blob: 14e4414053e530f02d20a8d392f0454bb4447c85 [file] [log] [blame]
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[];
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +000076extern const char kDtlsSetupFailureRtp[];
77extern const char kDtlsSetupFailureRtcp[];
buildbot@webrtc.org53df88c2014-08-07 22:46:01 +000078// Maximum number of received video streams that will be processed by webrtc
79// even if they are not signalled beforehand.
80extern const int kMaxUnsignalledRecvStreams;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000081
82// ICE state callback interface.
83class IceObserver {
84 public:
wu@webrtc.org364f2042013-11-20 21:49:41 +000085 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +000086 // Called any time the IceConnectionState changes
87 virtual void OnIceConnectionChange(
88 PeerConnectionInterface::IceConnectionState new_state) {}
89 // Called any time the IceGatheringState changes
90 virtual void OnIceGatheringChange(
91 PeerConnectionInterface::IceGatheringState new_state) {}
92 // New Ice candidate have been found.
93 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
94 // All Ice candidates have been found.
95 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
96 // (via PeerConnectionObserver)
97 virtual void OnIceComplete() {}
98
99 protected:
100 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +0000101
102 private:
103 DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000104};
105
106class WebRtcSession : public cricket::BaseSession,
107 public AudioProviderInterface,
108 public DataChannelFactory,
109 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000110 public DtmfProviderInterface,
111 public DataChannelProviderInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000112 public:
113 WebRtcSession(cricket::ChannelManager* channel_manager,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000114 rtc::Thread* signaling_thread,
115 rtc::Thread* worker_thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000116 cricket::PortAllocator* port_allocator,
117 MediaStreamSignaling* mediastream_signaling);
118 virtual ~WebRtcSession();
119
wu@webrtc.org97077a32013-10-25 21:18:33 +0000120 bool Initialize(const PeerConnectionFactoryInterface::Options& options,
121 const MediaConstraintsInterface* constraints,
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000122 DTLSIdentityServiceInterface* dtls_identity_service,
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000123 PeerConnectionInterface::IceTransportsType ice_transport_type,
124 PeerConnectionInterface::BundlePolicy bundle_policy);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000125 // Deletes the voice, video and data channel and changes the session state
126 // to STATE_RECEIVEDTERMINATE.
127 void Terminate();
128
129 void RegisterIceObserver(IceObserver* observer) {
130 ice_observer_ = observer;
131 }
132
133 virtual cricket::VoiceChannel* voice_channel() {
134 return voice_channel_.get();
135 }
136 virtual cricket::VideoChannel* video_channel() {
137 return video_channel_.get();
138 }
139 virtual cricket::DataChannel* data_channel() {
140 return data_channel_.get();
141 }
142
decurtis@webrtc.org487a4442015-01-15 22:55:07 +0000143 virtual const MediaStreamSignaling* mediastream_signaling() const {
144 return mediastream_signaling_;
145 }
146
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000147 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
148 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000149
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000150 // Get current ssl role from transport.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000151 bool GetSslRole(rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000152
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000153 // Generic error message callback from WebRtcSession.
154 // TODO - It may be necessary to supply error code as well.
155 sigslot::signal0<> SignalError;
156
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000157 void CreateOffer(
158 CreateSessionDescriptionObserver* observer,
159 const PeerConnectionInterface::RTCOfferAnswerOptions& options);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000160 void CreateAnswer(CreateSessionDescriptionObserver* observer,
161 const MediaConstraintsInterface* constraints);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000162 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000163 bool SetLocalDescription(SessionDescriptionInterface* desc,
164 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000165 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000166 bool SetRemoteDescription(SessionDescriptionInterface* desc,
167 std::string* err_desc);
168 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000169
mallinath@webrtc.org3d81b1b2014-09-09 14:38:10 +0000170 bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000171
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000172 const SessionDescriptionInterface* local_description() const {
173 return local_desc_.get();
174 }
175 const SessionDescriptionInterface* remote_description() const {
176 return remote_desc_.get();
177 }
178
179 // Get the id used as a media stream track's "id" field from ssrc.
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000180 virtual bool GetLocalTrackIdBySsrc(uint32 ssrc, std::string* track_id);
181 virtual bool GetRemoteTrackIdBySsrc(uint32 ssrc, std::string* track_id);
182
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000183
184 // AudioMediaProviderInterface implementation.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000185 void SetAudioPlayout(uint32 ssrc,
186 bool enable,
187 cricket::AudioRenderer* renderer) override;
188 void SetAudioSend(uint32 ssrc,
189 bool enable,
190 const cricket::AudioOptions& options,
191 cricket::AudioRenderer* renderer) override;
192 void SetAudioPlayoutVolume(uint32 ssrc, double volume) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000193
194 // Implements VideoMediaProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000195 bool SetCaptureDevice(uint32 ssrc, cricket::VideoCapturer* camera) override;
196 void SetVideoPlayout(uint32 ssrc,
197 bool enable,
198 cricket::VideoRenderer* renderer) override;
199 void SetVideoSend(uint32 ssrc,
200 bool enable,
201 const cricket::VideoOptions* options) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000202
203 // Implements DtmfProviderInterface.
204 virtual bool CanInsertDtmf(const std::string& track_id);
205 virtual bool InsertDtmf(const std::string& track_id,
206 int code, int duration);
207 virtual sigslot::signal0<>* GetOnDestroyedSignal();
208
wu@webrtc.org78187522013-10-07 23:32:02 +0000209 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000210 bool SendData(const cricket::SendDataParams& params,
211 const rtc::Buffer& payload,
212 cricket::SendDataResult* result) override;
213 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
214 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
215 void AddSctpDataStream(int sid) override;
216 void RemoveSctpDataStream(int sid) override;
217 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02 +0000218
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000219 // Returns stats for all channels of all transports.
220 // This avoids exposing the internal structures used to track them.
221 virtual bool GetTransportStats(cricket::SessionStats* stats);
222
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000223 // Implements DataChannelFactory.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000224 rtc::scoped_refptr<DataChannel> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000225 const std::string& label,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000226 const InternalDataChannelInit* config) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000227
228 cricket::DataChannelType data_channel_type() const;
229
wu@webrtc.org91053e72013-08-10 07:18:04 +0000230 bool IceRestartPending() const;
231
232 void ResetIceRestartLatch();
233
234 // Called when an SSLIdentity is generated or retrieved by
235 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000236 void OnIdentityReady(rtc::SSLIdentity* identity);
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +0000237 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000238
239 // For unit test.
240 bool waiting_for_identity() const;
241
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000242 void set_metrics_observer(
243 webrtc::MetricsObserverInterface* metrics_observer) {
244 metrics_observer_ = metrics_observer;
245 }
246
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000247 private:
248 // Indicates the type of SessionDescription in a call to SetLocalDescription
249 // and SetRemoteDescription.
250 enum Action {
251 kOffer,
252 kPrAnswer,
253 kAnswer,
254 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000255
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000256 // Invokes ConnectChannels() on transport proxies, which initiates ice
257 // candidates allocation.
258 bool StartCandidatesAllocation();
259 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000260 std::string* err_desc);
261 static Action GetAction(const std::string& type);
262
263 // Transport related callbacks, override from cricket::BaseSession.
264 virtual void OnTransportRequestSignaling(cricket::Transport* transport);
265 virtual void OnTransportConnecting(cricket::Transport* transport);
266 virtual void OnTransportWritable(cricket::Transport* transport);
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000267 virtual void OnTransportCompleted(cricket::Transport* transport);
268 virtual void OnTransportFailed(cricket::Transport* transport);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000269 virtual void OnTransportProxyCandidatesReady(
270 cricket::TransportProxy* proxy,
271 const cricket::Candidates& candidates);
272 virtual void OnCandidatesAllocationDone();
273
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000274 // Creates local session description with audio and video contents.
275 bool CreateDefaultLocalDescription();
276 // Enables media channels to allow sending of media.
277 void EnableChannels();
278 // Creates a JsepIceCandidate and adds it to the local session description
279 // and notify observers. Called when a new local candidate have been found.
280 void ProcessNewLocalCandidate(const std::string& content_name,
281 const cricket::Candidates& candidates);
282 // Returns the media index for a local ice candidate given the content name.
283 // Returns false if the local session description does not have a media
284 // content called |content_name|.
285 bool GetLocalCandidateMediaIndex(const std::string& content_name,
286 int* sdp_mline_index);
287 // Uses all remote candidates in |remote_desc| in this session.
288 bool UseCandidatesInSessionDescription(
289 const SessionDescriptionInterface* remote_desc);
290 // Uses |candidate| in this session.
291 bool UseCandidate(const IceCandidateInterface* candidate);
292 // Deletes the corresponding channel of contents that don't exist in |desc|.
293 // |desc| can be null. This means that all channels are deleted.
294 void RemoveUnusedChannelsAndTransports(
295 const cricket::SessionDescription* desc);
296
297 // Allocates media channels based on the |desc|. If |desc| doesn't have
298 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
299 // This method will also delete any existing media channels before creating.
300 bool CreateChannels(const cricket::SessionDescription* desc);
301
302 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000303 bool CreateVoiceChannel(const cricket::ContentInfo* content);
304 bool CreateVideoChannel(const cricket::ContentInfo* content);
305 bool CreateDataChannel(const cricket::ContentInfo* content);
306
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000307 // Copy the candidates from |saved_candidates_| to |dest_desc|.
308 // The |saved_candidates_| will be cleared after this function call.
309 void CopySavedCandidates(SessionDescriptionInterface* dest_desc);
310
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000311 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
312 // messages.
313 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
314 const cricket::ReceiveDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000315 const rtc::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000316
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000317 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000318 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
319
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000320 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000321 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000322 // Below methods are helper methods which verifies SDP.
323 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
324 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000325 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000326
327 // Check if a call to SetLocalDescription is acceptable with |action|.
328 bool ExpectSetLocalDescription(Action action);
329 // Check if a call to SetRemoteDescription is acceptable with |action|.
330 bool ExpectSetRemoteDescription(Action action);
331 // Verifies a=setup attribute as per RFC 5763.
332 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
333 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000334
jiayl@webrtc.orge10d28c2014-07-17 17:07:49 +0000335 // Returns true if we are ready to push down the remote candidate.
336 // |remote_desc| is the new remote description, or NULL if the current remote
337 // description should be used. Output |valid| is true if the candidate media
338 // index is valid.
339 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
340 const SessionDescriptionInterface* remote_desc,
341 bool* valid);
342
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000343 std::string GetSessionErrorMsg();
344
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000345 // Invoked when OnTransportCompleted is signaled to gather the usage
346 // of IPv4/IPv6 as best connection.
347 void ReportBestConnectionState(cricket::Transport* transport);
348
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000349 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
350 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
351 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000352 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000353 MediaStreamSignaling* mediastream_signaling_;
354 IceObserver* ice_observer_;
355 PeerConnectionInterface::IceConnectionState ice_connection_state_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000356 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
357 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000358 // Candidates that arrived before the remote description was set.
359 std::vector<IceCandidateInterface*> saved_candidates_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000360 // If the remote peer is using a older version of implementation.
361 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000362 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000363 // Specifies which kind of data channel is allowed. This is controlled
364 // by the chrome command-line flag and constraints:
365 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
366 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
367 // not set or false, SCTP is allowed (DCT_SCTP);
368 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
369 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
370 cricket::DataChannelType data_channel_type_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000371 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000372
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000373 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04 +0000374 webrtc_session_desc_factory_;
375
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000376 sigslot::signal0<> SignalVoiceChannelDestroyed;
377 sigslot::signal0<> SignalVideoChannelDestroyed;
378 sigslot::signal0<> SignalDataChannelDestroyed;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000379
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000380 // Member variables for caching global options.
381 cricket::AudioOptions audio_options_;
382 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000383 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000384
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000385 // Declares the bundle policy for the WebRTCSession.
386 PeerConnectionInterface::BundlePolicy bundle_policy_;
387
wu@webrtc.org364f2042013-11-20 21:49:41 +0000388 DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
389};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000390} // namespace webrtc
391
392#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_