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webrtc / src / f888bb58da04c5095759b5ec7ce2e1fa2cd414fd / . / talk / app / webrtc / peerconnection.cc
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/*
* libjingle
* Copyright 2012 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/peerconnection.h"
#include <vector>
#include <cctype> // for isdigit
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/dtmfsender.h"
#include "talk/app/webrtc/jsepicecandidate.h"
#include "talk/app/webrtc/jsepsessiondescription.h"
#include "talk/app/webrtc/mediaconstraintsinterface.h"
#include "talk/app/webrtc/mediastream.h"
#include "talk/app/webrtc/mediastreamproxy.h"
#include "talk/app/webrtc/mediastreamtrackproxy.h"
#include "talk/app/webrtc/remoteaudiosource.h"
#include "talk/app/webrtc/remoteaudiotrack.h"
#include "talk/app/webrtc/remotevideocapturer.h"
#include "talk/app/webrtc/rtpreceiver.h"
#include "talk/app/webrtc/rtpsender.h"
#include "talk/app/webrtc/streamcollection.h"
#include "talk/app/webrtc/videosource.h"
#include "talk/app/webrtc/videotrack.h"
#include "talk/media/sctp/sctpdataengine.h"
#include "talk/session/media/channelmanager.h"
#include "webrtc/base/arraysize.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/p2p/client/basicportallocator.h"
#include "webrtc/system_wrappers/include/field_trial.h"
namespace {
using webrtc::DataChannel;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStreamInterface;
using webrtc::PeerConnectionInterface;
using webrtc::RtpSenderInterface;
using webrtc::StreamCollection;
using webrtc::StunConfigurations;
using webrtc::TurnConfigurations;
typedef webrtc::PortAllocatorFactoryInterface::StunConfiguration
StunConfiguration;
typedef webrtc::PortAllocatorFactoryInterface::TurnConfiguration
TurnConfiguration;
static const char kDefaultStreamLabel[] = "default";
static const char kDefaultAudioTrackLabel[] = "defaulta0";
static const char kDefaultVideoTrackLabel[] = "defaultv0";
// The min number of tokens must present in Turn host uri.
// e.g. user@turn.example.org
static const size_t kTurnHostTokensNum = 2;
// Number of tokens must be preset when TURN uri has transport param.
static const size_t kTurnTransportTokensNum = 2;
// The default stun port.
static const int kDefaultStunPort = 3478;
static const int kDefaultStunTlsPort = 5349;
static const char kTransport[] = "transport";
static const char kUdpTransportType[] = "udp";
static const char kTcpTransportType[] = "tcp";
// NOTE: Must be in the same order as the ServiceType enum.
static const char* kValidIceServiceTypes[] = {"stun", "stuns", "turn", "turns"};
// NOTE: A loop below assumes that the first value of this enum is 0 and all
// other values are incremental.
enum ServiceType {
STUN = 0, // Indicates a STUN server.
STUNS, // Indicates a STUN server used with a TLS session.
TURN, // Indicates a TURN server
TURNS, // Indicates a TURN server used with a TLS session.
INVALID, // Unknown.
};
static_assert(INVALID == arraysize(kValidIceServiceTypes),
"kValidIceServiceTypes must have as many strings as ServiceType "
"has values.");
enum {
MSG_SET_SESSIONDESCRIPTION_SUCCESS = 0,
MSG_SET_SESSIONDESCRIPTION_FAILED,
MSG_CREATE_SESSIONDESCRIPTION_FAILED,
MSG_GETSTATS,
};
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;
};
// |in_str| should be of format
// stunURI = scheme ":" stun-host [ ":" stun-port ]
// scheme = "stun" / "stuns"
// stun-host = IP-literal / IPv4address / reg-name
// stun-port = *DIGIT
//
// draft-petithuguenin-behave-turn-uris-01
// turnURI = scheme ":" turn-host [ ":" turn-port ]
// turn-host = username@IP-literal / IPv4address / reg-name
bool GetServiceTypeAndHostnameFromUri(const std::string& in_str,
ServiceType* service_type,
std::string* hostname) {
const std::string::size_type colonpos = in_str.find(':');
if (colonpos == std::string::npos) {
LOG(LS_WARNING) << "Missing ':' in ICE URI: " << in_str;
return false;
}
if ((colonpos + 1) == in_str.length()) {
LOG(LS_WARNING) << "Empty hostname in ICE URI: " << in_str;
return false;
}
*service_type = INVALID;
for (size_t i = 0; i < arraysize(kValidIceServiceTypes); ++i) {
if (in_str.compare(0, colonpos, kValidIceServiceTypes[i]) == 0) {
*service_type = static_cast<ServiceType>(i);
break;
}
}
if (*service_type == INVALID) {
return false;
}
*hostname = in_str.substr(colonpos + 1, std::string::npos);
return true;
}
bool ParsePort(const std::string& in_str, int* port) {
// Make sure port only contains digits. FromString doesn't check this.
for (const char& c : in_str) {
if (!std::isdigit(c)) {
return false;
}
}
return rtc::FromString(in_str, port);
}
// This method parses IPv6 and IPv4 literal strings, along with hostnames in
// standard hostname:port format.
// Consider following formats as correct.
// |hostname:port|, |[IPV6 address]:port|, |IPv4 address|:port,
// |hostname|, |[IPv6 address]|, |IPv4 address|.
bool ParseHostnameAndPortFromString(const std::string& in_str,
std::string* host,
int* port) {
RTC_DCHECK(host->empty());
if (in_str.at(0) == '[') {
std::string::size_type closebracket = in_str.rfind(']');
if (closebracket != std::string::npos) {
std::string::size_type colonpos = in_str.find(':', closebracket);
if (std::string::npos != colonpos) {
if (!ParsePort(in_str.substr(closebracket + 2, std::string::npos),
port)) {
return false;
}
}
*host = in_str.substr(1, closebracket - 1);
} else {
return false;
}
} else {
std::string::size_type colonpos = in_str.find(':');
if (std::string::npos != colonpos) {
if (!ParsePort(in_str.substr(colonpos + 1, std::string::npos), port)) {
return false;
}
*host = in_str.substr(0, colonpos);
} else {
*host = in_str;
}
}
return !host->empty();
}
// Adds a StunConfiguration or TurnConfiguration to the appropriate list,
// by parsing |url| and using the username/password in |server|.
bool ParseIceServerUrl(const PeerConnectionInterface::IceServer& server,
const std::string& url,
StunConfigurations* stun_config,
TurnConfigurations* turn_config) {
// draft-nandakumar-rtcweb-stun-uri-01
// stunURI = scheme ":" stun-host [ ":" stun-port ]
// scheme = "stun" / "stuns"
// stun-host = IP-literal / IPv4address / reg-name
// stun-port = *DIGIT
// draft-petithuguenin-behave-turn-uris-01
// turnURI = scheme ":" turn-host [ ":" turn-port ]
// [ "?transport=" transport ]
// scheme = "turn" / "turns"
// transport = "udp" / "tcp" / transport-ext
// transport-ext = 1*unreserved
// turn-host = IP-literal / IPv4address / reg-name
// turn-port = *DIGIT
RTC_DCHECK(stun_config != nullptr);
RTC_DCHECK(turn_config != nullptr);
std::vector<std::string> tokens;
std::string turn_transport_type = kUdpTransportType;
RTC_DCHECK(!url.empty());
rtc::tokenize(url, '?', &tokens);
std::string uri_without_transport = tokens[0];
// Let's look into transport= param, if it exists.
if (tokens.size() == kTurnTransportTokensNum) { // ?transport= is present.
std::string uri_transport_param = tokens[1];
rtc::tokenize(uri_transport_param, '=', &tokens);
if (tokens[0] == kTransport) {
// As per above grammar transport param will be consist of lower case
// letters.
if (tokens[1] != kUdpTransportType && tokens[1] != kTcpTransportType) {
LOG(LS_WARNING) << "Transport param should always be udp or tcp.";
return false;
}
turn_transport_type = tokens[1];
}
}
std::string hoststring;
ServiceType service_type;
if (!GetServiceTypeAndHostnameFromUri(uri_without_transport,
&service_type,
&hoststring)) {
LOG(LS_WARNING) << "Invalid transport parameter in ICE URI: " << url;
return false;
}
// GetServiceTypeAndHostnameFromUri should never give an empty hoststring
RTC_DCHECK(!hoststring.empty());
// Let's break hostname.
tokens.clear();
rtc::tokenize_with_empty_tokens(hoststring, '@', &tokens);
std::string username(server.username);
if (tokens.size() > kTurnHostTokensNum) {
LOG(LS_WARNING) << "Invalid user@hostname format: " << hoststring;
return false;
}
if (tokens.size() == kTurnHostTokensNum) {
if (tokens[0].empty() || tokens[1].empty()) {
LOG(LS_WARNING) << "Invalid user@hostname format: " << hoststring;
return false;
}
username.assign(rtc::s_url_decode(tokens[0]));
hoststring = tokens[1];
} else {
hoststring = tokens[0];
}
int port = kDefaultStunPort;
if (service_type == TURNS) {
port = kDefaultStunTlsPort;
turn_transport_type = kTcpTransportType;
}
std::string address;
if (!ParseHostnameAndPortFromString(hoststring, &address, &port)) {
LOG(WARNING) << "Invalid hostname format: " << uri_without_transport;
return false;
}
if (port <= 0 || port > 0xffff) {
LOG(WARNING) << "Invalid port: " << port;
return false;
}
switch (service_type) {
case STUN:
case STUNS:
stun_config->push_back(StunConfiguration(address, port));
break;
case TURN:
case TURNS: {
bool secure = (service_type == TURNS);
turn_config->push_back(TurnConfiguration(address, port,
username,
server.password,
turn_transport_type,
secure));
break;
}
case INVALID:
default:
LOG(WARNING) << "Configuration not supported: " << url;
return false;
}
return true;
}
void ConvertToCricketIceServers(
const std::vector<StunConfiguration>& stuns,
const std::vector<TurnConfiguration>& turns,
cricket::ServerAddresses* cricket_stuns,
std::vector<cricket::RelayServerConfig>* cricket_turns) {
RTC_DCHECK(cricket_stuns && cricket_turns);
for (const StunConfiguration& stun : stuns) {
cricket_stuns->insert(stun.server);
}
int priority = static_cast<int>(turns.size() - 1);
for (const TurnConfiguration& turn : turns) {
cricket::RelayCredentials credentials(turn.username, turn.password);
cricket::RelayServerConfig relay_server(cricket::RELAY_TURN);
cricket::ProtocolType protocol;
// Using VERIFY because ParseIceServers should have already caught an
// invalid transport type.
if (!VERIFY(
cricket::StringToProto(turn.transport_type.c_str(), &protocol))) {
LOG(LS_WARNING) << "Ignoring TURN server " << turn.server << ". "
<< "Reason= Incorrect " << turn.transport_type
<< " transport parameter.";
} else {
relay_server.ports.push_back(
cricket::ProtocolAddress(turn.server, protocol, turn.secure));
relay_server.credentials = credentials;
relay_server.priority = priority;
cricket_turns->push_back(relay_server);
}
// First in the list gets highest priority.
--priority;
}
}
// 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) {
LOG(LS_ERROR) << "MediaStream with label " << new_stream->label()
<< " is already added.";
return false;
}
return true;
}
bool MediaContentDirectionHasSend(cricket::MediaContentDirection dir) {
return dir == cricket::MD_SENDONLY || dir == cricket::MD_SENDRECV;
}
// 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 MediaContentDirectionHasSend(desc->direction())
? desc->streams()
: std::vector<cricket::StreamParams>();
}
bool IsValidOfferToReceiveMedia(int value) {
typedef PeerConnectionInterface::RTCOfferAnswerOptions Options;
return (value >= Options::kUndefined) &&
(value <= Options::kMaxOfferToReceiveMedia);
}
// Add the stream and RTP data channel info to |session_options|.
void AddSendStreams(
cricket::MediaSessionOptions* session_options,
const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
rtp_data_channels) {
session_options->streams.clear();
for (const auto& sender : senders) {
session_options->AddSendStream(sender->media_type(), sender->id(),
sender->stream_id());
}
// 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) {
// |streamid| and |sync_label| are both set to the DataChannel label
// here so they can be signaled the same way as MediaStreams and Tracks.
// For MediaStreams, the sync_label is the MediaStream label and the
// track label is the same as |streamid|.
const std::string& streamid = channel->label();
const std::string& sync_label = channel->label();
session_options->AddSendStream(cricket::MEDIA_TYPE_DATA, streamid,
sync_label);
}
}
}
} // namespace
namespace webrtc {
// Factory class for creating remote MediaStreams and MediaStreamTracks.
class RemoteMediaStreamFactory {
public:
explicit RemoteMediaStreamFactory(rtc::Thread* signaling_thread,
cricket::ChannelManager* channel_manager)
: signaling_thread_(signaling_thread),
channel_manager_(channel_manager) {}
rtc::scoped_refptr<MediaStreamInterface> CreateMediaStream(
const std::string& stream_label) {
return MediaStreamProxy::Create(signaling_thread_,
MediaStream::Create(stream_label));
}
AudioTrackInterface* AddAudioTrack(uint32_t ssrc,
AudioProviderInterface* provider,
webrtc::MediaStreamInterface* stream,
const std::string& track_id) {
return AddTrack<AudioTrackInterface, RemoteAudioTrack, AudioTrackProxy>(
stream, track_id, RemoteAudioSource::Create(ssrc, provider));
}
VideoTrackInterface* AddVideoTrack(webrtc::MediaStreamInterface* stream,
const std::string& track_id) {
return AddTrack<VideoTrackInterface, VideoTrack, VideoTrackProxy>(
stream, track_id,
VideoSource::Create(channel_manager_, new RemoteVideoCapturer(),
nullptr)
.get());
}
private:
template <typename TI, typename T, typename TP, typename S>
TI* AddTrack(MediaStreamInterface* stream,
const std::string& track_id,
const S& source) {
rtc::scoped_refptr<TI> track(
TP::Create(signaling_thread_, T::Create(track_id, source)));
track->set_state(webrtc::MediaStreamTrackInterface::kLive);
if (stream->AddTrack(track)) {
return track;
}
return nullptr;
}
rtc::Thread* signaling_thread_;
cricket::ChannelManager* channel_manager_;
};
bool ConvertRtcOptionsForOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options) {
typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
if (!IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_audio) ||
!IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_video)) {
return false;
}
if (rtc_options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
session_options->recv_audio = (rtc_options.offer_to_receive_audio > 0);
}
if (rtc_options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
session_options->recv_video = (rtc_options.offer_to_receive_video > 0);
}
session_options->vad_enabled = rtc_options.voice_activity_detection;
session_options->transport_options.ice_restart = rtc_options.ice_restart;
session_options->bundle_enabled = rtc_options.use_rtp_mux;
return true;
}
bool ParseConstraintsForAnswer(const MediaConstraintsInterface* constraints,
cricket::MediaSessionOptions* session_options) {
bool value = false;
size_t mandatory_constraints_satisfied = 0;
// kOfferToReceiveAudio defaults to true according to spec.
if (!FindConstraint(constraints,
MediaConstraintsInterface::kOfferToReceiveAudio, &value,
&mandatory_constraints_satisfied) ||
value) {
session_options->recv_audio = true;
}
// kOfferToReceiveVideo defaults to false according to spec. But
// if it is an answer and video is offered, we should still accept video
// per default.
value = false;
if (!FindConstraint(constraints,
MediaConstraintsInterface::kOfferToReceiveVideo, &value,
&mandatory_constraints_satisfied) ||
value) {
session_options->recv_video = true;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kVoiceActivityDetection, &value,
&mandatory_constraints_satisfied)) {
session_options->vad_enabled = value;
}
if (FindConstraint(constraints, MediaConstraintsInterface::kUseRtpMux, &value,
&mandatory_constraints_satisfied)) {
session_options->bundle_enabled = value;
} else {
// kUseRtpMux defaults to true according to spec.
session_options->bundle_enabled = true;
}
if (FindConstraint(constraints, MediaConstraintsInterface::kIceRestart,
&value, &mandatory_constraints_satisfied)) {
session_options->transport_options.ice_restart = value;
} else {
// kIceRestart defaults to false according to spec.
session_options->transport_options.ice_restart = false;
}
if (!constraints) {
return true;
}
return mandatory_constraints_satisfied == constraints->GetMandatory().size();
}
bool ParseIceServers(const PeerConnectionInterface::IceServers& servers,
StunConfigurations* stun_config,
TurnConfigurations* turn_config) {
for (const webrtc::PeerConnectionInterface::IceServer& server : servers) {
if (!server.urls.empty()) {
for (const std::string& url : server.urls) {
if (url.empty()) {
LOG(LS_ERROR) << "Empty uri.";
return false;
}
if (!ParseIceServerUrl(server, url, stun_config, turn_config)) {
return false;
}
}
} else if (!server.uri.empty()) {
// Fallback to old .uri if new .urls isn't present.
if (!ParseIceServerUrl(server, server.uri, stun_config, turn_config)) {
return false;
}
} else {
LOG(LS_ERROR) << "Empty uri.";
return false;
}
}
return true;
}
PeerConnection::PeerConnection(PeerConnectionFactory* factory)
: factory_(factory),
observer_(NULL),
uma_observer_(NULL),
signaling_state_(kStable),
ice_state_(kIceNew),
ice_connection_state_(kIceConnectionNew),
ice_gathering_state_(kIceGatheringNew),
local_streams_(StreamCollection::Create()),
remote_streams_(StreamCollection::Create()) {}
PeerConnection::~PeerConnection() {
TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
RTC_DCHECK(signaling_thread()->IsCurrent());
// Need to detach RTP senders/receivers from WebRtcSession,
// since it's about to be destroyed.
for (const auto& sender : senders_) {
sender->Stop();
}
for (const auto& receiver : receivers_) {
receiver->Stop();
}
}
bool PeerConnection::Initialize(
const PeerConnectionInterface::RTCConfiguration& configuration,
const MediaConstraintsInterface* constraints,
PortAllocatorFactoryInterface* allocator_factory,
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
PeerConnectionObserver* observer) {
RTC_DCHECK(observer != nullptr);
if (!observer) {
return false;
}
// This Initialize function parses ICE servers an extra time, but it will
// be removed once all PortAllocaotrs support SetIceServers.
std::vector<PortAllocatorFactoryInterface::StunConfiguration> stun_config;
std::vector<PortAllocatorFactoryInterface::TurnConfiguration> turn_config;
if (!ParseIceServers(configuration.servers, &stun_config, &turn_config)) {
return false;
}
rtc::scoped_ptr<cricket::PortAllocator> allocator(
allocator_factory->CreatePortAllocator(stun_config, turn_config));
return Initialize(configuration, constraints, allocator.Pass(),
dtls_identity_store.Pass(), observer);
}
bool PeerConnection::Initialize(
const PeerConnectionInterface::RTCConfiguration& configuration,
const MediaConstraintsInterface* constraints,
rtc::scoped_ptr<cricket::PortAllocator> allocator,
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
PeerConnectionObserver* observer) {
TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
RTC_DCHECK(observer != nullptr);
if (!observer) {
return false;
}
observer_ = observer;
port_allocator_ = allocator.Pass();
std::vector<PortAllocatorFactoryInterface::StunConfiguration> stun_config;
std::vector<PortAllocatorFactoryInterface::TurnConfiguration> turn_config;
if (!ParseIceServers(configuration.servers, &stun_config, &turn_config)) {
return false;
}
cricket::ServerAddresses cricket_stuns;
std::vector<cricket::RelayServerConfig> cricket_turns;
ConvertToCricketIceServers(stun_config, turn_config, &cricket_stuns,
&cricket_turns);
port_allocator_->SetIceServers(cricket_stuns, cricket_turns);
// 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;
bool value;
// If IPv6 flag was specified, we'll not override it by experiment.
if (FindConstraint(constraints, MediaConstraintsInterface::kEnableIPv6,
&value, nullptr)) {
if (!value) {
portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
}
} else if (webrtc::field_trial::FindFullName("WebRTC-IPv6Default") ==
"Disabled") {
portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
}
if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
LOG(LS_INFO) << "TCP candidates are disabled.";
}
port_allocator_->set_flags(portallocator_flags);
// No step delay is used while allocating ports.
port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
media_controller_.reset(factory_->CreateMediaController());
remote_stream_factory_.reset(new RemoteMediaStreamFactory(
factory_->signaling_thread(), media_controller_->channel_manager()));
session_.reset(
new WebRtcSession(media_controller_.get(), factory_->signaling_thread(),
factory_->worker_thread(), port_allocator_.get()));
stats_.reset(new StatsCollector(this));
// Initialize the WebRtcSession. It creates transport channels etc.
if (!session_->Initialize(factory_->options(), constraints,
dtls_identity_store.Pass(), configuration)) {
return false;
}
// Register PeerConnection as receiver of local ice candidates.
// All the callbacks will be posted to the application from PeerConnection.
session_->RegisterIceObserver(this);
session_->SignalState.connect(this, &PeerConnection::OnSessionStateChange);
session_->SignalVoiceChannelDestroyed.connect(
this, &PeerConnection::OnVoiceChannelDestroyed);
session_->SignalVideoChannelDestroyed.connect(
this, &PeerConnection::OnVideoChannelDestroyed);
session_->SignalDataChannelCreated.connect(
this, &PeerConnection::OnDataChannelCreated);
session_->SignalDataChannelDestroyed.connect(
this, &PeerConnection::OnDataChannelDestroyed);
session_->SignalDataChannelOpenMessage.connect(
this, &PeerConnection::OnDataChannelOpenMessage);
return true;
}
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);
for (const auto& track : local_stream->GetAudioTracks()) {
auto sender = FindSenderForTrack(track.get());
if (sender == senders_.end()) {
// Normal case; we've never seen this track before.
AudioRtpSender* new_sender = new AudioRtpSender(
track.get(), local_stream->label(), session_.get(), stats_.get());
senders_.push_back(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 TrackInfo* track_info = FindTrackInfo(
local_audio_tracks_, local_stream->label(), track->id());
if (track_info) {
new_sender->SetSsrc(track_info->ssrc);
}
} else {
// 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)->set_stream_id(local_stream->label());
}
}
for (const auto& track : local_stream->GetVideoTracks()) {
auto sender = FindSenderForTrack(track.get());
if (sender == senders_.end()) {
// Normal case; we've never seen this track before.
VideoRtpSender* new_sender = new VideoRtpSender(
track.get(), local_stream->label(), session_.get());
senders_.push_back(new_sender);
const TrackInfo* track_info = FindTrackInfo(
local_video_tracks_, local_stream->label(), track->id());
if (track_info) {
new_sender->SetSsrc(track_info->ssrc);
}
} else {
// 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)->set_stream_id(local_stream->label());
}
}
stats_->AddStream(local_stream);
observer_->OnRenegotiationNeeded();
return true;
}
// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
// indefinitely, when we have unified plan SDP.
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
for (const auto& track : local_stream->GetAudioTracks()) {
auto sender = FindSenderForTrack(track.get());
if (sender == senders_.end()) {
LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
<< " doesn't exist.";
continue;
}
(*sender)->Stop();
senders_.erase(sender);
}
for (const auto& track : local_stream->GetVideoTracks()) {
auto sender = FindSenderForTrack(track.get());
if (sender == senders_.end()) {
LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
<< " doesn't exist.";
continue;
}
(*sender)->Stop();
senders_.erase(sender);
}
local_streams_->RemoveStream(local_stream);
if (IsClosed()) {
return;
}
observer_->OnRenegotiationNeeded();
}
rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
AudioTrackInterface* track) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateDtmfSender");
if (!track) {
LOG(LS_ERROR) << "CreateDtmfSender - track is NULL.";
return NULL;
}
if (!local_streams_->FindAudioTrack(track->id())) {
LOG(LS_ERROR) << "CreateDtmfSender is called with a non local audio track.";
return NULL;
}
rtc::scoped_refptr<DtmfSenderInterface> sender(
DtmfSender::Create(track, signaling_thread(), session_.get()));
if (!sender.get()) {
LOG(LS_ERROR) << "CreateDtmfSender failed on DtmfSender::Create.";
return NULL;
}
return DtmfSenderProxy::Create(signaling_thread(), sender.get());
}
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
const std::string& kind) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
RtpSenderInterface* new_sender;
if (kind == MediaStreamTrackInterface::kAudioKind) {
new_sender = new AudioRtpSender(session_.get(), stats_.get());
} else if (kind == MediaStreamTrackInterface::kVideoKind) {
new_sender = new VideoRtpSender(session_.get());
} else {
LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
return rtc::scoped_refptr<RtpSenderInterface>();
}
senders_.push_back(new_sender);
return RtpSenderProxy::Create(signaling_thread(), new_sender);
}
std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
const {
std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders;
for (const auto& sender : senders_) {
senders.push_back(RtpSenderProxy::Create(signaling_thread(), sender.get()));
}
return senders;
}
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
PeerConnection::GetReceivers() const {
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers;
for (const auto& receiver : receivers_) {
receivers.push_back(
RtpReceiverProxy::Create(signaling_thread(), receiver.get()));
}
return receivers;
}
bool PeerConnection::GetStats(StatsObserver* observer,
MediaStreamTrackInterface* track,
StatsOutputLevel level) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK(signaling_thread()->IsCurrent());
if (!VERIFY(observer != NULL)) {
LOG(LS_ERROR) << "GetStats - observer is NULL.";
return false;
}
stats_->UpdateStats(level);
signaling_thread()->Post(this, MSG_GETSTATS,
new GetStatsMsg(observer, track));
return true;
}
PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
return signaling_state_;
}
PeerConnectionInterface::IceState PeerConnection::ice_state() {
return ice_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();
rtc::scoped_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 (session_->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");
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
return;
}
RTCOfferAnswerOptions options;
bool value;
size_t mandatory_constraints = 0;
if (FindConstraint(constraints,
MediaConstraintsInterface::kOfferToReceiveAudio,
&value,
&mandatory_constraints)) {
options.offer_to_receive_audio =
value ? RTCOfferAnswerOptions::kOfferToReceiveMediaTrue : 0;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kOfferToReceiveVideo,
&value,
&mandatory_constraints)) {
options.offer_to_receive_video =
value ? RTCOfferAnswerOptions::kOfferToReceiveMediaTrue : 0;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kVoiceActivityDetection,
&value,
&mandatory_constraints)) {
options.voice_activity_detection = value;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kIceRestart,
&value,
&mandatory_constraints)) {
options.ice_restart = value;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kUseRtpMux,
&value,
&mandatory_constraints)) {
options.use_rtp_mux = value;
}
CreateOffer(observer, options);
}
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
return;
}
cricket::MediaSessionOptions session_options;
if (!GetOptionsForOffer(options, &session_options)) {
std::string error = "CreateOffer called with invalid options.";
LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(observer, error);
return;
}
session_->CreateOffer(observer, options, session_options);
}
void PeerConnection::CreateAnswer(
CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
return;
}
cricket::MediaSessionOptions session_options;
if (!GetOptionsForAnswer(constraints, &session_options)) {
std::string error = "CreateAnswer called with invalid constraints.";
LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(observer, error);
return;
}
session_->CreateAnswer(observer, constraints, session_options);
}
void PeerConnection::SetLocalDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {
TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
return;
}
if (!desc) {
PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
return;
}
// 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);
std::string error;
if (!session_->SetLocalDescription(desc, &error)) {
PostSetSessionDescriptionFailure(observer, error);
return;
}
// If setting the description decided our SSL role, allocate any necessary
// SCTP sids.
rtc::SSLRole role;
if (session_->data_channel_type() == cricket::DCT_SCTP &&
session_->GetSslRole(&role)) {
AllocateSctpSids(role);
}
// Update state and SSRC of local MediaStreams and DataChannels based on the
// local session description.
const cricket::ContentInfo* audio_content =
GetFirstAudioContent(desc->description());
if (audio_content) {
if (audio_content->rejected) {
RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
} else {
const cricket::AudioContentDescription* audio_desc =
static_cast<const cricket::AudioContentDescription*>(
audio_content->description);
UpdateLocalTracks(audio_desc->streams(), audio_desc->type());
}
}
const cricket::ContentInfo* video_content =
GetFirstVideoContent(desc->description());
if (video_content) {
if (video_content->rejected) {
RemoveTracks(cricket::MEDIA_TYPE_VIDEO);
} else {
const cricket::VideoContentDescription* video_desc =
static_cast<const cricket::VideoContentDescription*>(
video_content->description);
UpdateLocalTracks(video_desc->streams(), video_desc->type());
}
}
const cricket::ContentInfo* data_content =
GetFirstDataContent(desc->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());
}
}
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
signaling_thread()->Post(this, MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
// MaybeStartGathering needs to be called after posting
// MSG_SET_SESSIONDESCRIPTION_SUCCESS, so that we don't signal any candidates
// before signaling that SetLocalDescription completed.
session_->MaybeStartGathering();
}
void PeerConnection::SetRemoteDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {
TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
return;
}
if (!desc) {
PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
return;
}
// 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);
std::string error;
if (!session_->SetRemoteDescription(desc, &error)) {
PostSetSessionDescriptionFailure(observer, error);
return;
}
// If setting the description decided our SSL role, allocate any necessary
// SCTP sids.
rtc::SSLRole role;
if (session_->data_channel_type() == cricket::DCT_SCTP &&
session_->GetSslRole(&role)) {
AllocateSctpSids(role);
}
const cricket::SessionDescription* remote_desc = desc->description();
const cricket::ContentInfo* audio_content = GetFirstAudioContent(remote_desc);
const cricket::ContentInfo* video_content = GetFirstVideoContent(remote_desc);
const cricket::AudioContentDescription* audio_desc =
GetFirstAudioContentDescription(remote_desc);
const cricket::VideoContentDescription* video_desc =
GetFirstVideoContentDescription(remote_desc);
const cricket::DataContentDescription* data_desc =
GetFirstDataContentDescription(remote_desc);
// Check if the descriptions include streams, just in case the peer supports
// MSID, but doesn't indicate so with "a=msid-semantic".
if (remote_desc->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());
// Find all audio rtp streams and create corresponding remote AudioTracks
// and MediaStreams.
if (audio_content) {
if (audio_content->rejected) {
RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
} else {
bool default_audio_track_needed =
!remote_peer_supports_msid_ &&
MediaContentDirectionHasSend(audio_desc->direction());
UpdateRemoteStreamsList(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) {
RemoveTracks(cricket::MEDIA_TYPE_VIDEO);
} else {
bool default_video_track_needed =
!remote_peer_supports_msid_ &&
MediaContentDirectionHasSend(video_desc->direction());
UpdateRemoteStreamsList(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(new_stream);
}
UpdateEndedRemoteMediaStreams();
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
signaling_thread()->Post(this, MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
}
bool PeerConnection::SetConfiguration(const RTCConfiguration& config) {
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
if (port_allocator_) {
std::vector<PortAllocatorFactoryInterface::StunConfiguration> stuns;
std::vector<PortAllocatorFactoryInterface::TurnConfiguration> turns;
if (!ParseIceServers(config.servers, &stuns, &turns)) {
return false;
}
cricket::ServerAddresses cricket_stuns;
std::vector<cricket::RelayServerConfig> cricket_turns;
ConvertToCricketIceServers(stuns, turns, &cricket_stuns, &cricket_turns);
port_allocator_->SetIceServers(cricket_stuns, cricket_turns);
}
session_->SetIceConfig(session_->ParseIceConfig(config));
return session_->SetIceTransports(config.type);
}
bool PeerConnection::AddIceCandidate(
const IceCandidateInterface* ice_candidate) {
TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
return session_->ProcessIceMessage(ice_candidate);
}
void PeerConnection::RegisterUMAObserver(UMAObserver* observer) {
TRACE_EVENT0("webrtc", "PeerConnection::RegisterUmaObserver");
uma_observer_ = observer;
if (session_) {
session_->set_metrics_observer(uma_observer_);
}
// Send information about IPv4/IPv6 status.
if (uma_observer_ && port_allocator_) {
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);
}
}
}
const SessionDescriptionInterface* PeerConnection::local_description() const {
return session_->local_description();
}
const SessionDescriptionInterface* PeerConnection::remote_description() const {
return session_->remote_description();
}
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);
session_->Close();
}
void PeerConnection::OnSessionStateChange(WebRtcSession* /*session*/,
WebRtcSession::State state) {
switch (state) {
case WebRtcSession::STATE_INIT:
ChangeSignalingState(PeerConnectionInterface::kStable);
break;
case WebRtcSession::STATE_SENTOFFER:
ChangeSignalingState(PeerConnectionInterface::kHaveLocalOffer);
break;
case WebRtcSession::STATE_SENTPRANSWER:
ChangeSignalingState(PeerConnectionInterface::kHaveLocalPrAnswer);
break;
case WebRtcSession::STATE_RECEIVEDOFFER:
ChangeSignalingState(PeerConnectionInterface::kHaveRemoteOffer);
break;
case WebRtcSession::STATE_RECEIVEDPRANSWER:
ChangeSignalingState(PeerConnectionInterface::kHaveRemotePrAnswer);
break;
case WebRtcSession::STATE_INPROGRESS:
ChangeSignalingState(PeerConnectionInterface::kStable);
break;
case WebRtcSession::STATE_CLOSED:
ChangeSignalingState(PeerConnectionInterface::kClosed);
break;
default:
break;
}
}
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;
}
default:
RTC_DCHECK(false && "Not implemented");
break;
}
}
void PeerConnection::CreateAudioReceiver(MediaStreamInterface* stream,
AudioTrackInterface* audio_track,
uint32_t ssrc) {
receivers_.push_back(new AudioRtpReceiver(audio_track, ssrc, session_.get()));
}
void PeerConnection::CreateVideoReceiver(MediaStreamInterface* stream,
VideoTrackInterface* video_track,
uint32_t ssrc) {
receivers_.push_back(new VideoRtpReceiver(video_track, ssrc, session_.get()));
}
// TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
// description.
void PeerConnection::DestroyAudioReceiver(MediaStreamInterface* stream,
AudioTrackInterface* audio_track) {
auto it = FindReceiverForTrack(audio_track);
if (it == receivers_.end()) {
LOG(LS_WARNING) << "RtpReceiver for track with id " << audio_track->id()
<< " doesn't exist.";
} else {
(*it)->Stop();
receivers_.erase(it);
}
}
void PeerConnection::DestroyVideoReceiver(MediaStreamInterface* stream,
VideoTrackInterface* video_track) {
auto it = FindReceiverForTrack(video_track);
if (it == receivers_.end()) {
LOG(LS_WARNING) << "RtpReceiver for track with id " << video_track->id()
<< " doesn't exist.";
} else {
(*it)->Stop();
receivers_.erase(it);
}
}
void PeerConnection::OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) {
RTC_DCHECK(signaling_thread()->IsCurrent());
// After transitioning to "closed", ignore any additional states from
// WebRtcSession (such as "disconnected").
if (IsClosed()) {
return;
}
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(const IceCandidateInterface* candidate) {
RTC_DCHECK(signaling_thread()->IsCurrent());
observer_->OnIceCandidate(candidate);
}
void PeerConnection::OnIceComplete() {
RTC_DCHECK(signaling_thread()->IsCurrent());
observer_->OnIceComplete();
}
void PeerConnection::OnIceConnectionReceivingChange(bool receiving) {
RTC_DCHECK(signaling_thread()->IsCurrent());
observer_->OnIceConnectionReceivingChange(receiving);
}
void PeerConnection::ChangeSignalingState(
PeerConnectionInterface::SignalingState 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_);
observer_->OnStateChange(PeerConnectionObserver::kSignalingState);
}
void PeerConnection::PostSetSessionDescriptionFailure(
SetSessionDescriptionObserver* observer,
const std::string& error) {
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
msg->error = error;
signaling_thread()->Post(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(this, MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
}
bool PeerConnection::GetOptionsForOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options) {
if (!ConvertRtcOptionsForOffer(rtc_options, session_options)) {
return false;
}
AddSendStreams(session_options, senders_, rtp_data_channels_);
// Offer to receive audio/video if the constraint is not set and there are
// send streams, or we're currently receiving.
if (rtc_options.offer_to_receive_audio == RTCOfferAnswerOptions::kUndefined) {
session_options->recv_audio =
session_options->HasSendMediaStream(cricket::MEDIA_TYPE_AUDIO) ||
!remote_audio_tracks_.empty();
}
if (rtc_options.offer_to_receive_video == RTCOfferAnswerOptions::kUndefined) {
session_options->recv_video =
session_options->HasSendMediaStream(cricket::MEDIA_TYPE_VIDEO) ||
!remote_video_tracks_.empty();
}
session_options->bundle_enabled =
session_options->bundle_enabled &&
(session_options->has_audio() || session_options->has_video() ||
session_options->has_data());
if (session_->data_channel_type() == cricket::DCT_SCTP && HasDataChannels()) {
session_options->data_channel_type = cricket::DCT_SCTP;
}
return true;
}
bool PeerConnection::GetOptionsForAnswer(
const MediaConstraintsInterface* constraints,
cricket::MediaSessionOptions* session_options) {
session_options->recv_audio = false;
session_options->recv_video = false;
if (!ParseConstraintsForAnswer(constraints, session_options)) {
return false;
}
AddSendStreams(session_options, senders_, rtp_data_channels_);
session_options->bundle_enabled =
session_options->bundle_enabled &&
(session_options->has_audio() || session_options->has_video() ||
session_options->has_data());
// RTP data channel is handled in MediaSessionOptions::AddStream. SCTP streams
// are not signaled in the SDP so does not go through that path and must be
// handled here.
if (session_->data_channel_type() == cricket::DCT_SCTP) {
session_options->data_channel_type = cricket::DCT_SCTP;
}
return true;
}
void PeerConnection::RemoveTracks(cricket::MediaType media_type) {
UpdateLocalTracks(std::vector<cricket::StreamParams>(), media_type);
UpdateRemoteStreamsList(std::vector<cricket::StreamParams>(), false,
media_type, nullptr);
}
void PeerConnection::UpdateRemoteStreamsList(
const cricket::StreamParamsVec& streams,
bool default_track_needed,
cricket::MediaType media_type,
StreamCollection* new_streams) {
TrackInfos* current_tracks = GetRemoteTracks(media_type);
// Find removed tracks. I.e., tracks where the track id or ssrc don't match
// the new StreamParam.
auto track_it = current_tracks->begin();
while (track_it != current_tracks->end()) {
const TrackInfo& info = *track_it;
const cricket::StreamParams* params =
cricket::GetStreamBySsrc(streams, info.ssrc);
bool track_exists = params && params->id == info.track_id;
// If this is a default track, and we still need it, don't remove it.
if ((info.stream_label == kDefaultStreamLabel && default_track_needed) ||
track_exists) {
++track_it;
} else {
OnRemoteTrackRemoved(info.stream_label, info.track_id, media_type);
track_it = current_tracks->erase(track_it);
}
}
// Find new and active tracks.
for (const cricket::StreamParams& params : streams) {
// The sync_label is the MediaStream label and the |stream.id| is the
// track id.
const std::string& stream_label = params.sync_label;
const std::string& track_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 = remote_stream_factory_->CreateMediaStream(stream_label);
remote_streams_->AddStream(stream);
new_streams->AddStream(stream);
}
const TrackInfo* track_info =
FindTrackInfo(*current_tracks, stream_label, track_id);
if (!track_info) {
current_tracks->push_back(TrackInfo(stream_label, track_id, ssrc));
OnRemoteTrackSeen(stream_label, track_id, ssrc, media_type);
}
}
// Add default track if necessary.
if (default_track_needed) {
rtc::scoped_refptr<MediaStreamInterface> default_stream =
remote_streams_->find(kDefaultStreamLabel);
if (!default_stream) {
// Create the new default MediaStream.
default_stream =
remote_stream_factory_->CreateMediaStream(kDefaultStreamLabel);
remote_streams_->AddStream(default_stream);
new_streams->AddStream(default_stream);
}
std::string default_track_id = (media_type == cricket::MEDIA_TYPE_AUDIO)
? kDefaultAudioTrackLabel
: kDefaultVideoTrackLabel;
const TrackInfo* default_track_info =
FindTrackInfo(*current_tracks, kDefaultStreamLabel, default_track_id);
if (!default_track_info) {
current_tracks->push_back(
TrackInfo(kDefaultStreamLabel, default_track_id, 0));
OnRemoteTrackSeen(kDefaultStreamLabel, default_track_id, 0, media_type);
}
}
}
void PeerConnection::OnRemoteTrackSeen(const std::string& stream_label,
const std::string& track_id,
uint32_t ssrc,
cricket::MediaType media_type) {
MediaStreamInterface* stream = remote_streams_->find(stream_label);
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
AudioTrackInterface* audio_track = remote_stream_factory_->AddAudioTrack(
ssrc, session_.get(), stream, track_id);
CreateAudioReceiver(stream, audio_track, ssrc);
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
VideoTrackInterface* video_track =
remote_stream_factory_->AddVideoTrack(stream, track_id);
CreateVideoReceiver(stream, video_track, ssrc);
} else {
RTC_DCHECK(false && "Invalid media type");
}
}
void PeerConnection::OnRemoteTrackRemoved(const std::string& stream_label,
const std::string& track_id,
cricket::MediaType media_type) {
MediaStreamInterface* stream = remote_streams_->find(stream_label);
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
rtc::scoped_refptr<AudioTrackInterface> audio_track =
stream->FindAudioTrack(track_id);
if (audio_track) {
audio_track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
stream->RemoveTrack(audio_track);
DestroyAudioReceiver(stream, audio_track);
}
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
rtc::scoped_refptr<VideoTrackInterface> video_track =
stream->FindVideoTrack(track_id);
if (video_track) {
video_track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
stream->RemoveTrack(video_track);
DestroyVideoReceiver(stream, video_track);
}
} else {
ASSERT(false && "Invalid media type");
}
}
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 (const auto& stream : streams_to_remove) {
remote_streams_->RemoveStream(stream);
observer_->OnRemoveStream(stream);
}
}
void PeerConnection::EndRemoteTracks(cricket::MediaType media_type) {
TrackInfos* current_tracks = GetRemoteTracks(media_type);
for (TrackInfos::iterator track_it = current_tracks->begin();
track_it != current_tracks->end(); ++track_it) {
const TrackInfo& info = *track_it;
MediaStreamInterface* stream = remote_streams_->find(info.stream_label);
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
AudioTrackInterface* track = stream->FindAudioTrack(info.track_id);
// There's no guarantee the track is still available, e.g. the track may
// have been removed from the stream by javascript.
if (track) {
track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
}
}
if (media_type == cricket::MEDIA_TYPE_VIDEO) {
VideoTrackInterface* track = stream->FindVideoTrack(info.track_id);
// There's no guarantee the track is still available, e.g. the track may
// have been removed from the stream by javascript.
if (track) {
track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
}
}
}
}
void PeerConnection::UpdateLocalTracks(
const std::vector<cricket::StreamParams>& streams,
cricket::MediaType media_type) {
TrackInfos* current_tracks = GetLocalTracks(media_type);
// Find removed tracks. I.e., tracks where the track id, stream label or ssrc
// don't match the new StreamParam.
TrackInfos::iterator track_it = current_tracks->begin();
while (track_it != current_tracks->end()) {
const TrackInfo& info = *track_it;
const cricket::StreamParams* params =
cricket::GetStreamBySsrc(streams, info.ssrc);
if (!params || params->id != info.track_id ||
params->sync_label != info.stream_label) {
OnLocalTrackRemoved(info.stream_label, info.track_id, info.ssrc,
media_type);
track_it = current_tracks->erase(track_it);
} else {
++track_it;
}
}
// Find new and active tracks.
for (const cricket::StreamParams& params : streams) {
// The sync_label is the MediaStream label and the |stream.id| is the
// track id.
const std::string& stream_label = params.sync_label;
const std::string& track_id = params.id;
uint32_t ssrc = params.first_ssrc();
const TrackInfo* track_info =
FindTrackInfo(*current_tracks, stream_label, track_id);
if (!track_info) {
current_tracks->push_back(TrackInfo(stream_label, track_id, ssrc));
OnLocalTrackSeen(stream_label, track_id, params.first_ssrc(), media_type);
}
}
}
void PeerConnection::OnLocalTrackSeen(const std::string& stream_label,
const std::string& track_id,
uint32_t ssrc,
cricket::MediaType media_type) {
RtpSenderInterface* sender = FindSenderById(track_id);
if (!sender) {
LOG(LS_WARNING) << "An unknown RtpSender with id " << track_id
<< " has been configured in the local description.";
return;
}
if (sender->media_type() != media_type) {
LOG(LS_WARNING) << "An RtpSender has been configured in the local"
<< " description with an unexpected media type.";
return;
}
sender->set_stream_id(stream_label);
sender->SetSsrc(ssrc);
}
void PeerConnection::OnLocalTrackRemoved(const std::string& stream_label,
const std::string& track_id,
uint32_t ssrc,
cricket::MediaType media_type) {
RtpSenderInterface* sender = FindSenderById(track_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) {
LOG(LS_WARNING) << "An RtpSender has been configured in the local"
<< " description with an unexpected media type.";
return;
}
sender->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 (!VERIFY(data_channel_it != rtp_data_channels_.end())) {
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()) {
LOG(LS_WARNING) << "Remote peer requested a DataChannel but"
<< "CreateDataChannel failed.";
return;
}
channel->SetReceiveSsrc(remote_ssrc);
observer_->OnDataChannel(
DataChannelProxy::Create(signaling_thread(), channel));
}
rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
const std::string& label,
const InternalDataChannelInit* config) {
if (IsClosed()) {
return nullptr;
}
if (session_->data_channel_type() == cricket::DCT_NONE) {
LOG(LS_ERROR)
<< "InternalCreateDataChannel: Data is not supported in this call.";
return nullptr;
}
InternalDataChannelInit new_config =
config ? (*config) : InternalDataChannelInit();
if (session_->data_channel_type() == cricket::DCT_SCTP) {
if (new_config.id < 0) {
rtc::SSLRole role;
if (session_->GetSslRole(&role) &&
!sid_allocator_.AllocateSid(role, &new_config.id)) {
LOG(LS_ERROR) << "No id can be allocated for the SCTP data channel.";
return nullptr;
}
} else if (!sid_allocator_.ReserveSid(new_config.id)) {
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(
session_.get(), session_->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()) {
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);
}
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)) {
LOG(LS_ERROR) << "Failed to allocate SCTP sid.";
continue;
}
channel->SetSctpSid(sid);
}
}
}
void PeerConnection::OnSctpDataChannelClosed(DataChannel* channel) {
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());
}
sctp_data_channels_.erase(it);
return;
}
}
}
void PeerConnection::OnVoiceChannelDestroyed() {
EndRemoteTracks(cricket::MEDIA_TYPE_AUDIO);
}
void PeerConnection::OnVideoChannelDestroyed() {
EndRemoteTracks(cricket::MEDIA_TYPE_VIDEO);
}
void PeerConnection::OnDataChannelCreated() {
for (const auto& channel : sctp_data_channels_) {
channel->OnTransportChannelCreated();
}
}
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()) {
LOG(LS_ERROR) << "Failed to create DataChannel from the OPEN message.";
return;
}
observer_->OnDataChannel(
DataChannelProxy::Create(signaling_thread(), channel));
}
RtpSenderInterface* PeerConnection::FindSenderById(const std::string& id) {
auto it =
std::find_if(senders_.begin(), senders_.end(),
[id](const rtc::scoped_refptr<RtpSenderInterface>& sender) {
return sender->id() == id;
});
return it != senders_.end() ? it->get() : nullptr;
}
std::vector<rtc::scoped_refptr<RtpSenderInterface>>::iterator
PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) {
return std::find_if(
senders_.begin(), senders_.end(),
[track](const rtc::scoped_refptr<RtpSenderInterface>& sender) {
return sender->track() == track;
});
}
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>::iterator
PeerConnection::FindReceiverForTrack(MediaStreamTrackInterface* track) {
return std::find_if(
receivers_.begin(), receivers_.end(),
[track](const rtc::scoped_refptr<RtpReceiverInterface>& receiver) {
return receiver->track() == track;
});
}
PeerConnection::TrackInfos* PeerConnection::GetRemoteTracks(
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_tracks_
: &remote_video_tracks_;
}
PeerConnection::TrackInfos* PeerConnection::GetLocalTracks(
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_tracks_
: &local_video_tracks_;
}
const PeerConnection::TrackInfo* PeerConnection::FindTrackInfo(
const PeerConnection::TrackInfos& infos,
const std::string& stream_label,
const std::string track_id) const {
for (const TrackInfo& track_info : infos) {
if (track_info.stream_label == stream_label &&
track_info.track_id == track_id) {
return &track_info;
}
}
return nullptr;
}
DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
for (const auto& channel : sctp_data_channels_) {
if (channel->id() == sid) {
return channel;
}
}
return nullptr;
}
} // namespace webrtc