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| 1 /* | |
| 2 * libjingle | |
| 3 * Copyright 2012 Google Inc. | |
| 4 * | |
| 5 * Redistribution and use in source and binary forms, with or without | |
| 6 * modification, are permitted provided that the following conditions are met: | |
| 7 * | |
| 8 * 1. Redistributions of source code must retain the above copyright notice, | |
| 9 * this list of conditions and the following disclaimer. | |
| 10 * 2. Redistributions in binary form must reproduce the above copyright notice, | |
| 11 * this list of conditions and the following disclaimer in the documentation | |
| 12 * and/or other materials provided with the distribution. | |
| 13 * 3. The name of the author may not be used to endorse or promote products | |
| 14 * derived from this software without specific prior written permission. | |
| 15 * | |
| 16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED | |
| 17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF | |
| 18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO | |
| 19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, | |
| 20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, | |
| 21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; | |
| 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, | |
| 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR | |
| 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF | |
| 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | |
| 26 */ | |
| 27 | |
| 28 #ifndef TALK_APP_WEBRTC_WEBRTCSESSION_H_ | |
| 29 #define TALK_APP_WEBRTC_WEBRTCSESSION_H_ | |
| 30 | |
| 31 #include <string> | |
| 32 #include <vector> | |
| 33 | |
| 34 #include "talk/app/webrtc/datachannel.h" | |
| 35 #include "talk/app/webrtc/dtmfsender.h" | |
| 36 #include "talk/app/webrtc/mediacontroller.h" | |
| 37 #include "talk/app/webrtc/mediastreamprovider.h" | |
| 38 #include "talk/app/webrtc/peerconnectioninterface.h" | |
| 39 #include "talk/app/webrtc/statstypes.h" | |
| 40 #include "talk/media/base/mediachannel.h" | |
| 41 #include "talk/session/media/mediasession.h" | |
| 42 #include "webrtc/base/sigslot.h" | |
| 43 #include "webrtc/base/sslidentity.h" | |
| 44 #include "webrtc/base/thread.h" | |
| 45 #include "webrtc/p2p/base/transportcontroller.h" | |
| 46 | |
| 47 namespace cricket { | |
| 48 | |
| 49 class ChannelManager; | |
| 50 class DataChannel; | |
| 51 class StatsReport; | |
| 52 class VideoCapturer; | |
| 53 class VideoChannel; | |
| 54 class VoiceChannel; | |
| 55 | |
| 56 } // namespace cricket | |
| 57 | |
| 58 namespace webrtc { | |
| 59 | |
| 60 class IceRestartAnswerLatch; | |
| 61 class JsepIceCandidate; | |
| 62 class MediaStreamSignaling; | |
| 63 class WebRtcSessionDescriptionFactory; | |
| 64 | |
| 65 extern const char kBundleWithoutRtcpMux[]; | |
| 66 extern const char kCreateChannelFailed[]; | |
| 67 extern const char kInvalidCandidates[]; | |
| 68 extern const char kInvalidSdp[]; | |
| 69 extern const char kMlineMismatch[]; | |
| 70 extern const char kPushDownTDFailed[]; | |
| 71 extern const char kSdpWithoutDtlsFingerprint[]; | |
| 72 extern const char kSdpWithoutSdesCrypto[]; | |
| 73 extern const char kSdpWithoutIceUfragPwd[]; | |
| 74 extern const char kSdpWithoutSdesAndDtlsDisabled[]; | |
| 75 extern const char kSessionError[]; | |
| 76 extern const char kSessionErrorDesc[]; | |
| 77 extern const char kDtlsSetupFailureRtp[]; | |
| 78 extern const char kDtlsSetupFailureRtcp[]; | |
| 79 extern const char kEnableBundleFailed[]; | |
| 80 | |
| 81 // Maximum number of received video streams that will be processed by webrtc | |
| 82 // even if they are not signalled beforehand. | |
| 83 extern const int kMaxUnsignalledRecvStreams; | |
| 84 | |
| 85 // ICE state callback interface. | |
| 86 class IceObserver { | |
| 87 public: | |
| 88 IceObserver() {} | |
| 89 // Called any time the IceConnectionState changes | |
| 90 // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to | |
| 91 // conform to the w3c standard. | |
| 92 virtual void OnIceConnectionChange( | |
| 93 PeerConnectionInterface::IceConnectionState new_state) {} | |
| 94 // Called any time the IceGatheringState changes | |
| 95 virtual void OnIceGatheringChange( | |
| 96 PeerConnectionInterface::IceGatheringState new_state) {} | |
| 97 // New Ice candidate have been found. | |
| 98 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0; | |
| 99 // All Ice candidates have been found. | |
| 100 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange. | |
| 101 // (via PeerConnectionObserver) | |
| 102 virtual void OnIceComplete() {} | |
| 103 | |
| 104 // Called whenever the state changes between receiving and not receiving. | |
| 105 virtual void OnIceConnectionReceivingChange(bool receiving) {} | |
| 106 | |
| 107 protected: | |
| 108 ~IceObserver() {} | |
| 109 | |
| 110 private: | |
| 111 RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver); | |
| 112 }; | |
| 113 | |
| 114 // Statistics for all the transports of the session. | |
| 115 typedef std::map<std::string, cricket::TransportStats> TransportStatsMap; | |
| 116 typedef std::map<std::string, std::string> ProxyTransportMap; | |
| 117 | |
| 118 // TODO(pthatcher): Think of a better name for this. We already have | |
| 119 // a TransportStats in transport.h. Perhaps TransportsStats? | |
| 120 struct SessionStats { | |
| 121 ProxyTransportMap proxy_to_transport; | |
| 122 TransportStatsMap transport_stats; | |
| 123 }; | |
| 124 | |
| 125 // A WebRtcSession manages general session state. This includes negotiation | |
| 126 // of both the application-level and network-level protocols: the former | |
| 127 // defines what will be sent and the latter defines how it will be sent. Each | |
| 128 // network-level protocol is represented by a Transport object. Each Transport | |
| 129 // participates in the network-level negotiation. The individual streams of | |
| 130 // packets are represented by TransportChannels. The application-level protocol | |
| 131 // is represented by SessionDecription objects. | |
| 132 class WebRtcSession : public AudioProviderInterface, | |
| 133 public VideoProviderInterface, | |
| 134 public DtmfProviderInterface, | |
| 135 public DataChannelProviderInterface, | |
| 136 public sigslot::has_slots<> { | |
| 137 public: | |
| 138 enum State { | |
| 139 STATE_INIT = 0, | |
| 140 STATE_SENTOFFER, // Sent offer, waiting for answer. | |
| 141 STATE_RECEIVEDOFFER, // Received an offer. Need to send answer. | |
| 142 STATE_SENTPRANSWER, // Sent provisional answer. Need to send answer. | |
| 143 STATE_RECEIVEDPRANSWER, // Received provisional answer, waiting for answer. | |
| 144 STATE_INPROGRESS, // Offer/answer exchange completed. | |
| 145 STATE_CLOSED, // Close() was called. | |
| 146 }; | |
| 147 | |
| 148 enum Error { | |
| 149 ERROR_NONE = 0, // no error | |
| 150 ERROR_CONTENT = 1, // channel errors in SetLocalContent/SetRemoteContent | |
| 151 ERROR_TRANSPORT = 2, // transport error of some kind | |
| 152 }; | |
| 153 | |
| 154 WebRtcSession(webrtc::MediaControllerInterface* media_controller, | |
| 155 rtc::Thread* signaling_thread, | |
| 156 rtc::Thread* worker_thread, | |
| 157 cricket::PortAllocator* port_allocator); | |
| 158 virtual ~WebRtcSession(); | |
| 159 | |
| 160 // These are const to allow them to be called from const methods. | |
| 161 rtc::Thread* signaling_thread() const { return signaling_thread_; } | |
| 162 rtc::Thread* worker_thread() const { return worker_thread_; } | |
| 163 cricket::PortAllocator* port_allocator() const { return port_allocator_; } | |
| 164 | |
| 165 // The ID of this session. | |
| 166 const std::string& id() const { return sid_; } | |
| 167 | |
| 168 bool Initialize( | |
| 169 const PeerConnectionFactoryInterface::Options& options, | |
| 170 const MediaConstraintsInterface* constraints, | |
| 171 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, | |
| 172 const PeerConnectionInterface::RTCConfiguration& rtc_configuration); | |
| 173 // Deletes the voice, video and data channel and changes the session state | |
| 174 // to STATE_CLOSED. | |
| 175 void Close(); | |
| 176 | |
| 177 // Returns true if we were the initial offerer. | |
| 178 bool initial_offerer() const { return initial_offerer_; } | |
| 179 | |
| 180 // Returns the current state of the session. See the enum above for details. | |
| 181 // Each time the state changes, we will fire this signal. | |
| 182 State state() const { return state_; } | |
| 183 sigslot::signal2<WebRtcSession*, State> SignalState; | |
| 184 | |
| 185 // Returns the last error in the session. See the enum above for details. | |
| 186 Error error() const { return error_; } | |
| 187 const std::string& error_desc() const { return error_desc_; } | |
| 188 | |
| 189 void RegisterIceObserver(IceObserver* observer) { | |
| 190 ice_observer_ = observer; | |
| 191 } | |
| 192 | |
| 193 virtual cricket::VoiceChannel* voice_channel() { | |
| 194 return voice_channel_.get(); | |
| 195 } | |
| 196 virtual cricket::VideoChannel* video_channel() { | |
| 197 return video_channel_.get(); | |
| 198 } | |
| 199 virtual cricket::DataChannel* data_channel() { | |
| 200 return data_channel_.get(); | |
| 201 } | |
| 202 | |
| 203 void SetSdesPolicy(cricket::SecurePolicy secure_policy); | |
| 204 cricket::SecurePolicy SdesPolicy() const; | |
| 205 | |
| 206 // Get current ssl role from transport. | |
| 207 bool GetSslRole(const std::string& transport_name, rtc::SSLRole* role); | |
| 208 | |
| 209 // Get current SSL role for this channel's transport. | |
| 210 // If |transport| is null, returns false. | |
| 211 bool GetSslRole(const cricket::BaseChannel* channel, rtc::SSLRole* role); | |
| 212 | |
| 213 void CreateOffer( | |
| 214 CreateSessionDescriptionObserver* observer, | |
| 215 const PeerConnectionInterface::RTCOfferAnswerOptions& options, | |
| 216 const cricket::MediaSessionOptions& session_options); | |
| 217 void CreateAnswer(CreateSessionDescriptionObserver* observer, | |
| 218 const MediaConstraintsInterface* constraints, | |
| 219 const cricket::MediaSessionOptions& session_options); | |
| 220 // The ownership of |desc| will be transferred after this call. | |
| 221 bool SetLocalDescription(SessionDescriptionInterface* desc, | |
| 222 std::string* err_desc); | |
| 223 // The ownership of |desc| will be transferred after this call. | |
| 224 bool SetRemoteDescription(SessionDescriptionInterface* desc, | |
| 225 std::string* err_desc); | |
| 226 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate); | |
| 227 | |
| 228 bool SetIceTransports(PeerConnectionInterface::IceTransportsType type); | |
| 229 | |
| 230 cricket::IceConfig ParseIceConfig( | |
| 231 const PeerConnectionInterface::RTCConfiguration& config) const; | |
| 232 | |
| 233 void SetIceConfig(const cricket::IceConfig& ice_config); | |
| 234 | |
| 235 // Start gathering candidates for any new transports, or transports doing an | |
| 236 // ICE restart. | |
| 237 void MaybeStartGathering(); | |
| 238 | |
| 239 const SessionDescriptionInterface* local_description() const { | |
| 240 return local_desc_.get(); | |
| 241 } | |
| 242 const SessionDescriptionInterface* remote_description() const { | |
| 243 return remote_desc_.get(); | |
| 244 } | |
| 245 | |
| 246 // Get the id used as a media stream track's "id" field from ssrc. | |
| 247 virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id); | |
| 248 virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id); | |
| 249 | |
| 250 // AudioMediaProviderInterface implementation. | |
| 251 void SetAudioPlayout(uint32_t ssrc, bool enable) override; | |
| 252 void SetAudioSend(uint32_t ssrc, | |
| 253 bool enable, | |
| 254 const cricket::AudioOptions& options, | |
| 255 cricket::AudioRenderer* renderer) override; | |
| 256 void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override; | |
| 257 void SetRawAudioSink(uint32_t ssrc, | |
| 258 rtc::scoped_ptr<AudioSinkInterface> sink) override; | |
| 259 | |
| 260 // Implements VideoMediaProviderInterface. | |
| 261 bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override; | |
| 262 void SetVideoPlayout(uint32_t ssrc, | |
| 263 bool enable, | |
| 264 cricket::VideoRenderer* renderer) override; | |
| 265 void SetVideoSend(uint32_t ssrc, | |
| 266 bool enable, | |
| 267 const cricket::VideoOptions* options) override; | |
| 268 | |
| 269 // Implements DtmfProviderInterface. | |
| 270 virtual bool CanInsertDtmf(const std::string& track_id); | |
| 271 virtual bool InsertDtmf(const std::string& track_id, | |
| 272 int code, int duration); | |
| 273 virtual sigslot::signal0<>* GetOnDestroyedSignal(); | |
| 274 | |
| 275 // Implements DataChannelProviderInterface. | |
| 276 bool SendData(const cricket::SendDataParams& params, | |
| 277 const rtc::Buffer& payload, | |
| 278 cricket::SendDataResult* result) override; | |
| 279 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override; | |
| 280 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override; | |
| 281 void AddSctpDataStream(int sid) override; | |
| 282 void RemoveSctpDataStream(int sid) override; | |
| 283 bool ReadyToSendData() const override; | |
| 284 | |
| 285 // Returns stats for all channels of all transports. | |
| 286 // This avoids exposing the internal structures used to track them. | |
| 287 virtual bool GetTransportStats(SessionStats* stats); | |
| 288 | |
| 289 // Get stats for a specific channel | |
| 290 bool GetChannelTransportStats(cricket::BaseChannel* ch, SessionStats* stats); | |
| 291 | |
| 292 // virtual so it can be mocked in unit tests | |
| 293 virtual bool GetLocalCertificate( | |
| 294 const std::string& transport_name, | |
| 295 rtc::scoped_refptr<rtc::RTCCertificate>* certificate); | |
| 296 | |
| 297 // Caller owns returned certificate | |
| 298 virtual bool GetRemoteSSLCertificate(const std::string& transport_name, | |
| 299 rtc::SSLCertificate** cert); | |
| 300 | |
| 301 cricket::DataChannelType data_channel_type() const; | |
| 302 | |
| 303 bool IceRestartPending() const; | |
| 304 | |
| 305 void ResetIceRestartLatch(); | |
| 306 | |
| 307 // Called when an RTCCertificate is generated or retrieved by | |
| 308 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription. | |
| 309 void OnCertificateReady( | |
| 310 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate); | |
| 311 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp); | |
| 312 | |
| 313 // For unit test. | |
| 314 bool waiting_for_certificate_for_testing() const; | |
| 315 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing(); | |
| 316 | |
| 317 void set_metrics_observer( | |
| 318 webrtc::MetricsObserverInterface* metrics_observer) { | |
| 319 metrics_observer_ = metrics_observer; | |
| 320 } | |
| 321 | |
| 322 // Called when voice_channel_, video_channel_ and data_channel_ are created | |
| 323 // and destroyed. As a result of, for example, setting a new description. | |
| 324 sigslot::signal0<> SignalVoiceChannelCreated; | |
| 325 sigslot::signal0<> SignalVoiceChannelDestroyed; | |
| 326 sigslot::signal0<> SignalVideoChannelCreated; | |
| 327 sigslot::signal0<> SignalVideoChannelDestroyed; | |
| 328 sigslot::signal0<> SignalDataChannelCreated; | |
| 329 sigslot::signal0<> SignalDataChannelDestroyed; | |
| 330 // Called when the whole session is destroyed. | |
| 331 sigslot::signal0<> SignalDestroyed; | |
| 332 | |
| 333 // Called when a valid data channel OPEN message is received. | |
| 334 // std::string represents the data channel label. | |
| 335 sigslot::signal2<const std::string&, const InternalDataChannelInit&> | |
| 336 SignalDataChannelOpenMessage; | |
| 337 | |
| 338 private: | |
| 339 // Indicates the type of SessionDescription in a call to SetLocalDescription | |
| 340 // and SetRemoteDescription. | |
| 341 enum Action { | |
| 342 kOffer, | |
| 343 kPrAnswer, | |
| 344 kAnswer, | |
| 345 }; | |
| 346 | |
| 347 // Log session state. | |
| 348 void LogState(State old_state, State new_state); | |
| 349 | |
| 350 // Updates the state, signaling if necessary. | |
| 351 virtual void SetState(State state); | |
| 352 | |
| 353 // Updates the error state, signaling if necessary. | |
| 354 // TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|. | |
| 355 virtual void SetError(Error error, const std::string& error_desc); | |
| 356 | |
| 357 bool UpdateSessionState(Action action, cricket::ContentSource source, | |
| 358 std::string* err_desc); | |
| 359 static Action GetAction(const std::string& type); | |
| 360 // Push the media parts of the local or remote session description | |
| 361 // down to all of the channels. | |
| 362 bool PushdownMediaDescription(cricket::ContentAction action, | |
| 363 cricket::ContentSource source, | |
| 364 std::string* error_desc); | |
| 365 | |
| 366 bool PushdownTransportDescription(cricket::ContentSource source, | |
| 367 cricket::ContentAction action, | |
| 368 std::string* error_desc); | |
| 369 | |
| 370 // Helper methods to push local and remote transport descriptions. | |
| 371 bool PushdownLocalTransportDescription( | |
| 372 const cricket::SessionDescription* sdesc, | |
| 373 cricket::ContentAction action, | |
| 374 std::string* error_desc); | |
| 375 bool PushdownRemoteTransportDescription( | |
| 376 const cricket::SessionDescription* sdesc, | |
| 377 cricket::ContentAction action, | |
| 378 std::string* error_desc); | |
| 379 | |
| 380 // Returns true and the TransportInfo of the given |content_name| | |
| 381 // from |description|. Returns false if it's not available. | |
| 382 static bool GetTransportDescription( | |
| 383 const cricket::SessionDescription* description, | |
| 384 const std::string& content_name, | |
| 385 cricket::TransportDescription* info); | |
| 386 | |
| 387 cricket::BaseChannel* GetChannel(const std::string& content_name); | |
| 388 // Cause all the BaseChannels in the bundle group to have the same | |
| 389 // transport channel. | |
| 390 bool EnableBundle(const cricket::ContentGroup& bundle); | |
| 391 | |
| 392 // Enables media channels to allow sending of media. | |
| 393 void EnableChannels(); | |
| 394 // Returns the media index for a local ice candidate given the content name. | |
| 395 // Returns false if the local session description does not have a media | |
| 396 // content called |content_name|. | |
| 397 bool GetLocalCandidateMediaIndex(const std::string& content_name, | |
| 398 int* sdp_mline_index); | |
| 399 // Uses all remote candidates in |remote_desc| in this session. | |
| 400 bool UseCandidatesInSessionDescription( | |
| 401 const SessionDescriptionInterface* remote_desc); | |
| 402 // Uses |candidate| in this session. | |
| 403 bool UseCandidate(const IceCandidateInterface* candidate); | |
| 404 // Deletes the corresponding channel of contents that don't exist in |desc|. | |
| 405 // |desc| can be null. This means that all channels are deleted. | |
| 406 void RemoveUnusedChannels(const cricket::SessionDescription* desc); | |
| 407 | |
| 408 // Allocates media channels based on the |desc|. If |desc| doesn't have | |
| 409 // the BUNDLE option, this method will disable BUNDLE in PortAllocator. | |
| 410 // This method will also delete any existing media channels before creating. | |
| 411 bool CreateChannels(const cricket::SessionDescription* desc); | |
| 412 | |
| 413 // Helper methods to create media channels. | |
| 414 bool CreateVoiceChannel(const cricket::ContentInfo* content); | |
| 415 bool CreateVideoChannel(const cricket::ContentInfo* content); | |
| 416 bool CreateDataChannel(const cricket::ContentInfo* content); | |
| 417 | |
| 418 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN | |
| 419 // messages. | |
| 420 void OnDataChannelMessageReceived(cricket::DataChannel* channel, | |
| 421 const cricket::ReceiveDataParams& params, | |
| 422 const rtc::Buffer& payload); | |
| 423 | |
| 424 std::string BadStateErrMsg(State state); | |
| 425 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state); | |
| 426 void SetIceConnectionReceiving(bool receiving); | |
| 427 | |
| 428 bool ValidateBundleSettings(const cricket::SessionDescription* desc); | |
| 429 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content); | |
| 430 // Below methods are helper methods which verifies SDP. | |
| 431 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc, | |
| 432 cricket::ContentSource source, | |
| 433 std::string* err_desc); | |
| 434 | |
| 435 // Check if a call to SetLocalDescription is acceptable with |action|. | |
| 436 bool ExpectSetLocalDescription(Action action); | |
| 437 // Check if a call to SetRemoteDescription is acceptable with |action|. | |
| 438 bool ExpectSetRemoteDescription(Action action); | |
| 439 // Verifies a=setup attribute as per RFC 5763. | |
| 440 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc, | |
| 441 Action action); | |
| 442 | |
| 443 // Returns true if we are ready to push down the remote candidate. | |
| 444 // |remote_desc| is the new remote description, or NULL if the current remote | |
| 445 // description should be used. Output |valid| is true if the candidate media | |
| 446 // index is valid. | |
| 447 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate, | |
| 448 const SessionDescriptionInterface* remote_desc, | |
| 449 bool* valid); | |
| 450 | |
| 451 void OnTransportControllerConnectionState(cricket::IceConnectionState state); | |
| 452 void OnTransportControllerReceiving(bool receiving); | |
| 453 void OnTransportControllerGatheringState(cricket::IceGatheringState state); | |
| 454 void OnTransportControllerCandidatesGathered( | |
| 455 const std::string& transport_name, | |
| 456 const cricket::Candidates& candidates); | |
| 457 | |
| 458 std::string GetSessionErrorMsg(); | |
| 459 | |
| 460 // Invoked when TransportController connection completion is signaled. | |
| 461 // Reports stats for all transports in use. | |
| 462 void ReportTransportStats(); | |
| 463 | |
| 464 // Gather the usage of IPv4/IPv6 as best connection. | |
| 465 void ReportBestConnectionState(const cricket::TransportStats& stats); | |
| 466 | |
| 467 void ReportNegotiatedCiphers(const cricket::TransportStats& stats); | |
| 468 | |
| 469 void OnSentPacket_w(cricket::TransportChannel* channel, | |
| 470 const rtc::SentPacket& sent_packet); | |
| 471 | |
| 472 rtc::Thread* const signaling_thread_; | |
| 473 rtc::Thread* const worker_thread_; | |
| 474 cricket::PortAllocator* const port_allocator_; | |
| 475 | |
| 476 State state_ = STATE_INIT; | |
| 477 Error error_ = ERROR_NONE; | |
| 478 std::string error_desc_; | |
| 479 | |
| 480 const std::string sid_; | |
| 481 bool initial_offerer_ = false; | |
| 482 | |
| 483 rtc::scoped_ptr<cricket::TransportController> transport_controller_; | |
| 484 MediaControllerInterface* media_controller_; | |
| 485 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_; | |
| 486 rtc::scoped_ptr<cricket::VideoChannel> video_channel_; | |
| 487 rtc::scoped_ptr<cricket::DataChannel> data_channel_; | |
| 488 cricket::ChannelManager* channel_manager_; | |
| 489 IceObserver* ice_observer_; | |
| 490 PeerConnectionInterface::IceConnectionState ice_connection_state_; | |
| 491 bool ice_connection_receiving_; | |
| 492 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_; | |
| 493 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_; | |
| 494 // If the remote peer is using a older version of implementation. | |
| 495 bool older_version_remote_peer_; | |
| 496 bool dtls_enabled_; | |
| 497 // Specifies which kind of data channel is allowed. This is controlled | |
| 498 // by the chrome command-line flag and constraints: | |
| 499 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled, | |
| 500 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is | |
| 501 // not set or false, SCTP is allowed (DCT_SCTP); | |
| 502 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP); | |
| 503 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE). | |
| 504 cricket::DataChannelType data_channel_type_; | |
| 505 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_; | |
| 506 | |
| 507 rtc::scoped_ptr<WebRtcSessionDescriptionFactory> | |
| 508 webrtc_session_desc_factory_; | |
| 509 | |
| 510 // Member variables for caching global options. | |
| 511 cricket::AudioOptions audio_options_; | |
| 512 cricket::VideoOptions video_options_; | |
| 513 MetricsObserverInterface* metrics_observer_; | |
| 514 | |
| 515 // Declares the bundle policy for the WebRTCSession. | |
| 516 PeerConnectionInterface::BundlePolicy bundle_policy_; | |
| 517 | |
| 518 // Declares the RTCP mux policy for the WebRTCSession. | |
| 519 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_; | |
| 520 | |
| 521 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession); | |
| 522 }; | |
| 523 } // namespace webrtc | |
| 524 | |
| 525 #endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_ | |
| OLD | NEW |
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Chromium Code Reviews
Issue 1610243002:
Move talk/app/webrtc to webrtc/api (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master