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/*
* 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.
*/
#ifndef WEBRTC_PC_WEBRTCSESSION_H_
#define WEBRTC_PC_WEBRTCSESSION_H_
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/statstypes.h"
#include "webrtc/call/call.h"
#include "webrtc/p2p/base/candidate.h"
#include "webrtc/p2p/base/transportcontroller.h"
#include "webrtc/pc/datachannel.h"
#include "webrtc/pc/mediasession.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/rtc_base/optional.h"
#include "webrtc/rtc_base/sigslot.h"
#include "webrtc/rtc_base/sslidentity.h"
#include "webrtc/rtc_base/thread.h"
#ifdef HAVE_QUIC
#include "webrtc/pc/quicdatatransport.h"
#endif // HAVE_QUIC
namespace cricket {
class ChannelManager;
class RtpDataChannel;
class SctpTransportInternal;
class SctpTransportInternalFactory;
class StatsReport;
class VideoChannel;
class VoiceChannel;
#ifdef HAVE_QUIC
class QuicTransportChannel;
#endif // HAVE_QUIC
} // namespace cricket
namespace webrtc {
class IceRestartAnswerLatch;
class JsepIceCandidate;
class MediaStreamSignaling;
class RtcEventLog;
class WebRtcSessionDescriptionFactory;
extern const char kBundleWithoutRtcpMux[];
extern const char kCreateChannelFailed[];
extern const char kInvalidCandidates[];
extern const char kInvalidSdp[];
extern const char kMlineMismatch[];
extern const char kPushDownTDFailed[];
extern const char kSdpWithoutDtlsFingerprint[];
extern const char kSdpWithoutSdesCrypto[];
extern const char kSdpWithoutIceUfragPwd[];
extern const char kSdpWithoutSdesAndDtlsDisabled[];
extern const char kSessionError[];
extern const char kSessionErrorDesc[];
extern const char kDtlsSrtpSetupFailureRtp[];
extern const char kDtlsSrtpSetupFailureRtcp[];
extern const char kEnableBundleFailed[];
// Maximum number of received video streams that will be processed by webrtc
// even if they are not signalled beforehand.
extern const int kMaxUnsignalledRecvStreams;
// ICE state callback interface.
class IceObserver {
public:
IceObserver() {}
// Called any time the IceConnectionState changes
virtual void OnIceConnectionStateChange(
PeerConnectionInterface::IceConnectionState new_state) {}
// Called any time the IceGatheringState changes
virtual void OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) {}
// New Ice candidate have been found.
virtual void OnIceCandidate(
std::unique_ptr<IceCandidateInterface> candidate) = 0;
// Some local ICE candidates have been removed.
virtual void OnIceCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) = 0;
// Called whenever the state changes between receiving and not receiving.
virtual void OnIceConnectionReceivingChange(bool receiving) {}
protected:
~IceObserver() {}
private:
RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver);
};
// Statistics for all the transports of the session.
typedef std::map<std::string, cricket::TransportStats> TransportStatsMap;
typedef std::map<std::string, std::string> ProxyTransportMap;
// TODO(pthatcher): Think of a better name for this. We already have
// a TransportStats in transport.h. Perhaps TransportsStats?
struct SessionStats {
ProxyTransportMap proxy_to_transport;
TransportStatsMap transport_stats;
};
struct ChannelNamePair {
ChannelNamePair(
const std::string& content_name, const std::string& transport_name)
: content_name(content_name), transport_name(transport_name) {}
std::string content_name;
std::string transport_name;
};
struct ChannelNamePairs {
rtc::Optional<ChannelNamePair> voice;
rtc::Optional<ChannelNamePair> video;
rtc::Optional<ChannelNamePair> data;
};
// A WebRtcSession manages general session state. This includes negotiation
// of both the application-level and network-level protocols: the former
// defines what will be sent and the latter defines how it will be sent. Each
// network-level protocol is represented by a Transport object. Each Transport
// participates in the network-level negotiation. The individual streams of
// packets are represented by TransportChannels. The application-level protocol
// is represented by SessionDecription objects.
class WebRtcSession :
public DataChannelProviderInterface,
public sigslot::has_slots<> {
public:
enum State {
STATE_INIT = 0,
STATE_SENTOFFER, // Sent offer, waiting for answer.
STATE_RECEIVEDOFFER, // Received an offer. Need to send answer.
STATE_SENTPRANSWER, // Sent provisional answer. Need to send answer.
STATE_RECEIVEDPRANSWER, // Received provisional answer, waiting for answer.
STATE_INPROGRESS, // Offer/answer exchange completed.
STATE_CLOSED, // Close() was called.
};
enum Error {
ERROR_NONE = 0, // no error
ERROR_CONTENT = 1, // channel errors in SetLocalContent/SetRemoteContent
ERROR_TRANSPORT = 2, // transport error of some kind
};
// |sctp_factory| may be null, in which case SCTP is treated as unsupported.
WebRtcSession(
Call* call,
cricket::ChannelManager* channel_manager,
const cricket::MediaConfig& media_config,
RtcEventLog* event_log,
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
cricket::PortAllocator* port_allocator,
std::unique_ptr<cricket::TransportController> transport_controller,
std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory);
virtual ~WebRtcSession();
// These are const to allow them to be called from const methods.
rtc::Thread* network_thread() const { return network_thread_; }
rtc::Thread* worker_thread() const { return worker_thread_; }
rtc::Thread* signaling_thread() const { return signaling_thread_; }
// The ID of this session.
const std::string& id() const { return sid_; }
bool Initialize(
const PeerConnectionFactoryInterface::Options& options,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
// Deletes the voice, video and data channel and changes the session state
// to STATE_CLOSED.
void Close();
// Returns true if we were the initial offerer.
bool initial_offerer() const { return initial_offerer_; }
// Returns the current state of the session. See the enum above for details.
// Each time the state changes, we will fire this signal.
State state() const { return state_; }
sigslot::signal2<WebRtcSession*, State> SignalState;
// Returns the last error in the session. See the enum above for details.
Error error() const { return error_; }
const std::string& error_desc() const { return error_desc_; }
void RegisterIceObserver(IceObserver* observer) {
ice_observer_ = observer;
}
// Exposed for stats collecting.
// TODO(steveanton): Switch callers to use the plural form and remove these.
virtual cricket::VoiceChannel* voice_channel() {
if (voice_channels_.empty()) {
return nullptr;
} else {
return voice_channels_[0];
}
}
virtual cricket::VideoChannel* video_channel() {
if (video_channels_.empty()) {
return nullptr;
} else {
return video_channels_[0];
}
}
virtual std::vector<cricket::VoiceChannel*> voice_channels() const {
return voice_channels_;
}
virtual std::vector<cricket::VideoChannel*> video_channels() const {
return video_channels_;
}
// Only valid when using deprecated RTP data channels.
virtual cricket::RtpDataChannel* rtp_data_channel() {
return rtp_data_channel_.get();
}
virtual rtc::Optional<std::string> sctp_content_name() const {
return sctp_content_name_;
}
virtual rtc::Optional<std::string> sctp_transport_name() const {
return sctp_transport_name_;
}
cricket::BaseChannel* GetChannel(const std::string& content_name);
cricket::SecurePolicy SdesPolicy() const;
// Get current SSL role used by SCTP's underlying transport.
bool GetSctpSslRole(rtc::SSLRole* role);
// Get SSL role for an arbitrary m= section (handles bundling correctly).
// TODO(deadbeef): This is only used internally by the session description
// factory, it shouldn't really be public).
bool GetSslRole(const std::string& content_name, rtc::SSLRole* role);
void CreateOffer(
CreateSessionDescriptionObserver* observer,
const PeerConnectionInterface::RTCOfferAnswerOptions& options,
const cricket::MediaSessionOptions& session_options);
void CreateAnswer(CreateSessionDescriptionObserver* observer,
const cricket::MediaSessionOptions& session_options);
// The ownership of |desc| will be transferred after this call.
bool SetLocalDescription(SessionDescriptionInterface* desc,
std::string* err_desc);
// The ownership of |desc| will be transferred after this call.
bool SetRemoteDescription(SessionDescriptionInterface* desc,
std::string* err_desc);
bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
bool RemoveRemoteIceCandidates(
const std::vector<cricket::Candidate>& candidates);
cricket::IceConfig ParseIceConfig(
const PeerConnectionInterface::RTCConfiguration& config) const;
void SetIceConfig(const cricket::IceConfig& ice_config);
// Start gathering candidates for any new transports, or transports doing an
// ICE restart.
void MaybeStartGathering();
const SessionDescriptionInterface* local_description() const {
return pending_local_description_ ? pending_local_description_.get()
: current_local_description_.get();
}
const SessionDescriptionInterface* remote_description() const {
return pending_remote_description_ ? pending_remote_description_.get()
: current_remote_description_.get();
}
const SessionDescriptionInterface* current_local_description() const {
return current_local_description_.get();
}
const SessionDescriptionInterface* current_remote_description() const {
return current_remote_description_.get();
}
const SessionDescriptionInterface* pending_local_description() const {
return pending_local_description_.get();
}
const SessionDescriptionInterface* pending_remote_description() const {
return pending_remote_description_.get();
}
// Get the id used as a media stream track's "id" field from ssrc.
virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
// Implements DataChannelProviderInterface.
bool SendData(const cricket::SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
cricket::SendDataResult* result) override;
bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
void AddSctpDataStream(int sid) override;
void RemoveSctpDataStream(int sid) override;
bool ReadyToSendData() const override;
virtual Call::Stats GetCallStats();
// Returns stats for all channels of all transports.
// This avoids exposing the internal structures used to track them.
// The parameterless version creates |ChannelNamePairs| from |voice_channel|,
// |video_channel| and |voice_channel| if available - this requires it to be
// called on the signaling thread - and invokes the other |GetStats|. The
// other |GetStats| can be invoked on any thread; if not invoked on the
// network thread a thread hop will happen.
std::unique_ptr<SessionStats> GetStats_s();
virtual std::unique_ptr<SessionStats> GetStats(
const ChannelNamePairs& channel_name_pairs);
// virtual so it can be mocked in unit tests
virtual bool GetLocalCertificate(
const std::string& transport_name,
rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
// Caller owns returned certificate
virtual std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
const std::string& transport_name);
cricket::DataChannelType data_channel_type() const;
// Returns true if there was an ICE restart initiated by the remote offer.
bool IceRestartPending(const std::string& content_name) const;
// Set the "needs-ice-restart" flag as described in JSEP. After the flag is
// set, offers should generate new ufrags/passwords until an ICE restart
// occurs.
void SetNeedsIceRestartFlag();
// Returns true if the ICE restart flag above was set, and no ICE restart has
// occurred yet for this transport (by applying a local description with
// changed ufrag/password). If the transport has been deleted as a result of
// bundling, returns false.
bool NeedsIceRestart(const std::string& content_name) const;
// Called when an RTCCertificate is generated or retrieved by
// WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
void OnCertificateReady(
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
void OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp);
// For unit test.
bool waiting_for_certificate_for_testing() const;
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
void set_metrics_observer(
webrtc::MetricsObserverInterface* metrics_observer) {
metrics_observer_ = metrics_observer;
transport_controller_->SetMetricsObserver(metrics_observer);
}
// Called when voice_channel_, video_channel_ and
// rtp_data_channel_/sctp_transport_ are created and destroyed. As a result
// of, for example, setting a new description.
sigslot::signal0<> SignalVoiceChannelCreated;
sigslot::signal0<> SignalVoiceChannelDestroyed;
sigslot::signal0<> SignalVideoChannelCreated;
sigslot::signal0<> SignalVideoChannelDestroyed;
sigslot::signal0<> SignalDataChannelCreated;
sigslot::signal0<> SignalDataChannelDestroyed;
// Called when a valid data channel OPEN message is received.
// std::string represents the data channel label.
sigslot::signal2<const std::string&, const InternalDataChannelInit&>
SignalDataChannelOpenMessage;
#ifdef HAVE_QUIC
QuicDataTransport* quic_data_transport() {
return quic_data_transport_.get();
}
#endif // HAVE_QUIC
private:
// Indicates the type of SessionDescription in a call to SetLocalDescription
// and SetRemoteDescription.
enum Action {
kOffer,
kPrAnswer,
kAnswer,
};
// Non-const versions of local_description()/remote_description(), for use
// internally.
SessionDescriptionInterface* mutable_local_description() {
return pending_local_description_ ? pending_local_description_.get()
: current_local_description_.get();
}
SessionDescriptionInterface* mutable_remote_description() {
return pending_remote_description_ ? pending_remote_description_.get()
: current_remote_description_.get();
}
// Log session state.
void LogState(State old_state, State new_state);
// Updates the state, signaling if necessary.
virtual void SetState(State state);
// Updates the error state, signaling if necessary.
// TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|.
virtual void SetError(Error error, const std::string& error_desc);
bool UpdateSessionState(Action action, cricket::ContentSource source,
std::string* err_desc);
static Action GetAction(const std::string& type);
// Push the media parts of the local or remote session description
// down to all of the channels.
bool PushdownMediaDescription(cricket::ContentAction action,
cricket::ContentSource source,
std::string* error_desc);
bool PushdownSctpParameters_n(cricket::ContentSource source);
bool PushdownTransportDescription(cricket::ContentSource source,
cricket::ContentAction action,
std::string* error_desc);
// Helper methods to push local and remote transport descriptions.
bool PushdownLocalTransportDescription(
const cricket::SessionDescription* sdesc,
cricket::ContentAction action,
std::string* error_desc);
bool PushdownRemoteTransportDescription(
const cricket::SessionDescription* sdesc,
cricket::ContentAction action,
std::string* error_desc);
// Returns true and the TransportInfo of the given |content_name|
// from |description|. Returns false if it's not available.
static bool GetTransportDescription(
const cricket::SessionDescription* description,
const std::string& content_name,
cricket::TransportDescription* info);
// Returns the name of the transport channel when BUNDLE is enabled, or
// nullptr if the channel is not part of any bundle.
const std::string* GetBundleTransportName(
const cricket::ContentInfo* content,
const cricket::ContentGroup* bundle);
// Cause all the BaseChannels in the bundle group to have the same
// transport channel.
bool EnableBundle(const cricket::ContentGroup& bundle);
// Enables media channels to allow sending of media.
void EnableChannels();
// Returns the media index for a local ice candidate given the content name.
// Returns false if the local session description does not have a media
// content called |content_name|.
bool GetLocalCandidateMediaIndex(const std::string& content_name,
int* sdp_mline_index);
// Uses all remote candidates in |remote_desc| in this session.
bool UseCandidatesInSessionDescription(
const SessionDescriptionInterface* remote_desc);
// Uses |candidate| in this session.
bool UseCandidate(const IceCandidateInterface* candidate);
// Deletes the corresponding channel of contents that don't exist in |desc|.
// |desc| can be null. This means that all channels are deleted.
void RemoveUnusedChannels(const cricket::SessionDescription* desc);
// Allocates media channels based on the |desc|. If |desc| doesn't have
// the BUNDLE option, this method will disable BUNDLE in PortAllocator.
// This method will also delete any existing media channels before creating.
bool CreateChannels(const cricket::SessionDescription* desc);
// Helper methods to create media channels.
bool CreateVoiceChannel(const cricket::ContentInfo* content,
const std::string* bundle_transport);
bool CreateVideoChannel(const cricket::ContentInfo* content,
const std::string* bundle_transport);
bool CreateDataChannel(const cricket::ContentInfo* content,
const std::string* bundle_transport);
std::unique_ptr<SessionStats> GetStats_n(
const ChannelNamePairs& channel_name_pairs);
bool CreateSctpTransport_n(const std::string& content_name,
const std::string& transport_name);
// For bundling.
void ChangeSctpTransport_n(const std::string& transport_name);
void DestroySctpTransport_n();
// SctpTransport signal handlers. Needed to marshal signals from the network
// to signaling thread.
void OnSctpTransportReadyToSendData_n();
// This may be called with "false" if the direction of the m= section causes
// us to tear down the SCTP connection.
void OnSctpTransportReadyToSendData_s(bool ready);
void OnSctpTransportDataReceived_n(const cricket::ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& payload);
// Beyond just firing the signal to the signaling thread, listens to SCTP
// CONTROL messages on unused SIDs and processes them as OPEN messages.
void OnSctpTransportDataReceived_s(const cricket::ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& payload);
void OnSctpStreamClosedRemotely_n(int sid);
std::string BadStateErrMsg(State state);
void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
void SetIceConnectionReceiving(bool receiving);
bool ValidateBundleSettings(const cricket::SessionDescription* desc);
bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
// Below methods are helper methods which verifies SDP.
bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
cricket::ContentSource source,
std::string* err_desc);
// Check if a call to SetLocalDescription is acceptable with |action|.
bool ExpectSetLocalDescription(Action action);
// Check if a call to SetRemoteDescription is acceptable with |action|.
bool ExpectSetRemoteDescription(Action action);
// Verifies a=setup attribute as per RFC 5763.
bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
Action action);
// Returns true if we are ready to push down the remote candidate.
// |remote_desc| is the new remote description, or NULL if the current remote
// description should be used. Output |valid| is true if the candidate media
// index is valid.
bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
const SessionDescriptionInterface* remote_desc,
bool* valid);
// Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
// this session.
bool SrtpRequired() const;
// TransportController signal handlers.
void OnTransportControllerConnectionState(cricket::IceConnectionState state);
void OnTransportControllerReceiving(bool receiving);
void OnTransportControllerGatheringState(cricket::IceGatheringState state);
void OnTransportControllerCandidatesGathered(
const std::string& transport_name,
const std::vector<cricket::Candidate>& candidates);
void OnTransportControllerCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates);
void OnTransportControllerDtlsHandshakeError(rtc::SSLHandshakeError error);
std::string GetSessionErrorMsg();
// Invoked when TransportController connection completion is signaled.
// Reports stats for all transports in use.
void ReportTransportStats();
// Gather the usage of IPv4/IPv6 as best connection.
void ReportBestConnectionState(const cricket::TransportStats& stats);
void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
void OnSentPacket_w(const rtc::SentPacket& sent_packet);
const std::string GetTransportName(const std::string& content_name);
void DestroyRtcpTransport_n(const std::string& transport_name);
void RemoveAndDestroyVideoChannel(cricket::VideoChannel* video_channel);
void DestroyVideoChannel(cricket::VideoChannel* video_channel);
void RemoveAndDestroyVoiceChannel(cricket::VoiceChannel* voice_channel);
void DestroyVoiceChannel(cricket::VoiceChannel* voice_channel);
void DestroyDataChannel();
rtc::Thread* const network_thread_;
rtc::Thread* const worker_thread_;
rtc::Thread* const signaling_thread_;
State state_ = STATE_INIT;
Error error_ = ERROR_NONE;
std::string error_desc_;
const std::string sid_;
bool initial_offerer_ = false;
const std::unique_ptr<cricket::TransportController> transport_controller_;
const std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory_;
const cricket::MediaConfig media_config_;
RtcEventLog* event_log_;
Call* call_;
// TODO(steveanton): voice_channels_ and video_channels_ used to be a single
// VoiceChannel/VideoChannel respectively but are being changed to support
// multiple m= lines in unified plan. But until more work is done, these can
// only have 0 or 1 channel each.
// These channels are owned by ChannelManager.
std::vector<cricket::VoiceChannel*> voice_channels_;
std::vector<cricket::VideoChannel*> video_channels_;
// |rtp_data_channel_| is used if in RTP data channel mode, |sctp_transport_|
// when using SCTP.
// TODO(steveanton): This should be changed to a bare pointer because
// WebRtcSession doesn't actually own the RtpDataChannel
// (ChannelManager does).
std::unique_ptr<cricket::RtpDataChannel> rtp_data_channel_;
std::unique_ptr<cricket::SctpTransportInternal> sctp_transport_;
// |sctp_transport_name_| keeps track of what DTLS transport the SCTP
// transport is using (which can change due to bundling).
rtc::Optional<std::string> sctp_transport_name_;
// |sctp_content_name_| is the content name (MID) in SDP.
rtc::Optional<std::string> sctp_content_name_;
// Value cached on signaling thread. Only updated when SctpReadyToSendData
// fires on the signaling thread.
bool sctp_ready_to_send_data_ = false;
// Same as signals provided by SctpTransport, but these are guaranteed to
// fire on the signaling thread, whereas SctpTransport fires on the networking
// thread.
// |sctp_invoker_| is used so that any signals queued on the signaling thread
// from the network thread are immediately discarded if the SctpTransport is
// destroyed (due to m= section being rejected).
// TODO(deadbeef): Use a proxy object to ensure that method calls/signals
// are marshalled to the right thread. Could almost use proxy.h for this,
// but it doesn't have a mechanism for marshalling sigslot::signals
std::unique_ptr<rtc::AsyncInvoker> sctp_invoker_;
sigslot::signal1<bool> SignalSctpReadyToSendData;
sigslot::signal2<const cricket::ReceiveDataParams&,
const rtc::CopyOnWriteBuffer&>
SignalSctpDataReceived;
sigslot::signal1<int> SignalSctpStreamClosedRemotely;
cricket::ChannelManager* channel_manager_;
IceObserver* ice_observer_;
PeerConnectionInterface::IceConnectionState ice_connection_state_;
bool ice_connection_receiving_;
std::unique_ptr<SessionDescriptionInterface> current_local_description_;
std::unique_ptr<SessionDescriptionInterface> pending_local_description_;
std::unique_ptr<SessionDescriptionInterface> current_remote_description_;
std::unique_ptr<SessionDescriptionInterface> pending_remote_description_;
// If the remote peer is using a older version of implementation.
bool older_version_remote_peer_;
bool dtls_enabled_;
// Specifies which kind of data channel is allowed. This is controlled
// by the chrome command-line flag and constraints:
// 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
// constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
// not set or false, SCTP is allowed (DCT_SCTP);
// 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
// 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
// The data channel type could be DCT_QUIC if the QUIC data channel is
// enabled.
cricket::DataChannelType data_channel_type_;
// List of content names for which the remote side triggered an ICE restart.
std::set<std::string> pending_ice_restarts_;
std::unique_ptr<WebRtcSessionDescriptionFactory> webrtc_session_desc_factory_;
// Member variables for caching global options.
cricket::AudioOptions audio_options_;
cricket::VideoOptions video_options_;
MetricsObserverInterface* metrics_observer_;
// Declares the bundle policy for the WebRTCSession.
PeerConnectionInterface::BundlePolicy bundle_policy_;
// Declares the RTCP mux policy for the WebRTCSession.
PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
bool received_first_video_packet_ = false;
bool received_first_audio_packet_ = false;
#ifdef HAVE_QUIC
std::unique_ptr<QuicDataTransport> quic_data_transport_;
#endif // HAVE_QUIC
RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
};
} // namespace webrtc
#endif // WEBRTC_PC_WEBRTCSESSION_H_