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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
jlmiller@webrtc.org5f93d0a2015-01-20 21:36:13 +00003 * Copyright 2012 Google Inc.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00004 *
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
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000033#include "talk/app/webrtc/datachannel.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000034#include "talk/app/webrtc/dtmfsender.h"
35#include "talk/app/webrtc/mediastreamprovider.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000036#include "talk/app/webrtc/peerconnectioninterface.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000037#include "talk/app/webrtc/statstypes.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000038#include "talk/media/base/mediachannel.h"
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000039#include "webrtc/p2p/base/session.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000040#include "talk/session/media/mediasession.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000041#include "webrtc/base/sigslot.h"
42#include "webrtc/base/thread.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000043
44namespace cricket {
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000045
wu@webrtc.org364f2042013-11-20 21:49:41 +000046class BaseChannel;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000047class ChannelManager;
48class DataChannel;
49class StatsReport;
50class Transport;
51class VideoCapturer;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000052class VideoChannel;
53class VoiceChannel;
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000054
henrike@webrtc.org28e20752013-07-10 00:45:36 +000055} // namespace cricket
56
57namespace webrtc {
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000058
henrike@webrtc.org28e20752013-07-10 00:45:36 +000059class IceRestartAnswerLatch;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000060class JsepIceCandidate;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000061class MediaStreamSignaling;
wu@webrtc.org91053e72013-08-10 07:18:04 +000062class WebRtcSessionDescriptionFactory;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000063
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000064extern const char kBundleWithoutRtcpMux[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000065extern const char kCreateChannelFailed[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000066extern const char kInvalidCandidates[];
67extern const char kInvalidSdp[];
68extern const char kMlineMismatch[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000069extern const char kPushDownTDFailed[];
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +000070extern const char kSdpWithoutDtlsFingerprint[];
71extern const char kSdpWithoutSdesCrypto[];
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000072extern const char kSdpWithoutIceUfragPwd[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000073extern const char kSdpWithoutSdesAndDtlsDisabled[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000074extern const char kSessionError[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000075extern const char kSessionErrorDesc[];
buildbot@webrtc.org53df88c2014-08-07 22:46:01 +000076// Maximum number of received video streams that will be processed by webrtc
77// even if they are not signalled beforehand.
78extern const int kMaxUnsignalledRecvStreams;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000079
80// ICE state callback interface.
81class IceObserver {
82 public:
wu@webrtc.org364f2042013-11-20 21:49:41 +000083 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +000084 // Called any time the IceConnectionState changes
85 virtual void OnIceConnectionChange(
86 PeerConnectionInterface::IceConnectionState new_state) {}
87 // Called any time the IceGatheringState changes
88 virtual void OnIceGatheringChange(
89 PeerConnectionInterface::IceGatheringState new_state) {}
90 // New Ice candidate have been found.
91 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
92 // All Ice candidates have been found.
93 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
94 // (via PeerConnectionObserver)
95 virtual void OnIceComplete() {}
96
97 protected:
98 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +000099
100 private:
101 DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000102};
103
104class WebRtcSession : public cricket::BaseSession,
105 public AudioProviderInterface,
106 public DataChannelFactory,
107 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000108 public DtmfProviderInterface,
109 public DataChannelProviderInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000110 public:
111 WebRtcSession(cricket::ChannelManager* channel_manager,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000112 rtc::Thread* signaling_thread,
113 rtc::Thread* worker_thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000114 cricket::PortAllocator* port_allocator,
115 MediaStreamSignaling* mediastream_signaling);
116 virtual ~WebRtcSession();
117
wu@webrtc.org97077a32013-10-25 21:18:33 +0000118 bool Initialize(const PeerConnectionFactoryInterface::Options& options,
119 const MediaConstraintsInterface* constraints,
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000120 DTLSIdentityServiceInterface* dtls_identity_service,
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000121 PeerConnectionInterface::IceTransportsType ice_transport_type,
122 PeerConnectionInterface::BundlePolicy bundle_policy);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000123 // Deletes the voice, video and data channel and changes the session state
124 // to STATE_RECEIVEDTERMINATE.
125 void Terminate();
126
127 void RegisterIceObserver(IceObserver* observer) {
128 ice_observer_ = observer;
129 }
130
131 virtual cricket::VoiceChannel* voice_channel() {
132 return voice_channel_.get();
133 }
134 virtual cricket::VideoChannel* video_channel() {
135 return video_channel_.get();
136 }
137 virtual cricket::DataChannel* data_channel() {
138 return data_channel_.get();
139 }
140
decurtis@webrtc.org487a4442015-01-15 22:55:07 +0000141 virtual const MediaStreamSignaling* mediastream_signaling() const {
142 return mediastream_signaling_;
143 }
144
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000145 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
146 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000147
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000148 // Get current ssl role from transport.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000149 bool GetSslRole(rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000150
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000151 // Generic error message callback from WebRtcSession.
152 // TODO - It may be necessary to supply error code as well.
153 sigslot::signal0<> SignalError;
154
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000155 void CreateOffer(
156 CreateSessionDescriptionObserver* observer,
157 const PeerConnectionInterface::RTCOfferAnswerOptions& options);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000158 void CreateAnswer(CreateSessionDescriptionObserver* observer,
159 const MediaConstraintsInterface* constraints);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000160 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000161 bool SetLocalDescription(SessionDescriptionInterface* desc,
162 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000163 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000164 bool SetRemoteDescription(SessionDescriptionInterface* desc,
165 std::string* err_desc);
166 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000167
mallinath@webrtc.org3d81b1b2014-09-09 14:38:10 +0000168 bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000169
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000170 const SessionDescriptionInterface* local_description() const {
171 return local_desc_.get();
172 }
173 const SessionDescriptionInterface* remote_description() const {
174 return remote_desc_.get();
175 }
176
177 // Get the id used as a media stream track's "id" field from ssrc.
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000178 virtual bool GetLocalTrackIdBySsrc(uint32 ssrc, std::string* track_id);
179 virtual bool GetRemoteTrackIdBySsrc(uint32 ssrc, std::string* track_id);
180
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000181
182 // AudioMediaProviderInterface implementation.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000183 void SetAudioPlayout(uint32 ssrc,
184 bool enable,
185 cricket::AudioRenderer* renderer) override;
186 void SetAudioSend(uint32 ssrc,
187 bool enable,
188 const cricket::AudioOptions& options,
189 cricket::AudioRenderer* renderer) override;
190 void SetAudioPlayoutVolume(uint32 ssrc, double volume) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000191
192 // Implements VideoMediaProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000193 bool SetCaptureDevice(uint32 ssrc, cricket::VideoCapturer* camera) override;
194 void SetVideoPlayout(uint32 ssrc,
195 bool enable,
196 cricket::VideoRenderer* renderer) override;
197 void SetVideoSend(uint32 ssrc,
198 bool enable,
199 const cricket::VideoOptions* options) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000200
201 // Implements DtmfProviderInterface.
202 virtual bool CanInsertDtmf(const std::string& track_id);
203 virtual bool InsertDtmf(const std::string& track_id,
204 int code, int duration);
205 virtual sigslot::signal0<>* GetOnDestroyedSignal();
206
wu@webrtc.org78187522013-10-07 23:32:02 +0000207 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000208 bool SendData(const cricket::SendDataParams& params,
209 const rtc::Buffer& payload,
210 cricket::SendDataResult* result) override;
211 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
212 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
213 void AddSctpDataStream(int sid) override;
214 void RemoveSctpDataStream(int sid) override;
215 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02 +0000216
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000217 // Returns stats for all channels of all transports.
218 // This avoids exposing the internal structures used to track them.
219 virtual bool GetTransportStats(cricket::SessionStats* stats);
220
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000221 // Implements DataChannelFactory.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000222 rtc::scoped_refptr<DataChannel> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000223 const std::string& label,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000224 const InternalDataChannelInit* config) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000225
226 cricket::DataChannelType data_channel_type() const;
227
wu@webrtc.org91053e72013-08-10 07:18:04 +0000228 bool IceRestartPending() const;
229
230 void ResetIceRestartLatch();
231
232 // Called when an SSLIdentity is generated or retrieved by
233 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000234 void OnIdentityReady(rtc::SSLIdentity* identity);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000235
236 // For unit test.
237 bool waiting_for_identity() const;
238
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000239 void set_metrics_observer(
240 webrtc::MetricsObserverInterface* metrics_observer) {
241 metrics_observer_ = metrics_observer;
242 }
243
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000244 private:
245 // Indicates the type of SessionDescription in a call to SetLocalDescription
246 // and SetRemoteDescription.
247 enum Action {
248 kOffer,
249 kPrAnswer,
250 kAnswer,
251 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000252
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000253 // Invokes ConnectChannels() on transport proxies, which initiates ice
254 // candidates allocation.
255 bool StartCandidatesAllocation();
256 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000257 std::string* err_desc);
258 static Action GetAction(const std::string& type);
259
260 // Transport related callbacks, override from cricket::BaseSession.
261 virtual void OnTransportRequestSignaling(cricket::Transport* transport);
262 virtual void OnTransportConnecting(cricket::Transport* transport);
263 virtual void OnTransportWritable(cricket::Transport* transport);
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000264 virtual void OnTransportCompleted(cricket::Transport* transport);
265 virtual void OnTransportFailed(cricket::Transport* transport);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000266 virtual void OnTransportProxyCandidatesReady(
267 cricket::TransportProxy* proxy,
268 const cricket::Candidates& candidates);
269 virtual void OnCandidatesAllocationDone();
270
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000271 // Creates local session description with audio and video contents.
272 bool CreateDefaultLocalDescription();
273 // Enables media channels to allow sending of media.
274 void EnableChannels();
275 // Creates a JsepIceCandidate and adds it to the local session description
276 // and notify observers. Called when a new local candidate have been found.
277 void ProcessNewLocalCandidate(const std::string& content_name,
278 const cricket::Candidates& candidates);
279 // Returns the media index for a local ice candidate given the content name.
280 // Returns false if the local session description does not have a media
281 // content called |content_name|.
282 bool GetLocalCandidateMediaIndex(const std::string& content_name,
283 int* sdp_mline_index);
284 // Uses all remote candidates in |remote_desc| in this session.
285 bool UseCandidatesInSessionDescription(
286 const SessionDescriptionInterface* remote_desc);
287 // Uses |candidate| in this session.
288 bool UseCandidate(const IceCandidateInterface* candidate);
289 // Deletes the corresponding channel of contents that don't exist in |desc|.
290 // |desc| can be null. This means that all channels are deleted.
291 void RemoveUnusedChannelsAndTransports(
292 const cricket::SessionDescription* desc);
293
294 // Allocates media channels based on the |desc|. If |desc| doesn't have
295 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
296 // This method will also delete any existing media channels before creating.
297 bool CreateChannels(const cricket::SessionDescription* desc);
298
299 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000300 bool CreateVoiceChannel(const cricket::ContentInfo* content);
301 bool CreateVideoChannel(const cricket::ContentInfo* content);
302 bool CreateDataChannel(const cricket::ContentInfo* content);
303
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000304 // Copy the candidates from |saved_candidates_| to |dest_desc|.
305 // The |saved_candidates_| will be cleared after this function call.
306 void CopySavedCandidates(SessionDescriptionInterface* dest_desc);
307
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000308 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
309 // messages.
310 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
311 const cricket::ReceiveDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000312 const rtc::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000313
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000314 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000315 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
316
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000317 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000318 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000319 // Below methods are helper methods which verifies SDP.
320 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
321 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000322 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000323
324 // Check if a call to SetLocalDescription is acceptable with |action|.
325 bool ExpectSetLocalDescription(Action action);
326 // Check if a call to SetRemoteDescription is acceptable with |action|.
327 bool ExpectSetRemoteDescription(Action action);
328 // Verifies a=setup attribute as per RFC 5763.
329 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
330 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000331
jiayl@webrtc.orge10d28c2014-07-17 17:07:49 +0000332 // Returns true if we are ready to push down the remote candidate.
333 // |remote_desc| is the new remote description, or NULL if the current remote
334 // description should be used. Output |valid| is true if the candidate media
335 // index is valid.
336 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
337 const SessionDescriptionInterface* remote_desc,
338 bool* valid);
339
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000340 std::string GetSessionErrorMsg();
341
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000342 // Invoked when OnTransportCompleted is signaled to gather the usage
343 // of IPv4/IPv6 as best connection.
344 void ReportBestConnectionState(cricket::Transport* transport);
345
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000346 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
347 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
348 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000349 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000350 MediaStreamSignaling* mediastream_signaling_;
351 IceObserver* ice_observer_;
352 PeerConnectionInterface::IceConnectionState ice_connection_state_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000353 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
354 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000355 // Candidates that arrived before the remote description was set.
356 std::vector<IceCandidateInterface*> saved_candidates_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000357 // If the remote peer is using a older version of implementation.
358 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000359 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000360 // Specifies which kind of data channel is allowed. This is controlled
361 // by the chrome command-line flag and constraints:
362 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
363 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
364 // not set or false, SCTP is allowed (DCT_SCTP);
365 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
366 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
367 cricket::DataChannelType data_channel_type_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000368 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000369
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000370 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04 +0000371 webrtc_session_desc_factory_;
372
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000373 sigslot::signal0<> SignalVoiceChannelDestroyed;
374 sigslot::signal0<> SignalVideoChannelDestroyed;
375 sigslot::signal0<> SignalDataChannelDestroyed;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000376
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000377 // Member variables for caching global options.
378 cricket::AudioOptions audio_options_;
379 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000380 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000381
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000382 // Declares the bundle policy for the WebRTCSession.
383 PeerConnectionInterface::BundlePolicy bundle_policy_;
384
wu@webrtc.org364f2042013-11-20 21:49:41 +0000385 DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
386};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000387} // namespace webrtc
388
389#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_