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henrike@webrtc.org28e20752013-07-10 00:45:361/*
2 * libjingle
jlmiller@webrtc.org5f93d0a2015-01-20 21:36:133 * Copyright 2012 Google Inc.
henrike@webrtc.org28e20752013-07-10 00:45:364 *
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>
deadbeefcbecd352015-09-23 18:50:2732#include <vector>
henrike@webrtc.org28e20752013-07-10 00:45:3633
buildbot@webrtc.orga09a9992014-08-13 17:26:0834#include "talk/app/webrtc/datachannel.h"
henrike@webrtc.org28e20752013-07-10 00:45:3635#include "talk/app/webrtc/dtmfsender.h"
Fredrik Solenberg709ed672015-09-15 10:26:3336#include "talk/app/webrtc/mediacontroller.h"
henrike@webrtc.org28e20752013-07-10 00:45:3637#include "talk/app/webrtc/mediastreamprovider.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:0838#include "talk/app/webrtc/peerconnectioninterface.h"
henrike@webrtc.org28e20752013-07-10 00:45:3639#include "talk/app/webrtc/statstypes.h"
henrike@webrtc.org28e20752013-07-10 00:45:3640#include "talk/media/base/mediachannel.h"
henrike@webrtc.org269fb4b2014-10-28 22:20:1141#include "webrtc/p2p/base/session.h"
henrike@webrtc.org28e20752013-07-10 00:45:3642#include "talk/session/media/mediasession.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:0843#include "webrtc/base/sigslot.h"
Henrik Boström5e56c592015-08-11 08:33:1344#include "webrtc/base/sslidentity.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:0845#include "webrtc/base/thread.h"
henrike@webrtc.org28e20752013-07-10 00:45:3646
47namespace cricket {
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:2848
wu@webrtc.org364f2042013-11-20 21:49:4149class BaseChannel;
henrike@webrtc.org28e20752013-07-10 00:45:3650class ChannelManager;
51class DataChannel;
52class StatsReport;
henrike@webrtc.org28e20752013-07-10 00:45:3653class VideoCapturer;
henrike@webrtc.org28e20752013-07-10 00:45:3654class VideoChannel;
55class VoiceChannel;
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:2856
henrike@webrtc.org28e20752013-07-10 00:45:3657} // namespace cricket
58
59namespace webrtc {
buildbot@webrtc.org41451d42014-05-03 05:39:4560
henrike@webrtc.org28e20752013-07-10 00:45:3661class IceRestartAnswerLatch;
buildbot@webrtc.org41451d42014-05-03 05:39:4562class JsepIceCandidate;
henrike@webrtc.org28e20752013-07-10 00:45:3663class MediaStreamSignaling;
wu@webrtc.org91053e72013-08-10 07:18:0464class WebRtcSessionDescriptionFactory;
henrike@webrtc.org28e20752013-07-10 00:45:3665
henrike@webrtc.org1e09a712013-07-26 19:17:5966extern const char kBundleWithoutRtcpMux[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:5467extern const char kCreateChannelFailed[];
henrike@webrtc.org28e20752013-07-10 00:45:3668extern const char kInvalidCandidates[];
69extern const char kInvalidSdp[];
70extern const char kMlineMismatch[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:5471extern const char kPushDownTDFailed[];
henrike@webrtc.orgb90991d2014-03-04 19:54:5772extern const char kSdpWithoutDtlsFingerprint[];
73extern const char kSdpWithoutSdesCrypto[];
mallinath@webrtc.org19f27e62013-10-13 17:18:2774extern const char kSdpWithoutIceUfragPwd[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:5475extern const char kSdpWithoutSdesAndDtlsDisabled[];
henrike@webrtc.org28e20752013-07-10 00:45:3676extern const char kSessionError[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:5477extern const char kSessionErrorDesc[];
pthatcher@webrtc.org4eeef582015-03-16 19:34:2378extern const char kDtlsSetupFailureRtp[];
79extern const char kDtlsSetupFailureRtcp[];
deadbeefcbecd352015-09-23 18:50:2780extern const char kEnableBundleFailed[];
81
buildbot@webrtc.org53df88c2014-08-07 22:46:0182// Maximum number of received video streams that will be processed by webrtc
83// even if they are not signalled beforehand.
84extern const int kMaxUnsignalledRecvStreams;
henrike@webrtc.org28e20752013-07-10 00:45:3685
86// ICE state callback interface.
87class IceObserver {
88 public:
wu@webrtc.org364f2042013-11-20 21:49:4189 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:3690 // Called any time the IceConnectionState changes
Peter Thatcher54360512015-07-08 18:08:3591 // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to
92 // conform to the w3c standard.
henrike@webrtc.org28e20752013-07-10 00:45:3693 virtual void OnIceConnectionChange(
94 PeerConnectionInterface::IceConnectionState new_state) {}
95 // Called any time the IceGatheringState changes
96 virtual void OnIceGatheringChange(
97 PeerConnectionInterface::IceGatheringState new_state) {}
98 // New Ice candidate have been found.
99 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
100 // All Ice candidates have been found.
101 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
102 // (via PeerConnectionObserver)
103 virtual void OnIceComplete() {}
104
Peter Thatcher54360512015-07-08 18:08:35105 // Called whenever the state changes between receiving and not receiving.
106 virtual void OnIceConnectionReceivingChange(bool receiving) {}
107
henrike@webrtc.org28e20752013-07-10 00:45:36108 protected:
109 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41110
111 private:
henrikg3c089d72015-09-16 12:37:44112 RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36113};
114
115class WebRtcSession : public cricket::BaseSession,
116 public AudioProviderInterface,
117 public DataChannelFactory,
118 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02119 public DtmfProviderInterface,
120 public DataChannelProviderInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36121 public:
122 WebRtcSession(cricket::ChannelManager* channel_manager,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52123 rtc::Thread* signaling_thread,
124 rtc::Thread* worker_thread,
henrike@webrtc.org28e20752013-07-10 00:45:36125 cricket::PortAllocator* port_allocator,
126 MediaStreamSignaling* mediastream_signaling);
127 virtual ~WebRtcSession();
128
Henrik Lundin64dad832015-05-11 10:44:23129 bool Initialize(
130 const PeerConnectionFactoryInterface::Options& options,
131 const MediaConstraintsInterface* constraints,
Henrik Boström5e56c592015-08-11 08:33:13132 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
Henrik Lundin64dad832015-05-11 10:44:23133 const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
henrike@webrtc.org28e20752013-07-10 00:45:36134 // Deletes the voice, video and data channel and changes the session state
135 // to STATE_RECEIVEDTERMINATE.
136 void Terminate();
137
138 void RegisterIceObserver(IceObserver* observer) {
139 ice_observer_ = observer;
140 }
141
142 virtual cricket::VoiceChannel* voice_channel() {
143 return voice_channel_.get();
144 }
145 virtual cricket::VideoChannel* video_channel() {
146 return video_channel_.get();
147 }
148 virtual cricket::DataChannel* data_channel() {
149 return data_channel_.get();
150 }
151
decurtis@webrtc.org487a4442015-01-15 22:55:07152 virtual const MediaStreamSignaling* mediastream_signaling() const {
153 return mediastream_signaling_;
154 }
155
henrike@webrtc.orgb90991d2014-03-04 19:54:57156 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
157 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36158
sergeyu@chromium.org0be6aa02013-08-23 23:21:25159 // Get current ssl role from transport.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52160 bool GetSslRole(rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25161
henrike@webrtc.org28e20752013-07-10 00:45:36162 // Generic error message callback from WebRtcSession.
163 // TODO - It may be necessary to supply error code as well.
164 sigslot::signal0<> SignalError;
165
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16166 void CreateOffer(
167 CreateSessionDescriptionObserver* observer,
168 const PeerConnectionInterface::RTCOfferAnswerOptions& options);
wu@webrtc.org91053e72013-08-10 07:18:04169 void CreateAnswer(CreateSessionDescriptionObserver* observer,
170 const MediaConstraintsInterface* constraints);
henrike@webrtc.org28654cb2013-07-22 21:07:49171 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36172 bool SetLocalDescription(SessionDescriptionInterface* desc,
173 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49174 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36175 bool SetRemoteDescription(SessionDescriptionInterface* desc,
176 std::string* err_desc);
177 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45178
mallinath@webrtc.org3d81b1b2014-09-09 14:38:10179 bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
buildbot@webrtc.org41451d42014-05-03 05:39:45180
henrike@webrtc.org28e20752013-07-10 00:45:36181 const SessionDescriptionInterface* local_description() const {
182 return local_desc_.get();
183 }
184 const SessionDescriptionInterface* remote_description() const {
185 return remote_desc_.get();
186 }
pthatcher@webrtc.org592470b2015-03-16 21:15:37187 // TODO(pthatcher): Cleanup the distinction between
188 // SessionDescription and SessionDescriptionInterface and remove
189 // these if possible.
190 const cricket::SessionDescription* base_local_description() const {
191 return BaseSession::local_description();
192 }
193 const cricket::SessionDescription* base_remote_description() const {
194 return BaseSession::remote_description();
195 }
henrike@webrtc.org28e20752013-07-10 00:45:36196
197 // Get the id used as a media stream track's "id" field from ssrc.
xians@webrtc.org4cb01282014-06-12 14:57:05198 virtual bool GetLocalTrackIdBySsrc(uint32 ssrc, std::string* track_id);
199 virtual bool GetRemoteTrackIdBySsrc(uint32 ssrc, std::string* track_id);
200
henrike@webrtc.org28e20752013-07-10 00:45:36201 // AudioMediaProviderInterface implementation.
kjellander@webrtc.org14665ff2015-03-04 12:58:35202 void SetAudioPlayout(uint32 ssrc,
203 bool enable,
204 cricket::AudioRenderer* renderer) override;
205 void SetAudioSend(uint32 ssrc,
206 bool enable,
207 const cricket::AudioOptions& options,
208 cricket::AudioRenderer* renderer) override;
209 void SetAudioPlayoutVolume(uint32 ssrc, double volume) override;
henrike@webrtc.org28e20752013-07-10 00:45:36210
211 // Implements VideoMediaProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35212 bool SetCaptureDevice(uint32 ssrc, cricket::VideoCapturer* camera) override;
213 void SetVideoPlayout(uint32 ssrc,
214 bool enable,
215 cricket::VideoRenderer* renderer) override;
216 void SetVideoSend(uint32 ssrc,
217 bool enable,
218 const cricket::VideoOptions* options) override;
henrike@webrtc.org28e20752013-07-10 00:45:36219
220 // Implements DtmfProviderInterface.
221 virtual bool CanInsertDtmf(const std::string& track_id);
222 virtual bool InsertDtmf(const std::string& track_id,
223 int code, int duration);
224 virtual sigslot::signal0<>* GetOnDestroyedSignal();
225
wu@webrtc.org78187522013-10-07 23:32:02226 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35227 bool SendData(const cricket::SendDataParams& params,
228 const rtc::Buffer& payload,
229 cricket::SendDataResult* result) override;
230 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
231 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
232 void AddSctpDataStream(int sid) override;
233 void RemoveSctpDataStream(int sid) override;
234 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02235
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40236 // Returns stats for all channels of all transports.
237 // This avoids exposing the internal structures used to track them.
238 virtual bool GetTransportStats(cricket::SessionStats* stats);
239
deadbeefcbecd352015-09-23 18:50:27240 // Get stats for a specific channel
241 bool GetChannelTransportStats(cricket::BaseChannel* ch,
242 cricket::SessionStats* stats);
243
244 // virtual so it can be mocked in unit tests
245 virtual bool GetLocalCertificate(
246 const std::string& transport_name,
247 rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
248
249 // Caller owns returned certificate
250 virtual bool GetRemoteSSLCertificate(const std::string& transport_name,
251 rtc::SSLCertificate** cert);
252
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58253 // Implements DataChannelFactory.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52254 rtc::scoped_refptr<DataChannel> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36255 const std::string& label,
kjellander@webrtc.org14665ff2015-03-04 12:58:35256 const InternalDataChannelInit* config) override;
henrike@webrtc.org28e20752013-07-10 00:45:36257
258 cricket::DataChannelType data_channel_type() const;
259
wu@webrtc.org91053e72013-08-10 07:18:04260 bool IceRestartPending() const;
261
262 void ResetIceRestartLatch();
263
Henrik Boströmd8281982015-08-27 08:12:24264 // Called when an RTCCertificate is generated or retrieved by
wu@webrtc.org91053e72013-08-10 07:18:04265 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
Henrik Boströmd8281982015-08-27 08:12:24266 void OnCertificateReady(
267 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
pthatcher@webrtc.org4eeef582015-03-16 19:34:23268 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
wu@webrtc.org91053e72013-08-10 07:18:04269
270 // For unit test.
Henrik Boströmd8281982015-08-27 08:12:24271 bool waiting_for_certificate_for_testing() const;
deadbeefcbecd352015-09-23 18:50:27272 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
wu@webrtc.org91053e72013-08-10 07:18:04273
guoweis@webrtc.org7169afd2014-12-04 17:59:29274 void set_metrics_observer(
275 webrtc::MetricsObserverInterface* metrics_observer) {
276 metrics_observer_ = metrics_observer;
277 }
278
henrike@webrtc.org28e20752013-07-10 00:45:36279 private:
280 // Indicates the type of SessionDescription in a call to SetLocalDescription
281 // and SetRemoteDescription.
282 enum Action {
283 kOffer,
284 kPrAnswer,
285 kAnswer,
286 };
wu@webrtc.org91053e72013-08-10 07:18:04287
henrike@webrtc.org28e20752013-07-10 00:45:36288 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36289 std::string* err_desc);
290 static Action GetAction(const std::string& type);
pthatcher@webrtc.org592470b2015-03-16 21:15:37291 // Push the media parts of the local or remote session description
292 // down to all of the channels.
293 bool PushdownMediaDescription(cricket::ContentAction action,
294 cricket::ContentSource source,
295 std::string* error_desc);
296
deadbeefcbecd352015-09-23 18:50:27297 cricket::BaseChannel* GetChannel(const std::string& content_name);
298 // Cause all the BaseChannels in the bundle group to have the same
299 // transport channel.
300 bool EnableBundle(const cricket::ContentGroup& bundle);
henrike@webrtc.org28e20752013-07-10 00:45:36301
henrike@webrtc.org28e20752013-07-10 00:45:36302 // Enables media channels to allow sending of media.
303 void EnableChannels();
henrike@webrtc.org28e20752013-07-10 00:45:36304 // Returns the media index for a local ice candidate given the content name.
305 // Returns false if the local session description does not have a media
306 // content called |content_name|.
307 bool GetLocalCandidateMediaIndex(const std::string& content_name,
308 int* sdp_mline_index);
309 // Uses all remote candidates in |remote_desc| in this session.
310 bool UseCandidatesInSessionDescription(
311 const SessionDescriptionInterface* remote_desc);
312 // Uses |candidate| in this session.
313 bool UseCandidate(const IceCandidateInterface* candidate);
314 // Deletes the corresponding channel of contents that don't exist in |desc|.
315 // |desc| can be null. This means that all channels are deleted.
deadbeefcbecd352015-09-23 18:50:27316 void RemoveUnusedChannels(const cricket::SessionDescription* desc);
henrike@webrtc.org28e20752013-07-10 00:45:36317
318 // Allocates media channels based on the |desc|. If |desc| doesn't have
319 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
320 // This method will also delete any existing media channels before creating.
321 bool CreateChannels(const cricket::SessionDescription* desc);
322
323 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59324 bool CreateVoiceChannel(const cricket::ContentInfo* content);
325 bool CreateVideoChannel(const cricket::ContentInfo* content);
326 bool CreateDataChannel(const cricket::ContentInfo* content);
327
henrike@webrtc.org28e20752013-07-10 00:45:36328 // Copy the candidates from |saved_candidates_| to |dest_desc|.
329 // The |saved_candidates_| will be cleared after this function call.
330 void CopySavedCandidates(SessionDescriptionInterface* dest_desc);
331
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58332 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
333 // messages.
334 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
335 const cricket::ReceiveDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52336 const rtc::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36337
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54338 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36339 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
Peter Thatcher54360512015-07-08 18:08:35340 void SetIceConnectionReceiving(bool receiving);
henrike@webrtc.org28e20752013-07-10 00:45:36341
sergeyu@chromium.org0be6aa02013-08-23 23:21:25342 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59343 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25344 // Below methods are helper methods which verifies SDP.
345 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
346 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54347 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25348
349 // Check if a call to SetLocalDescription is acceptable with |action|.
350 bool ExpectSetLocalDescription(Action action);
351 // Check if a call to SetRemoteDescription is acceptable with |action|.
352 bool ExpectSetRemoteDescription(Action action);
353 // Verifies a=setup attribute as per RFC 5763.
354 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
355 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59356
jiayl@webrtc.orge10d28c2014-07-17 17:07:49357 // Returns true if we are ready to push down the remote candidate.
358 // |remote_desc| is the new remote description, or NULL if the current remote
359 // description should be used. Output |valid| is true if the candidate media
360 // index is valid.
361 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
362 const SessionDescriptionInterface* remote_desc,
363 bool* valid);
364
deadbeefcbecd352015-09-23 18:50:27365 void OnTransportControllerConnectionState(cricket::IceConnectionState state);
366 void OnTransportControllerReceiving(bool receiving);
367 void OnTransportControllerGatheringState(cricket::IceGatheringState state);
368 void OnTransportControllerCandidatesGathered(
369 const std::string& transport_name,
370 const cricket::Candidates& candidates);
371
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54372 std::string GetSessionErrorMsg();
373
deadbeefcbecd352015-09-23 18:50:27374 // Invoked when TransportController connection completion is signaled.
375 // Reports stats for all transports in use.
376 void ReportTransportStats();
377
378 // Gather the usage of IPv4/IPv6 as best connection.
jbauchac8869e2015-07-03 08:36:14379 void ReportBestConnectionState(const cricket::TransportStats& stats);
380
381 void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
guoweis@webrtc.org7169afd2014-12-04 17:59:29382
Fredrik Solenberg709ed672015-09-15 10:26:33383 rtc::scoped_ptr<MediaControllerInterface> media_controller_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52384 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
385 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
386 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36387 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36388 MediaStreamSignaling* mediastream_signaling_;
389 IceObserver* ice_observer_;
390 PeerConnectionInterface::IceConnectionState ice_connection_state_;
Peter Thatcher54360512015-07-08 18:08:35391 bool ice_connection_receiving_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52392 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
393 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36394 // Candidates that arrived before the remote description was set.
395 std::vector<IceCandidateInterface*> saved_candidates_;
henrike@webrtc.org28e20752013-07-10 00:45:36396 // If the remote peer is using a older version of implementation.
397 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10398 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36399 // Specifies which kind of data channel is allowed. This is controlled
400 // by the chrome command-line flag and constraints:
401 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
402 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
403 // not set or false, SCTP is allowed (DCT_SCTP);
404 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
405 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
406 cricket::DataChannelType data_channel_type_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52407 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04408
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52409 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04410 webrtc_session_desc_factory_;
411
henrike@webrtc.org28e20752013-07-10 00:45:36412 sigslot::signal0<> SignalVoiceChannelDestroyed;
413 sigslot::signal0<> SignalVideoChannelDestroyed;
414 sigslot::signal0<> SignalDataChannelDestroyed;
henrike@webrtc.org28e20752013-07-10 00:45:36415
henrike@webrtc.org6e3dbc22014-03-25 17:09:47416 // Member variables for caching global options.
417 cricket::AudioOptions audio_options_;
418 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29419 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47420
pthatcher@webrtc.org877ac762015-02-04 22:03:09421 // Declares the bundle policy for the WebRTCSession.
422 PeerConnectionInterface::BundlePolicy bundle_policy_;
423
Peter Thatcheraf55ccc2015-05-21 14:48:41424 // Declares the RTCP mux policy for the WebRTCSession.
425 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
426
henrikg3c089d72015-09-16 12:37:44427 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
wu@webrtc.org364f2042013-11-20 21:49:41428};
henrike@webrtc.org28e20752013-07-10 00:45:36429} // namespace webrtc
430
431#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_