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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.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000183 virtual void SetAudioPlayout(uint32 ssrc, bool enable,
184 cricket::AudioRenderer* renderer) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000185 virtual void SetAudioSend(uint32 ssrc, bool enable,
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000186 const cricket::AudioOptions& options,
187 cricket::AudioRenderer* renderer) OVERRIDE;
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000188 virtual void SetAudioPlayoutVolume(uint32 ssrc, double volume) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000189
190 // Implements VideoMediaProviderInterface.
191 virtual bool SetCaptureDevice(uint32 ssrc,
192 cricket::VideoCapturer* camera) OVERRIDE;
193 virtual void SetVideoPlayout(uint32 ssrc,
194 bool enable,
195 cricket::VideoRenderer* renderer) OVERRIDE;
196 virtual void SetVideoSend(uint32 ssrc, bool enable,
197 const cricket::VideoOptions* options) OVERRIDE;
198
199 // Implements DtmfProviderInterface.
200 virtual bool CanInsertDtmf(const std::string& track_id);
201 virtual bool InsertDtmf(const std::string& track_id,
202 int code, int duration);
203 virtual sigslot::signal0<>* GetOnDestroyedSignal();
204
wu@webrtc.org78187522013-10-07 23:32:02 +0000205 // Implements DataChannelProviderInterface.
206 virtual bool SendData(const cricket::SendDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000207 const rtc::Buffer& payload,
wu@webrtc.org78187522013-10-07 23:32:02 +0000208 cricket::SendDataResult* result) OVERRIDE;
209 virtual bool ConnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE;
210 virtual void DisconnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE;
bemasc@webrtc.org9b5467e2014-12-04 23:16:52 +0000211 virtual void AddSctpDataStream(int sid) OVERRIDE;
212 virtual void RemoveSctpDataStream(int sid) OVERRIDE;
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +0000213 virtual bool ReadyToSendData() const OVERRIDE;
wu@webrtc.org78187522013-10-07 23:32:02 +0000214
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000215 // Implements DataChannelFactory.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000216 rtc::scoped_refptr<DataChannel> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000217 const std::string& label,
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000218 const InternalDataChannelInit* config) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000219
220 cricket::DataChannelType data_channel_type() const;
221
wu@webrtc.org91053e72013-08-10 07:18:04 +0000222 bool IceRestartPending() const;
223
224 void ResetIceRestartLatch();
225
226 // Called when an SSLIdentity is generated or retrieved by
227 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000228 void OnIdentityReady(rtc::SSLIdentity* identity);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000229
230 // For unit test.
231 bool waiting_for_identity() const;
232
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000233 void set_metrics_observer(
234 webrtc::MetricsObserverInterface* metrics_observer) {
235 metrics_observer_ = metrics_observer;
236 }
237
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000238 private:
239 // Indicates the type of SessionDescription in a call to SetLocalDescription
240 // and SetRemoteDescription.
241 enum Action {
242 kOffer,
243 kPrAnswer,
244 kAnswer,
245 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000246
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000247 // Invokes ConnectChannels() on transport proxies, which initiates ice
248 // candidates allocation.
249 bool StartCandidatesAllocation();
250 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000251 std::string* err_desc);
252 static Action GetAction(const std::string& type);
253
254 // Transport related callbacks, override from cricket::BaseSession.
255 virtual void OnTransportRequestSignaling(cricket::Transport* transport);
256 virtual void OnTransportConnecting(cricket::Transport* transport);
257 virtual void OnTransportWritable(cricket::Transport* transport);
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000258 virtual void OnTransportCompleted(cricket::Transport* transport);
259 virtual void OnTransportFailed(cricket::Transport* transport);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000260 virtual void OnTransportProxyCandidatesReady(
261 cricket::TransportProxy* proxy,
262 const cricket::Candidates& candidates);
263 virtual void OnCandidatesAllocationDone();
264
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000265 // Creates local session description with audio and video contents.
266 bool CreateDefaultLocalDescription();
267 // Enables media channels to allow sending of media.
268 void EnableChannels();
269 // Creates a JsepIceCandidate and adds it to the local session description
270 // and notify observers. Called when a new local candidate have been found.
271 void ProcessNewLocalCandidate(const std::string& content_name,
272 const cricket::Candidates& candidates);
273 // Returns the media index for a local ice candidate given the content name.
274 // Returns false if the local session description does not have a media
275 // content called |content_name|.
276 bool GetLocalCandidateMediaIndex(const std::string& content_name,
277 int* sdp_mline_index);
278 // Uses all remote candidates in |remote_desc| in this session.
279 bool UseCandidatesInSessionDescription(
280 const SessionDescriptionInterface* remote_desc);
281 // Uses |candidate| in this session.
282 bool UseCandidate(const IceCandidateInterface* candidate);
283 // Deletes the corresponding channel of contents that don't exist in |desc|.
284 // |desc| can be null. This means that all channels are deleted.
285 void RemoveUnusedChannelsAndTransports(
286 const cricket::SessionDescription* desc);
287
288 // Allocates media channels based on the |desc|. If |desc| doesn't have
289 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
290 // This method will also delete any existing media channels before creating.
291 bool CreateChannels(const cricket::SessionDescription* desc);
292
293 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000294 bool CreateVoiceChannel(const cricket::ContentInfo* content);
295 bool CreateVideoChannel(const cricket::ContentInfo* content);
296 bool CreateDataChannel(const cricket::ContentInfo* content);
297
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000298 // Copy the candidates from |saved_candidates_| to |dest_desc|.
299 // The |saved_candidates_| will be cleared after this function call.
300 void CopySavedCandidates(SessionDescriptionInterface* dest_desc);
301
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000302 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
303 // messages.
304 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
305 const cricket::ReceiveDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000306 const rtc::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000307
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000308 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000309 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
310
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000311 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000312 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000313 // Below methods are helper methods which verifies SDP.
314 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
315 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000316 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000317
318 // Check if a call to SetLocalDescription is acceptable with |action|.
319 bool ExpectSetLocalDescription(Action action);
320 // Check if a call to SetRemoteDescription is acceptable with |action|.
321 bool ExpectSetRemoteDescription(Action action);
322 // Verifies a=setup attribute as per RFC 5763.
323 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
324 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000325
jiayl@webrtc.orge10d28c2014-07-17 17:07:49 +0000326 // Returns true if we are ready to push down the remote candidate.
327 // |remote_desc| is the new remote description, or NULL if the current remote
328 // description should be used. Output |valid| is true if the candidate media
329 // index is valid.
330 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
331 const SessionDescriptionInterface* remote_desc,
332 bool* valid);
333
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000334 std::string GetSessionErrorMsg();
335
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000336 // Invoked when OnTransportCompleted is signaled to gather the usage
337 // of IPv4/IPv6 as best connection.
338 void ReportBestConnectionState(cricket::Transport* transport);
339
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000340 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
341 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
342 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000343 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000344 MediaStreamSignaling* mediastream_signaling_;
345 IceObserver* ice_observer_;
346 PeerConnectionInterface::IceConnectionState ice_connection_state_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000347 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
348 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000349 // Candidates that arrived before the remote description was set.
350 std::vector<IceCandidateInterface*> saved_candidates_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000351 // If the remote peer is using a older version of implementation.
352 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000353 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000354 // Specifies which kind of data channel is allowed. This is controlled
355 // by the chrome command-line flag and constraints:
356 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
357 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
358 // not set or false, SCTP is allowed (DCT_SCTP);
359 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
360 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
361 cricket::DataChannelType data_channel_type_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000362 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000363
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000364 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04 +0000365 webrtc_session_desc_factory_;
366
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000367 sigslot::signal0<> SignalVoiceChannelDestroyed;
368 sigslot::signal0<> SignalVideoChannelDestroyed;
369 sigslot::signal0<> SignalDataChannelDestroyed;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000370
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000371 // Member variables for caching global options.
372 cricket::AudioOptions audio_options_;
373 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000374 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000375
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000376 // Declares the bundle policy for the WebRTCSession.
377 PeerConnectionInterface::BundlePolicy bundle_policy_;
378
wu@webrtc.org364f2042013-11-20 21:49:41 +0000379 DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
380};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000381} // namespace webrtc
382
383#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_