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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[];
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
Peter Thatcher54360512015-07-08 11:08:35 -070087 // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to
88 // conform to the w3c standard.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000089 virtual void OnIceConnectionChange(
90 PeerConnectionInterface::IceConnectionState new_state) {}
91 // Called any time the IceGatheringState changes
92 virtual void OnIceGatheringChange(
93 PeerConnectionInterface::IceGatheringState new_state) {}
94 // New Ice candidate have been found.
95 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
96 // All Ice candidates have been found.
97 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
98 // (via PeerConnectionObserver)
99 virtual void OnIceComplete() {}
100
Peter Thatcher54360512015-07-08 11:08:35 -0700101 // Called whenever the state changes between receiving and not receiving.
102 virtual void OnIceConnectionReceivingChange(bool receiving) {}
103
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000104 protected:
105 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +0000106
107 private:
108 DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000109};
110
111class WebRtcSession : public cricket::BaseSession,
112 public AudioProviderInterface,
113 public DataChannelFactory,
114 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000115 public DtmfProviderInterface,
116 public DataChannelProviderInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000117 public:
118 WebRtcSession(cricket::ChannelManager* channel_manager,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000119 rtc::Thread* signaling_thread,
120 rtc::Thread* worker_thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000121 cricket::PortAllocator* port_allocator,
122 MediaStreamSignaling* mediastream_signaling);
123 virtual ~WebRtcSession();
124
Henrik Lundin64dad832015-05-11 12:44:23 +0200125 bool Initialize(
126 const PeerConnectionFactoryInterface::Options& options,
127 const MediaConstraintsInterface* constraints,
128 DTLSIdentityServiceInterface* dtls_identity_service,
129 const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000130 // Deletes the voice, video and data channel and changes the session state
131 // to STATE_RECEIVEDTERMINATE.
132 void Terminate();
133
134 void RegisterIceObserver(IceObserver* observer) {
135 ice_observer_ = observer;
136 }
137
138 virtual cricket::VoiceChannel* voice_channel() {
139 return voice_channel_.get();
140 }
141 virtual cricket::VideoChannel* video_channel() {
142 return video_channel_.get();
143 }
144 virtual cricket::DataChannel* data_channel() {
145 return data_channel_.get();
146 }
147
decurtis@webrtc.org487a4442015-01-15 22:55:07 +0000148 virtual const MediaStreamSignaling* mediastream_signaling() const {
149 return mediastream_signaling_;
150 }
151
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000152 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
153 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000154
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000155 // Get current ssl role from transport.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000156 bool GetSslRole(rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000157
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000158 // Generic error message callback from WebRtcSession.
159 // TODO - It may be necessary to supply error code as well.
160 sigslot::signal0<> SignalError;
161
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000162 void CreateOffer(
163 CreateSessionDescriptionObserver* observer,
164 const PeerConnectionInterface::RTCOfferAnswerOptions& options);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000165 void CreateAnswer(CreateSessionDescriptionObserver* observer,
166 const MediaConstraintsInterface* constraints);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000167 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000168 bool SetLocalDescription(SessionDescriptionInterface* desc,
169 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000170 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000171 bool SetRemoteDescription(SessionDescriptionInterface* desc,
172 std::string* err_desc);
173 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000174
mallinath@webrtc.org3d81b1b2014-09-09 14:38:10 +0000175 bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000176
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000177 const SessionDescriptionInterface* local_description() const {
178 return local_desc_.get();
179 }
180 const SessionDescriptionInterface* remote_description() const {
181 return remote_desc_.get();
182 }
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000183 // TODO(pthatcher): Cleanup the distinction between
184 // SessionDescription and SessionDescriptionInterface and remove
185 // these if possible.
186 const cricket::SessionDescription* base_local_description() const {
187 return BaseSession::local_description();
188 }
189 const cricket::SessionDescription* base_remote_description() const {
190 return BaseSession::remote_description();
191 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000192
193 // Get the id used as a media stream track's "id" field from ssrc.
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000194 virtual bool GetLocalTrackIdBySsrc(uint32 ssrc, std::string* track_id);
195 virtual bool GetRemoteTrackIdBySsrc(uint32 ssrc, std::string* track_id);
196
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000197
198 // AudioMediaProviderInterface implementation.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000199 void SetAudioPlayout(uint32 ssrc,
200 bool enable,
201 cricket::AudioRenderer* renderer) override;
202 void SetAudioSend(uint32 ssrc,
203 bool enable,
204 const cricket::AudioOptions& options,
205 cricket::AudioRenderer* renderer) override;
206 void SetAudioPlayoutVolume(uint32 ssrc, double volume) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000207
208 // Implements VideoMediaProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000209 bool SetCaptureDevice(uint32 ssrc, cricket::VideoCapturer* camera) override;
210 void SetVideoPlayout(uint32 ssrc,
211 bool enable,
212 cricket::VideoRenderer* renderer) override;
213 void SetVideoSend(uint32 ssrc,
214 bool enable,
215 const cricket::VideoOptions* options) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000216
217 // Implements DtmfProviderInterface.
218 virtual bool CanInsertDtmf(const std::string& track_id);
219 virtual bool InsertDtmf(const std::string& track_id,
220 int code, int duration);
221 virtual sigslot::signal0<>* GetOnDestroyedSignal();
222
wu@webrtc.org78187522013-10-07 23:32:02 +0000223 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000224 bool SendData(const cricket::SendDataParams& params,
225 const rtc::Buffer& payload,
226 cricket::SendDataResult* result) override;
227 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
228 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
229 void AddSctpDataStream(int sid) override;
230 void RemoveSctpDataStream(int sid) override;
231 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02 +0000232
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000233 // Returns stats for all channels of all transports.
234 // This avoids exposing the internal structures used to track them.
235 virtual bool GetTransportStats(cricket::SessionStats* stats);
236
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000237 // Implements DataChannelFactory.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000238 rtc::scoped_refptr<DataChannel> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000239 const std::string& label,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000240 const InternalDataChannelInit* config) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000241
242 cricket::DataChannelType data_channel_type() const;
243
wu@webrtc.org91053e72013-08-10 07:18:04 +0000244 bool IceRestartPending() const;
245
246 void ResetIceRestartLatch();
247
248 // Called when an SSLIdentity is generated or retrieved by
249 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000250 void OnIdentityReady(rtc::SSLIdentity* identity);
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +0000251 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000252
253 // For unit test.
254 bool waiting_for_identity() const;
255
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000256 void set_metrics_observer(
257 webrtc::MetricsObserverInterface* metrics_observer) {
258 metrics_observer_ = metrics_observer;
259 }
260
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000261 protected:
262 // Don't fire a new description. The only thing it's used for is to
263 // push new media descriptions to the BaseChannels. But in
264 // WebRtcSession, we just push to the BaseChannels directly, so we
265 // don't need this (and it would cause the descriptions to be pushed
266 // down twice).
267 // TODO(pthatcher): Remove this method and signal completely from
268 // BaseSession once all the subclasses of BaseSession push to
269 // BaseChannels directly rather than relying on the signal, or once
270 // BaseChannel no longer listens to the event and requires
271 // descriptions to be pushed down.
272 virtual void SignalNewDescription() override {}
273
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000274 private:
275 // Indicates the type of SessionDescription in a call to SetLocalDescription
276 // and SetRemoteDescription.
277 enum Action {
278 kOffer,
279 kPrAnswer,
280 kAnswer,
281 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000282
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000283 // Invokes ConnectChannels() on transport proxies, which initiates ice
284 // candidates allocation.
285 bool StartCandidatesAllocation();
286 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000287 std::string* err_desc);
288 static Action GetAction(const std::string& type);
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000289 // Push the media parts of the local or remote session description
290 // down to all of the channels.
291 bool PushdownMediaDescription(cricket::ContentAction action,
292 cricket::ContentSource source,
293 std::string* error_desc);
294
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000295
296 // Transport related callbacks, override from cricket::BaseSession.
297 virtual void OnTransportRequestSignaling(cricket::Transport* transport);
298 virtual void OnTransportConnecting(cricket::Transport* transport);
299 virtual void OnTransportWritable(cricket::Transport* transport);
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000300 virtual void OnTransportCompleted(cricket::Transport* transport);
301 virtual void OnTransportFailed(cricket::Transport* transport);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000302 virtual void OnTransportProxyCandidatesReady(
303 cricket::TransportProxy* proxy,
304 const cricket::Candidates& candidates);
305 virtual void OnCandidatesAllocationDone();
Peter Thatcher54360512015-07-08 11:08:35 -0700306 void OnTransportReceiving(cricket::Transport* transport) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000307
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000308 // Enables media channels to allow sending of media.
309 void EnableChannels();
310 // Creates a JsepIceCandidate and adds it to the local session description
311 // and notify observers. Called when a new local candidate have been found.
312 void ProcessNewLocalCandidate(const std::string& content_name,
313 const cricket::Candidates& candidates);
314 // Returns the media index for a local ice candidate given the content name.
315 // Returns false if the local session description does not have a media
316 // content called |content_name|.
317 bool GetLocalCandidateMediaIndex(const std::string& content_name,
318 int* sdp_mline_index);
319 // Uses all remote candidates in |remote_desc| in this session.
320 bool UseCandidatesInSessionDescription(
321 const SessionDescriptionInterface* remote_desc);
322 // Uses |candidate| in this session.
323 bool UseCandidate(const IceCandidateInterface* candidate);
324 // Deletes the corresponding channel of contents that don't exist in |desc|.
325 // |desc| can be null. This means that all channels are deleted.
326 void RemoveUnusedChannelsAndTransports(
327 const cricket::SessionDescription* desc);
328
329 // Allocates media channels based on the |desc|. If |desc| doesn't have
330 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
331 // This method will also delete any existing media channels before creating.
332 bool CreateChannels(const cricket::SessionDescription* desc);
333
334 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000335 bool CreateVoiceChannel(const cricket::ContentInfo* content);
336 bool CreateVideoChannel(const cricket::ContentInfo* content);
337 bool CreateDataChannel(const cricket::ContentInfo* content);
338
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000339 // Copy the candidates from |saved_candidates_| to |dest_desc|.
340 // The |saved_candidates_| will be cleared after this function call.
341 void CopySavedCandidates(SessionDescriptionInterface* dest_desc);
342
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000343 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
344 // messages.
345 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
346 const cricket::ReceiveDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000347 const rtc::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000348
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000349 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000350 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
Peter Thatcher54360512015-07-08 11:08:35 -0700351 void SetIceConnectionReceiving(bool receiving);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000352
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000353 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000354 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000355 // Below methods are helper methods which verifies SDP.
356 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
357 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000358 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000359
360 // Check if a call to SetLocalDescription is acceptable with |action|.
361 bool ExpectSetLocalDescription(Action action);
362 // Check if a call to SetRemoteDescription is acceptable with |action|.
363 bool ExpectSetRemoteDescription(Action action);
364 // Verifies a=setup attribute as per RFC 5763.
365 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
366 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000367
jiayl@webrtc.orge10d28c2014-07-17 17:07:49 +0000368 // Returns true if we are ready to push down the remote candidate.
369 // |remote_desc| is the new remote description, or NULL if the current remote
370 // description should be used. Output |valid| is true if the candidate media
371 // index is valid.
372 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
373 const SessionDescriptionInterface* remote_desc,
374 bool* valid);
375
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000376 std::string GetSessionErrorMsg();
377
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000378 // Invoked when OnTransportCompleted is signaled to gather the usage
379 // of IPv4/IPv6 as best connection.
jbauchac8869e2015-07-03 01:36:14 -0700380 void ReportBestConnectionState(const cricket::TransportStats& stats);
381
382 void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000383
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000384 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
385 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
386 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000387 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000388 MediaStreamSignaling* mediastream_signaling_;
389 IceObserver* ice_observer_;
390 PeerConnectionInterface::IceConnectionState ice_connection_state_;
Peter Thatcher54360512015-07-08 11:08:35 -0700391 bool ice_connection_receiving_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000392 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
393 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000394 // Candidates that arrived before the remote description was set.
395 std::vector<IceCandidateInterface*> saved_candidates_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000396 // If the remote peer is using a older version of implementation.
397 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000398 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000399 // Specifies which kind of data channel is allowed. This is controlled
400 // by the chrome command-line flag and constraints:
401 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
402 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
403 // not set or false, SCTP is allowed (DCT_SCTP);
404 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
405 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
406 cricket::DataChannelType data_channel_type_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000407 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000408
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000409 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04 +0000410 webrtc_session_desc_factory_;
411
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000412 sigslot::signal0<> SignalVoiceChannelDestroyed;
413 sigslot::signal0<> SignalVideoChannelDestroyed;
414 sigslot::signal0<> SignalDataChannelDestroyed;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000415
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000416 // Member variables for caching global options.
417 cricket::AudioOptions audio_options_;
418 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000419 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000420
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000421 // Declares the bundle policy for the WebRTCSession.
422 PeerConnectionInterface::BundlePolicy bundle_policy_;
423
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700424 // Declares the RTCP mux policy for the WebRTCSession.
425 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
426
wu@webrtc.org364f2042013-11-20 21:49:41 +0000427 DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
428};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000429} // namespace webrtc
430
431#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_