| /* |
| * 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 PC_WEBRTCSESSION_H_ |
| #define PC_WEBRTCSESSION_H_ |
| |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <vector> |
| |
| #include "api/optional.h" |
| #include "api/peerconnectioninterface.h" |
| #include "api/statstypes.h" |
| #include "call/call.h" |
| #include "p2p/base/candidate.h" |
| #include "p2p/base/transportcontroller.h" |
| #include "pc/datachannel.h" |
| #include "pc/mediasession.h" |
| #include "rtc_base/constructormagic.h" |
| #include "rtc_base/sigslot.h" |
| #include "rtc_base/sslidentity.h" |
| #include "rtc_base/thread.h" |
| |
| #ifdef HAVE_QUIC |
| #include "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 kMlineMismatchInAnswer[]; |
| extern const char kMlineMismatchInSubsequentOffer[]; |
| 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, |
| }; |
| |
| // Return all managed, non-null channels. |
| std::vector<cricket::BaseChannel*> Channels() const; |
| |
| // 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 // PC_WEBRTCSESSION_H_ |