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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"
Henrik Boström5e56c592015-08-11 10:33:13 +020042#include "webrtc/base/sslidentity.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000043#include "webrtc/base/thread.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000044
45namespace cricket {
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000046
wu@webrtc.org364f2042013-11-20 21:49:41 +000047class BaseChannel;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000048class ChannelManager;
49class DataChannel;
50class StatsReport;
51class Transport;
52class VideoCapturer;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000053class VideoChannel;
54class VoiceChannel;
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000055
henrike@webrtc.org28e20752013-07-10 00:45:36 +000056} // namespace cricket
57
58namespace webrtc {
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000059
henrike@webrtc.org28e20752013-07-10 00:45:36 +000060class IceRestartAnswerLatch;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000061class JsepIceCandidate;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000062class MediaStreamSignaling;
wu@webrtc.org91053e72013-08-10 07:18:04 +000063class WebRtcSessionDescriptionFactory;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000064
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000065extern const char kBundleWithoutRtcpMux[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000066extern const char kCreateChannelFailed[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000067extern const char kInvalidCandidates[];
68extern const char kInvalidSdp[];
69extern const char kMlineMismatch[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000070extern const char kPushDownTDFailed[];
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +000071extern const char kSdpWithoutDtlsFingerprint[];
72extern const char kSdpWithoutSdesCrypto[];
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000073extern const char kSdpWithoutIceUfragPwd[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000074extern const char kSdpWithoutSdesAndDtlsDisabled[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000075extern const char kSessionError[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000076extern const char kSessionErrorDesc[];
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +000077extern const char kDtlsSetupFailureRtp[];
78extern const char kDtlsSetupFailureRtcp[];
buildbot@webrtc.org53df88c2014-08-07 22:46:01 +000079// Maximum number of received video streams that will be processed by webrtc
80// even if they are not signalled beforehand.
81extern const int kMaxUnsignalledRecvStreams;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000082
83// ICE state callback interface.
84class IceObserver {
85 public:
wu@webrtc.org364f2042013-11-20 21:49:41 +000086 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +000087 // Called any time the IceConnectionState changes
Peter Thatcher54360512015-07-08 11:08:35 -070088 // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to
89 // conform to the w3c standard.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000090 virtual void OnIceConnectionChange(
91 PeerConnectionInterface::IceConnectionState new_state) {}
92 // Called any time the IceGatheringState changes
93 virtual void OnIceGatheringChange(
94 PeerConnectionInterface::IceGatheringState new_state) {}
95 // New Ice candidate have been found.
96 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
97 // All Ice candidates have been found.
98 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
99 // (via PeerConnectionObserver)
100 virtual void OnIceComplete() {}
101
Peter Thatcher54360512015-07-08 11:08:35 -0700102 // Called whenever the state changes between receiving and not receiving.
103 virtual void OnIceConnectionReceivingChange(bool receiving) {}
104
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000105 protected:
106 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +0000107
108 private:
109 DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000110};
111
112class WebRtcSession : public cricket::BaseSession,
113 public AudioProviderInterface,
114 public DataChannelFactory,
115 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000116 public DtmfProviderInterface,
117 public DataChannelProviderInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000118 public:
119 WebRtcSession(cricket::ChannelManager* channel_manager,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000120 rtc::Thread* signaling_thread,
121 rtc::Thread* worker_thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000122 cricket::PortAllocator* port_allocator,
123 MediaStreamSignaling* mediastream_signaling);
124 virtual ~WebRtcSession();
125
Henrik Lundin64dad832015-05-11 12:44:23 +0200126 bool Initialize(
127 const PeerConnectionFactoryInterface::Options& options,
128 const MediaConstraintsInterface* constraints,
Henrik Boström5e56c592015-08-11 10:33:13 +0200129 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
Henrik Lundin64dad832015-05-11 12:44:23 +0200130 const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000131 // Deletes the voice, video and data channel and changes the session state
132 // to STATE_RECEIVEDTERMINATE.
133 void Terminate();
134
135 void RegisterIceObserver(IceObserver* observer) {
136 ice_observer_ = observer;
137 }
138
139 virtual cricket::VoiceChannel* voice_channel() {
140 return voice_channel_.get();
141 }
142 virtual cricket::VideoChannel* video_channel() {
143 return video_channel_.get();
144 }
145 virtual cricket::DataChannel* data_channel() {
146 return data_channel_.get();
147 }
148
decurtis@webrtc.org487a4442015-01-15 22:55:07 +0000149 virtual const MediaStreamSignaling* mediastream_signaling() const {
150 return mediastream_signaling_;
151 }
152
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000153 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
154 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000155
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000156 // Get current ssl role from transport.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000157 bool GetSslRole(rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000158
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000159 // Generic error message callback from WebRtcSession.
160 // TODO - It may be necessary to supply error code as well.
161 sigslot::signal0<> SignalError;
162
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000163 void CreateOffer(
164 CreateSessionDescriptionObserver* observer,
165 const PeerConnectionInterface::RTCOfferAnswerOptions& options);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000166 void CreateAnswer(CreateSessionDescriptionObserver* observer,
167 const MediaConstraintsInterface* constraints);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000168 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000169 bool SetLocalDescription(SessionDescriptionInterface* desc,
170 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000171 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000172 bool SetRemoteDescription(SessionDescriptionInterface* desc,
173 std::string* err_desc);
174 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000175
mallinath@webrtc.org3d81b1b2014-09-09 14:38:10 +0000176 bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000177
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000178 const SessionDescriptionInterface* local_description() const {
179 return local_desc_.get();
180 }
181 const SessionDescriptionInterface* remote_description() const {
182 return remote_desc_.get();
183 }
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000184 // TODO(pthatcher): Cleanup the distinction between
185 // SessionDescription and SessionDescriptionInterface and remove
186 // these if possible.
187 const cricket::SessionDescription* base_local_description() const {
188 return BaseSession::local_description();
189 }
190 const cricket::SessionDescription* base_remote_description() const {
191 return BaseSession::remote_description();
192 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000193
194 // Get the id used as a media stream track's "id" field from ssrc.
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000195 virtual bool GetLocalTrackIdBySsrc(uint32 ssrc, std::string* track_id);
196 virtual bool GetRemoteTrackIdBySsrc(uint32 ssrc, std::string* track_id);
197
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000198
199 // AudioMediaProviderInterface implementation.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000200 void SetAudioPlayout(uint32 ssrc,
201 bool enable,
202 cricket::AudioRenderer* renderer) override;
203 void SetAudioSend(uint32 ssrc,
204 bool enable,
205 const cricket::AudioOptions& options,
206 cricket::AudioRenderer* renderer) override;
207 void SetAudioPlayoutVolume(uint32 ssrc, double volume) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000208
209 // Implements VideoMediaProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000210 bool SetCaptureDevice(uint32 ssrc, cricket::VideoCapturer* camera) override;
211 void SetVideoPlayout(uint32 ssrc,
212 bool enable,
213 cricket::VideoRenderer* renderer) override;
214 void SetVideoSend(uint32 ssrc,
215 bool enable,
216 const cricket::VideoOptions* options) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000217
218 // Implements DtmfProviderInterface.
219 virtual bool CanInsertDtmf(const std::string& track_id);
220 virtual bool InsertDtmf(const std::string& track_id,
221 int code, int duration);
222 virtual sigslot::signal0<>* GetOnDestroyedSignal();
223
wu@webrtc.org78187522013-10-07 23:32:02 +0000224 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000225 bool SendData(const cricket::SendDataParams& params,
226 const rtc::Buffer& payload,
227 cricket::SendDataResult* result) override;
228 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
229 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
230 void AddSctpDataStream(int sid) override;
231 void RemoveSctpDataStream(int sid) override;
232 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02 +0000233
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000234 // Returns stats for all channels of all transports.
235 // This avoids exposing the internal structures used to track them.
236 virtual bool GetTransportStats(cricket::SessionStats* stats);
237
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000238 // Implements DataChannelFactory.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000239 rtc::scoped_refptr<DataChannel> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000240 const std::string& label,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000241 const InternalDataChannelInit* config) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000242
243 cricket::DataChannelType data_channel_type() const;
244
wu@webrtc.org91053e72013-08-10 07:18:04 +0000245 bool IceRestartPending() const;
246
247 void ResetIceRestartLatch();
248
Henrik Boströmd8281982015-08-27 10:12:24 +0200249 // Called when an RTCCertificate is generated or retrieved by
wu@webrtc.org91053e72013-08-10 07:18:04 +0000250 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
Henrik Boströmd8281982015-08-27 10:12:24 +0200251 void OnCertificateReady(
252 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +0000253 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000254
255 // For unit test.
Henrik Boströmd8281982015-08-27 10:12:24 +0200256 bool waiting_for_certificate_for_testing() const;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000257
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000258 void set_metrics_observer(
259 webrtc::MetricsObserverInterface* metrics_observer) {
260 metrics_observer_ = metrics_observer;
261 }
262
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000263 private:
264 // Indicates the type of SessionDescription in a call to SetLocalDescription
265 // and SetRemoteDescription.
266 enum Action {
267 kOffer,
268 kPrAnswer,
269 kAnswer,
270 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000271
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000272 // Invokes ConnectChannels() on transport proxies, which initiates ice
273 // candidates allocation.
274 bool StartCandidatesAllocation();
275 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000276 std::string* err_desc);
277 static Action GetAction(const std::string& type);
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000278 // Push the media parts of the local or remote session description
279 // down to all of the channels.
280 bool PushdownMediaDescription(cricket::ContentAction action,
281 cricket::ContentSource source,
282 std::string* error_desc);
283
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000284
285 // Transport related callbacks, override from cricket::BaseSession.
286 virtual void OnTransportRequestSignaling(cricket::Transport* transport);
287 virtual void OnTransportConnecting(cricket::Transport* transport);
288 virtual void OnTransportWritable(cricket::Transport* transport);
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000289 virtual void OnTransportCompleted(cricket::Transport* transport);
290 virtual void OnTransportFailed(cricket::Transport* transport);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000291 virtual void OnTransportProxyCandidatesReady(
292 cricket::TransportProxy* proxy,
293 const cricket::Candidates& candidates);
294 virtual void OnCandidatesAllocationDone();
Peter Thatcher54360512015-07-08 11:08:35 -0700295 void OnTransportReceiving(cricket::Transport* transport) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000296
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000297 // Enables media channels to allow sending of media.
298 void EnableChannels();
299 // Creates a JsepIceCandidate and adds it to the local session description
300 // and notify observers. Called when a new local candidate have been found.
301 void ProcessNewLocalCandidate(const std::string& content_name,
302 const cricket::Candidates& candidates);
303 // Returns the media index for a local ice candidate given the content name.
304 // Returns false if the local session description does not have a media
305 // content called |content_name|.
306 bool GetLocalCandidateMediaIndex(const std::string& content_name,
307 int* sdp_mline_index);
308 // Uses all remote candidates in |remote_desc| in this session.
309 bool UseCandidatesInSessionDescription(
310 const SessionDescriptionInterface* remote_desc);
311 // Uses |candidate| in this session.
312 bool UseCandidate(const IceCandidateInterface* candidate);
313 // Deletes the corresponding channel of contents that don't exist in |desc|.
314 // |desc| can be null. This means that all channels are deleted.
315 void RemoveUnusedChannelsAndTransports(
316 const cricket::SessionDescription* desc);
317
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 19:17:59 +0000324 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 00:45:36 +0000328 // 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 10:00:58 +0000332 // 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 17:36:52 +0000336 const rtc::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000337
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000338 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000339 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
Peter Thatcher54360512015-07-08 11:08:35 -0700340 void SetIceConnectionReceiving(bool receiving);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000341
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000342 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000343 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000344 // Below methods are helper methods which verifies SDP.
345 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
346 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000347 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000348
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 19:17:59 +0000356
jiayl@webrtc.orge10d28c2014-07-17 17:07:49 +0000357 // 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
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000365 std::string GetSessionErrorMsg();
366
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000367 // Invoked when OnTransportCompleted is signaled to gather the usage
368 // of IPv4/IPv6 as best connection.
jbauchac8869e2015-07-03 01:36:14 -0700369 void ReportBestConnectionState(const cricket::TransportStats& stats);
370
371 void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000372
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000373 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
374 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
375 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000376 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000377 MediaStreamSignaling* mediastream_signaling_;
378 IceObserver* ice_observer_;
379 PeerConnectionInterface::IceConnectionState ice_connection_state_;
Peter Thatcher54360512015-07-08 11:08:35 -0700380 bool ice_connection_receiving_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000381 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
382 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000383 // Candidates that arrived before the remote description was set.
384 std::vector<IceCandidateInterface*> saved_candidates_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000385 // If the remote peer is using a older version of implementation.
386 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000387 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000388 // Specifies which kind of data channel is allowed. This is controlled
389 // by the chrome command-line flag and constraints:
390 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
391 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
392 // not set or false, SCTP is allowed (DCT_SCTP);
393 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
394 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
395 cricket::DataChannelType data_channel_type_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000396 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000397
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000398 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04 +0000399 webrtc_session_desc_factory_;
400
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000401 sigslot::signal0<> SignalVoiceChannelDestroyed;
402 sigslot::signal0<> SignalVideoChannelDestroyed;
403 sigslot::signal0<> SignalDataChannelDestroyed;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000404
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000405 // Member variables for caching global options.
406 cricket::AudioOptions audio_options_;
407 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000408 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000409
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000410 // Declares the bundle policy for the WebRTCSession.
411 PeerConnectionInterface::BundlePolicy bundle_policy_;
412
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700413 // Declares the RTCP mux policy for the WebRTCSession.
414 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
415
wu@webrtc.org364f2042013-11-20 21:49:41 +0000416 DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
417};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000418} // namespace webrtc
419
420#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_