_index >= remote_content_size) {
RTC_LOG(LS_ERROR)
<< "ReadyToUseRemoteCandidate: Invalid candidate media index "
<< mediacontent_index;
*valid = false;
return false;
}
cricket::ContentInfo content =
current_remote_desc->description()->contents()[mediacontent_index];
const std::string transport_name = GetTransportName(content.name);
if (transport_name.empty()) {
return false;
}
return transport_controller_->ReadyForRemoteCandidates(transport_name);
}
bool PeerConnection::SrtpRequired() const {
return dtls_enabled_ ||
webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED;
}
void PeerConnection::OnTransportControllerGatheringState(
cricket::IceGatheringState state) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (state == cricket::kIceGatheringGathering) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringGathering);
} else if (state == cricket::kIceGatheringComplete) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringComplete);
}
}
void PeerConnection::ReportTransportStats() {
// Use a set so we don't report the same stats twice if two channels share
// a transport.
std::set<std::string> transport_names;
if (voice_channel()) {
transport_names.insert(voice_channel()->transport_name());
}
if (video_channel()) {
transport_names.insert(video_channel()->transport_name());
}
if (rtp_data_channel()) {
transport_names.insert(rtp_data_channel()->transport_name());
}
if (sctp_transport_name_) {
transport_names.insert(*sctp_transport_name_);
}
for (const auto& name : transport_names) {
cricket::TransportStats stats;
if (transport_controller_->GetStats(name, &stats)) {
ReportBestConnectionState(stats);
ReportNegotiatedCiphers(stats);
}
}
}
// Walk through the ConnectionInfos to gather best connection usage
// for IPv4 and IPv6.
void PeerConnection::ReportBestConnectionState(
const cricket::TransportStats& stats) {
RTC_DCHECK(metrics_observer());
for (cricket::TransportChannelStatsList::const_iterator it =
stats.channel_stats.begin();
it != stats.channel_stats.end(); ++it) {
for (cricket::ConnectionInfos::const_iterator it_info =
it->connection_infos.begin();
it_info != it->connection_infos.end(); ++it_info) {
if (!it_info->best_connection) {
continue;
}
PeerConnectionEnumCounterType type = kPeerConnectionEnumCounterMax;
const cricket::Candidate& local = it_info->local_candidate;
const cricket::Candidate& remote = it_info->remote_candidate;
// Increment the counter for IceCandidatePairType.
if (local.protocol() == cricket::TCP_PROTOCOL_NAME ||
(local.type() == RELAY_PORT_TYPE &&
local.relay_protocol() == cricket::TCP_PROTOCOL_NAME)) {
type = kEnumCounterIceCandidatePairTypeTcp;
} else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
type = kEnumCounterIceCandidatePairTypeUdp;
} else {
RTC_CHECK(0);
}
metrics_observer()->IncrementEnumCounter(
type, GetIceCandidatePairCounter(local, remote),
kIceCandidatePairMax);
// Increment the counter for IP type.
if (local.address().family() == AF_INET) {
metrics_observer()->IncrementEnumCounter(
kEnumCounterAddressFamily, kBestConnections_IPv4,
kPeerConnectionAddressFamilyCounter_Max);
} else if (local.address().family() == AF_INET6) {
metrics_observer()->IncrementEnumCounter(
kEnumCounterAddressFamily, kBestConnections_IPv6,
kPeerConnectionAddressFamilyCounter_Max);
} else {
RTC_CHECK(0);
}
return;
}
}
}
void PeerConnection::ReportNegotiatedCiphers(
const cricket::TransportStats& stats) {
RTC_DCHECK(metrics_observer());
if (!dtls_enabled_ || stats.channel_stats.empty()) {
return;
}
int srtp_crypto_suite = stats.channel_stats[0].srtp_crypto_suite;
int ssl_cipher_suite = stats.channel_stats[0].ssl_cipher_suite;
if (srtp_crypto_suite == rtc::SRTP_INVALID_CRYPTO_SUITE &&
ssl_cipher_suite == rtc::TLS_NULL_WITH_NULL_NULL) {
return;
}
PeerConnectionEnumCounterType srtp_counter_type;
PeerConnectionEnumCounterType ssl_counter_type;
if (stats.transport_name == cricket::CN_AUDIO) {
srtp_counter_type = kEnumCounterAudioSrtpCipher;
ssl_counter_type = kEnumCounterAudioSslCipher;
} else if (stats.transport_name == cricket::CN_VIDEO) {
srtp_counter_type = kEnumCounterVideoSrtpCipher;
ssl_counter_type = kEnumCounterVideoSslCipher;
} else if (stats.transport_name == cricket::CN_DATA) {
srtp_counter_type = kEnumCounterDataSrtpCipher;
ssl_counter_type = kEnumCounterDataSslCipher;
} else {
RTC_NOTREACHED();
return;
}
if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE) {
metrics_observer()->IncrementSparseEnumCounter(srtp_counter_type,
srtp_crypto_suite);
}
if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
metrics_observer()->IncrementSparseEnumCounter(ssl_counter_type,
ssl_cipher_suite);
}
}
void PeerConnection::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
RTC_DCHECK(worker_thread()->IsCurrent());
RTC_DCHECK(call_);
call_->OnSentPacket(sent_packet);
}
const std::string PeerConnection::GetTransportName(
const std::string& content_name) {
cricket::BaseChannel* channel = GetChannel(content_name);
if (channel) {
return channel->transport_name();
}
if (sctp_transport_) {
RTC_DCHECK(sctp_content_name_);
RTC_DCHECK(sctp_transport_name_);
if (content_name == *sctp_content_name_) {
return *sctp_transport_name_;
}
}
// Return an empty string if failed to retrieve the transport name.
return "";
}
void PeerConnection::DestroyRtcpTransport_n(const std::string& transport_name) {
RTC_DCHECK(network_thread()->IsCurrent());
transport_controller_->DestroyDtlsTransport_n(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
void PeerConnection::DestroyTransceiverChannel(
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
transceiver) {
RTC_DCHECK(transceiver);
cricket::BaseChannel* channel = transceiver->internal()->channel();
if (channel) {
transceiver->internal()->SetChannel(nullptr);
DestroyBaseChannel(channel);
}
}
void PeerConnection::DestroyDataChannel() {
if (rtp_data_channel_) {
OnDataChannelDestroyed();
DestroyBaseChannel(rtp_data_channel_);
rtp_data_channel_ = nullptr;
}
// Note: Cannot use rtc::Bind to create a functor to invoke because it will
// grab a reference to this PeerConnection. If this is called from the
// PeerConnection destructor, the RefCountedObject vtable will have already
// been destroyed (since it is a subclass of PeerConnection) and using
// rtc::Bind will cause "Pure virtual function called" error to appear.
if (sctp_transport_) {
OnDataChannelDestroyed();
network_thread()->Invoke<void>(RTC_FROM_HERE,
[this] { DestroySctpTransport_n(); });
}
}
void PeerConnection::DestroyBaseChannel(cricket::BaseChannel* channel) {
RTC_DCHECK(channel);
RTC_DCHECK(channel->rtp_dtls_transport());
// Need to cache these before destroying the base channel so that we do not
// access uninitialized memory.
const std::string transport_name =
channel->rtp_dtls_transport()->transport_name();
const bool need_to_delete_rtcp = (channel->rtcp_dtls_transport() != nullptr);
switch (channel->media_type()) {
case cricket::MEDIA_TYPE_AUDIO:
channel_manager()->DestroyVoiceChannel(
static_cast<cricket::VoiceChannel*>(channel));
break;
case cricket::MEDIA_TYPE_VIDEO:
channel_manager()->DestroyVideoChannel(
static_cast<cricket::VideoChannel*>(channel));
break;
case cricket::MEDIA_TYPE_DATA:
channel_manager()->DestroyRtpDataChannel(
static_cast<cricket::RtpDataChannel*>(channel));
break;
default:
RTC_NOTREACHED() << "Unknown media type: " << channel->media_type();
break;
}
// |channel| can no longer be used.
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
if (need_to_delete_rtcp) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
}
} // namespace webrtc