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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>
deadbeefcbecd352015-09-23 11:50:27 -070032#include <vector>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000033
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000034#include "talk/app/webrtc/datachannel.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000035#include "talk/app/webrtc/dtmfsender.h"
Fredrik Solenberg709ed672015-09-15 12:26:33 +020036#include "talk/app/webrtc/mediacontroller.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000037#include "talk/app/webrtc/mediastreamprovider.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000038#include "talk/app/webrtc/peerconnectioninterface.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000039#include "talk/app/webrtc/statstypes.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000040#include "talk/media/base/mediachannel.h"
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000041#include "webrtc/p2p/base/session.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000042#include "talk/session/media/mediasession.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000043#include "webrtc/base/sigslot.h"
Henrik Boström5e56c592015-08-11 10:33:13 +020044#include "webrtc/base/sslidentity.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000045#include "webrtc/base/thread.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000046
47namespace cricket {
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000048
wu@webrtc.org364f2042013-11-20 21:49:41 +000049class BaseChannel;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000050class ChannelManager;
51class DataChannel;
52class StatsReport;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000053class VideoCapturer;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000054class VideoChannel;
55class VoiceChannel;
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000056
henrike@webrtc.org28e20752013-07-10 00:45:36 +000057} // namespace cricket
58
59namespace webrtc {
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000060
henrike@webrtc.org28e20752013-07-10 00:45:36 +000061class IceRestartAnswerLatch;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000062class JsepIceCandidate;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000063class MediaStreamSignaling;
wu@webrtc.org91053e72013-08-10 07:18:04 +000064class WebRtcSessionDescriptionFactory;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000065
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000066extern const char kBundleWithoutRtcpMux[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000067extern const char kCreateChannelFailed[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000068extern const char kInvalidCandidates[];
69extern const char kInvalidSdp[];
70extern const char kMlineMismatch[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000071extern const char kPushDownTDFailed[];
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +000072extern const char kSdpWithoutDtlsFingerprint[];
73extern const char kSdpWithoutSdesCrypto[];
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000074extern const char kSdpWithoutIceUfragPwd[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000075extern const char kSdpWithoutSdesAndDtlsDisabled[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000076extern const char kSessionError[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000077extern const char kSessionErrorDesc[];
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +000078extern const char kDtlsSetupFailureRtp[];
79extern const char kDtlsSetupFailureRtcp[];
deadbeefcbecd352015-09-23 11:50:27 -070080extern const char kEnableBundleFailed[];
81
buildbot@webrtc.org53df88c2014-08-07 22:46:01 +000082// Maximum number of received video streams that will be processed by webrtc
83// even if they are not signalled beforehand.
84extern const int kMaxUnsignalledRecvStreams;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000085
86// ICE state callback interface.
87class IceObserver {
88 public:
wu@webrtc.org364f2042013-11-20 21:49:41 +000089 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +000090 // Called any time the IceConnectionState changes
Peter Thatcher54360512015-07-08 11:08:35 -070091 // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to
92 // conform to the w3c standard.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000093 virtual void OnIceConnectionChange(
94 PeerConnectionInterface::IceConnectionState new_state) {}
95 // Called any time the IceGatheringState changes
96 virtual void OnIceGatheringChange(
97 PeerConnectionInterface::IceGatheringState new_state) {}
98 // New Ice candidate have been found.
99 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
100 // All Ice candidates have been found.
101 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
102 // (via PeerConnectionObserver)
103 virtual void OnIceComplete() {}
104
Peter Thatcher54360512015-07-08 11:08:35 -0700105 // Called whenever the state changes between receiving and not receiving.
106 virtual void OnIceConnectionReceivingChange(bool receiving) {}
107
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000108 protected:
109 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +0000110
111 private:
henrikg3c089d72015-09-16 05:37:44 -0700112 RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000113};
114
115class WebRtcSession : public cricket::BaseSession,
116 public AudioProviderInterface,
117 public DataChannelFactory,
118 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000119 public DtmfProviderInterface,
120 public DataChannelProviderInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000121 public:
122 WebRtcSession(cricket::ChannelManager* channel_manager,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000123 rtc::Thread* signaling_thread,
124 rtc::Thread* worker_thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000125 cricket::PortAllocator* port_allocator,
126 MediaStreamSignaling* mediastream_signaling);
127 virtual ~WebRtcSession();
128
Henrik Lundin64dad832015-05-11 12:44:23 +0200129 bool Initialize(
130 const PeerConnectionFactoryInterface::Options& options,
131 const MediaConstraintsInterface* constraints,
Henrik Boström5e56c592015-08-11 10:33:13 +0200132 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
Henrik Lundin64dad832015-05-11 12:44:23 +0200133 const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000134 // Deletes the voice, video and data channel and changes the session state
135 // to STATE_RECEIVEDTERMINATE.
136 void Terminate();
137
138 void RegisterIceObserver(IceObserver* observer) {
139 ice_observer_ = observer;
140 }
141
142 virtual cricket::VoiceChannel* voice_channel() {
143 return voice_channel_.get();
144 }
145 virtual cricket::VideoChannel* video_channel() {
146 return video_channel_.get();
147 }
148 virtual cricket::DataChannel* data_channel() {
149 return data_channel_.get();
150 }
151
decurtis@webrtc.org487a4442015-01-15 22:55:07 +0000152 virtual const MediaStreamSignaling* mediastream_signaling() const {
153 return mediastream_signaling_;
154 }
155
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000156 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
157 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000158
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000159 // Get current ssl role from transport.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000160 bool GetSslRole(rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000161
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000162 // Generic error message callback from WebRtcSession.
163 // TODO - It may be necessary to supply error code as well.
164 sigslot::signal0<> SignalError;
165
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000166 void CreateOffer(
167 CreateSessionDescriptionObserver* observer,
168 const PeerConnectionInterface::RTCOfferAnswerOptions& options);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000169 void CreateAnswer(CreateSessionDescriptionObserver* observer,
170 const MediaConstraintsInterface* constraints);
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 SetLocalDescription(SessionDescriptionInterface* desc,
173 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000174 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000175 bool SetRemoteDescription(SessionDescriptionInterface* desc,
176 std::string* err_desc);
177 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000178
mallinath@webrtc.org3d81b1b2014-09-09 14:38:10 +0000179 bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000180
honghaiz1f429e32015-09-28 07:57:34 -0700181 cricket::IceConfig ParseIceConfig(
182 const PeerConnectionInterface::RTCConfiguration& config) const;
183
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000184 const SessionDescriptionInterface* local_description() const {
185 return local_desc_.get();
186 }
187 const SessionDescriptionInterface* remote_description() const {
188 return remote_desc_.get();
189 }
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000190 // TODO(pthatcher): Cleanup the distinction between
191 // SessionDescription and SessionDescriptionInterface and remove
192 // these if possible.
193 const cricket::SessionDescription* base_local_description() const {
194 return BaseSession::local_description();
195 }
196 const cricket::SessionDescription* base_remote_description() const {
197 return BaseSession::remote_description();
198 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000199
200 // Get the id used as a media stream track's "id" field from ssrc.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200201 virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
202 virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000203
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000204 // AudioMediaProviderInterface implementation.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200205 void SetAudioPlayout(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000206 bool enable,
207 cricket::AudioRenderer* renderer) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200208 void SetAudioSend(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000209 bool enable,
210 const cricket::AudioOptions& options,
211 cricket::AudioRenderer* renderer) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200212 void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000213
214 // Implements VideoMediaProviderInterface.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200215 bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override;
216 void SetVideoPlayout(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000217 bool enable,
218 cricket::VideoRenderer* renderer) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200219 void SetVideoSend(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000220 bool enable,
221 const cricket::VideoOptions* options) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000222
223 // Implements DtmfProviderInterface.
224 virtual bool CanInsertDtmf(const std::string& track_id);
225 virtual bool InsertDtmf(const std::string& track_id,
226 int code, int duration);
227 virtual sigslot::signal0<>* GetOnDestroyedSignal();
228
wu@webrtc.org78187522013-10-07 23:32:02 +0000229 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000230 bool SendData(const cricket::SendDataParams& params,
231 const rtc::Buffer& payload,
232 cricket::SendDataResult* result) override;
233 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
234 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
235 void AddSctpDataStream(int sid) override;
236 void RemoveSctpDataStream(int sid) override;
237 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02 +0000238
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000239 // Returns stats for all channels of all transports.
240 // This avoids exposing the internal structures used to track them.
241 virtual bool GetTransportStats(cricket::SessionStats* stats);
242
deadbeefcbecd352015-09-23 11:50:27 -0700243 // Get stats for a specific channel
244 bool GetChannelTransportStats(cricket::BaseChannel* ch,
245 cricket::SessionStats* stats);
246
247 // virtual so it can be mocked in unit tests
248 virtual bool GetLocalCertificate(
249 const std::string& transport_name,
250 rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
251
252 // Caller owns returned certificate
253 virtual bool GetRemoteSSLCertificate(const std::string& transport_name,
254 rtc::SSLCertificate** cert);
255
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000256 // Implements DataChannelFactory.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000257 rtc::scoped_refptr<DataChannel> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000258 const std::string& label,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000259 const InternalDataChannelInit* config) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000260
261 cricket::DataChannelType data_channel_type() const;
262
wu@webrtc.org91053e72013-08-10 07:18:04 +0000263 bool IceRestartPending() const;
264
265 void ResetIceRestartLatch();
266
Henrik Boströmd8281982015-08-27 10:12:24 +0200267 // Called when an RTCCertificate is generated or retrieved by
wu@webrtc.org91053e72013-08-10 07:18:04 +0000268 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
Henrik Boströmd8281982015-08-27 10:12:24 +0200269 void OnCertificateReady(
270 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +0000271 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000272
273 // For unit test.
Henrik Boströmd8281982015-08-27 10:12:24 +0200274 bool waiting_for_certificate_for_testing() const;
deadbeefcbecd352015-09-23 11:50:27 -0700275 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000276
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000277 void set_metrics_observer(
278 webrtc::MetricsObserverInterface* metrics_observer) {
279 metrics_observer_ = metrics_observer;
280 }
281
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000282 private:
283 // Indicates the type of SessionDescription in a call to SetLocalDescription
284 // and SetRemoteDescription.
285 enum Action {
286 kOffer,
287 kPrAnswer,
288 kAnswer,
289 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000290
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000291 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000292 std::string* err_desc);
293 static Action GetAction(const std::string& type);
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000294 // Push the media parts of the local or remote session description
295 // down to all of the channels.
296 bool PushdownMediaDescription(cricket::ContentAction action,
297 cricket::ContentSource source,
298 std::string* error_desc);
299
deadbeefcbecd352015-09-23 11:50:27 -0700300 cricket::BaseChannel* GetChannel(const std::string& content_name);
301 // Cause all the BaseChannels in the bundle group to have the same
302 // transport channel.
303 bool EnableBundle(const cricket::ContentGroup& bundle);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000304
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000305 // Enables media channels to allow sending of media.
306 void EnableChannels();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000307 // Returns the media index for a local ice candidate given the content name.
308 // Returns false if the local session description does not have a media
309 // content called |content_name|.
310 bool GetLocalCandidateMediaIndex(const std::string& content_name,
311 int* sdp_mline_index);
312 // Uses all remote candidates in |remote_desc| in this session.
313 bool UseCandidatesInSessionDescription(
314 const SessionDescriptionInterface* remote_desc);
315 // Uses |candidate| in this session.
316 bool UseCandidate(const IceCandidateInterface* candidate);
317 // Deletes the corresponding channel of contents that don't exist in |desc|.
318 // |desc| can be null. This means that all channels are deleted.
deadbeefcbecd352015-09-23 11:50:27 -0700319 void RemoveUnusedChannels(const cricket::SessionDescription* desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000320
321 // Allocates media channels based on the |desc|. If |desc| doesn't have
322 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
323 // This method will also delete any existing media channels before creating.
324 bool CreateChannels(const cricket::SessionDescription* desc);
325
326 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000327 bool CreateVoiceChannel(const cricket::ContentInfo* content);
328 bool CreateVideoChannel(const cricket::ContentInfo* content);
329 bool CreateDataChannel(const cricket::ContentInfo* content);
330
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000331 // Copy the candidates from |saved_candidates_| to |dest_desc|.
332 // The |saved_candidates_| will be cleared after this function call.
333 void CopySavedCandidates(SessionDescriptionInterface* dest_desc);
334
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000335 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
336 // messages.
337 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
338 const cricket::ReceiveDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000339 const rtc::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000340
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000341 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000342 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
Peter Thatcher54360512015-07-08 11:08:35 -0700343 void SetIceConnectionReceiving(bool receiving);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000344
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000345 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000346 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000347 // Below methods are helper methods which verifies SDP.
348 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
349 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000350 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000351
352 // Check if a call to SetLocalDescription is acceptable with |action|.
353 bool ExpectSetLocalDescription(Action action);
354 // Check if a call to SetRemoteDescription is acceptable with |action|.
355 bool ExpectSetRemoteDescription(Action action);
356 // Verifies a=setup attribute as per RFC 5763.
357 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
358 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000359
jiayl@webrtc.orge10d28c2014-07-17 17:07:49 +0000360 // Returns true if we are ready to push down the remote candidate.
361 // |remote_desc| is the new remote description, or NULL if the current remote
362 // description should be used. Output |valid| is true if the candidate media
363 // index is valid.
364 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
365 const SessionDescriptionInterface* remote_desc,
366 bool* valid);
367
deadbeefcbecd352015-09-23 11:50:27 -0700368 void OnTransportControllerConnectionState(cricket::IceConnectionState state);
369 void OnTransportControllerReceiving(bool receiving);
370 void OnTransportControllerGatheringState(cricket::IceGatheringState state);
371 void OnTransportControllerCandidatesGathered(
372 const std::string& transport_name,
373 const cricket::Candidates& candidates);
374
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000375 std::string GetSessionErrorMsg();
376
deadbeefcbecd352015-09-23 11:50:27 -0700377 // Invoked when TransportController connection completion is signaled.
378 // Reports stats for all transports in use.
379 void ReportTransportStats();
380
381 // Gather the usage of IPv4/IPv6 as best connection.
jbauchac8869e2015-07-03 01:36:14 -0700382 void ReportBestConnectionState(const cricket::TransportStats& stats);
383
384 void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000385
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200386 rtc::scoped_ptr<MediaControllerInterface> media_controller_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000387 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
388 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
389 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000390 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000391 MediaStreamSignaling* mediastream_signaling_;
392 IceObserver* ice_observer_;
393 PeerConnectionInterface::IceConnectionState ice_connection_state_;
Peter Thatcher54360512015-07-08 11:08:35 -0700394 bool ice_connection_receiving_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000395 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
396 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000397 // Candidates that arrived before the remote description was set.
398 std::vector<IceCandidateInterface*> saved_candidates_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000399 // If the remote peer is using a older version of implementation.
400 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000401 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000402 // Specifies which kind of data channel is allowed. This is controlled
403 // by the chrome command-line flag and constraints:
404 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
405 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
406 // not set or false, SCTP is allowed (DCT_SCTP);
407 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
408 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
409 cricket::DataChannelType data_channel_type_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000410 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000411
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000412 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04 +0000413 webrtc_session_desc_factory_;
414
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000415 sigslot::signal0<> SignalVoiceChannelDestroyed;
416 sigslot::signal0<> SignalVideoChannelDestroyed;
417 sigslot::signal0<> SignalDataChannelDestroyed;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000418
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000419 // Member variables for caching global options.
420 cricket::AudioOptions audio_options_;
421 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000422 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000423
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000424 // Declares the bundle policy for the WebRTCSession.
425 PeerConnectionInterface::BundlePolicy bundle_policy_;
426
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700427 // Declares the RTCP mux policy for the WebRTCSession.
428 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
429
henrikg3c089d72015-09-16 05:37:44 -0700430 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
wu@webrtc.org364f2042013-11-20 21:49:41 +0000431};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000432} // namespace webrtc
433
434#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_