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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.org28e20752013-07-10 00:45:36 +000041#include "talk/session/media/mediasession.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000042#include "webrtc/base/sigslot.h"
Henrik Boström5e56c592015-08-11 10:33:13 +020043#include "webrtc/base/sslidentity.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000044#include "webrtc/base/thread.h"
Tommif888bb52015-12-12 01:37:01 +010045#include "webrtc/p2p/base/transportcontroller.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000046
47namespace cricket {
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000048
henrike@webrtc.org28e20752013-07-10 00:45:36 +000049class ChannelManager;
50class DataChannel;
51class StatsReport;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000052class 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[];
deadbeefcbecd352015-09-23 11:50:27 -070079extern const char kEnableBundleFailed[];
80
buildbot@webrtc.org53df88c2014-08-07 22:46:01 +000081// Maximum number of received video streams that will be processed by webrtc
82// even if they are not signalled beforehand.
83extern const int kMaxUnsignalledRecvStreams;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000084
85// ICE state callback interface.
86class IceObserver {
87 public:
wu@webrtc.org364f2042013-11-20 21:49:41 +000088 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +000089 // Called any time the IceConnectionState changes
Peter Thatcher54360512015-07-08 11:08:35 -070090 // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to
91 // conform to the w3c standard.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000092 virtual void OnIceConnectionChange(
93 PeerConnectionInterface::IceConnectionState new_state) {}
94 // Called any time the IceGatheringState changes
95 virtual void OnIceGatheringChange(
96 PeerConnectionInterface::IceGatheringState new_state) {}
97 // New Ice candidate have been found.
98 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
99 // All Ice candidates have been found.
100 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
101 // (via PeerConnectionObserver)
102 virtual void OnIceComplete() {}
103
Peter Thatcher54360512015-07-08 11:08:35 -0700104 // Called whenever the state changes between receiving and not receiving.
105 virtual void OnIceConnectionReceivingChange(bool receiving) {}
106
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000107 protected:
108 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +0000109
110 private:
henrikg3c089d72015-09-16 05:37:44 -0700111 RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000112};
113
deadbeefd59daf82015-10-14 15:02:44 -0700114// Statistics for all the transports of the session.
115typedef std::map<std::string, cricket::TransportStats> TransportStatsMap;
116typedef std::map<std::string, std::string> ProxyTransportMap;
117
118// TODO(pthatcher): Think of a better name for this. We already have
119// a TransportStats in transport.h. Perhaps TransportsStats?
120struct SessionStats {
121 ProxyTransportMap proxy_to_transport;
122 TransportStatsMap transport_stats;
123};
124
125// A WebRtcSession manages general session state. This includes negotiation
126// of both the application-level and network-level protocols: the former
127// defines what will be sent and the latter defines how it will be sent. Each
128// network-level protocol is represented by a Transport object. Each Transport
129// participates in the network-level negotiation. The individual streams of
130// packets are represented by TransportChannels. The application-level protocol
131// is represented by SessionDecription objects.
132class WebRtcSession : public AudioProviderInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000133 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000134 public DtmfProviderInterface,
deadbeefd59daf82015-10-14 15:02:44 -0700135 public DataChannelProviderInterface,
136 public sigslot::has_slots<> {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000137 public:
deadbeefd59daf82015-10-14 15:02:44 -0700138 enum State {
139 STATE_INIT = 0,
140 STATE_SENTOFFER, // Sent offer, waiting for answer.
141 STATE_RECEIVEDOFFER, // Received an offer. Need to send answer.
142 STATE_SENTPRANSWER, // Sent provisional answer. Need to send answer.
143 STATE_RECEIVEDPRANSWER, // Received provisional answer, waiting for answer.
144 STATE_INPROGRESS, // Offer/answer exchange completed.
145 STATE_CLOSED, // Close() was called.
146 };
147
148 enum Error {
149 ERROR_NONE = 0, // no error
150 ERROR_CONTENT = 1, // channel errors in SetLocalContent/SetRemoteContent
151 ERROR_TRANSPORT = 2, // transport error of some kind
152 };
153
stefanc1aeaf02015-10-15 07:26:07 -0700154 WebRtcSession(webrtc::MediaControllerInterface* media_controller,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000155 rtc::Thread* signaling_thread,
156 rtc::Thread* worker_thread,
deadbeefab9b2d12015-10-14 11:33:11 -0700157 cricket::PortAllocator* port_allocator);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000158 virtual ~WebRtcSession();
159
deadbeefd59daf82015-10-14 15:02:44 -0700160 // These are const to allow them to be called from const methods.
161 rtc::Thread* signaling_thread() const { return signaling_thread_; }
162 rtc::Thread* worker_thread() const { return worker_thread_; }
163 cricket::PortAllocator* port_allocator() const { return port_allocator_; }
164
165 // The ID of this session.
166 const std::string& id() const { return sid_; }
167
Henrik Lundin64dad832015-05-11 12:44:23 +0200168 bool Initialize(
169 const PeerConnectionFactoryInterface::Options& options,
170 const MediaConstraintsInterface* constraints,
Henrik Boström5e56c592015-08-11 10:33:13 +0200171 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
Henrik Lundin64dad832015-05-11 12:44:23 +0200172 const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000173 // Deletes the voice, video and data channel and changes the session state
deadbeefd59daf82015-10-14 15:02:44 -0700174 // to STATE_CLOSED.
175 void Close();
176
177 // Returns true if we were the initial offerer.
178 bool initial_offerer() const { return initial_offerer_; }
179
180 // Returns the current state of the session. See the enum above for details.
181 // Each time the state changes, we will fire this signal.
182 State state() const { return state_; }
183 sigslot::signal2<WebRtcSession*, State> SignalState;
184
185 // Returns the last error in the session. See the enum above for details.
186 Error error() const { return error_; }
187 const std::string& error_desc() const { return error_desc_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000188
189 void RegisterIceObserver(IceObserver* observer) {
190 ice_observer_ = observer;
191 }
192
193 virtual cricket::VoiceChannel* voice_channel() {
194 return voice_channel_.get();
195 }
196 virtual cricket::VideoChannel* video_channel() {
197 return video_channel_.get();
198 }
199 virtual cricket::DataChannel* data_channel() {
200 return data_channel_.get();
201 }
202
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000203 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
204 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000205
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000206 // Get current ssl role from transport.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000207 bool GetSslRole(rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000208
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000209 void CreateOffer(
210 CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 11:33:11 -0700211 const PeerConnectionInterface::RTCOfferAnswerOptions& options,
212 const cricket::MediaSessionOptions& session_options);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000213 void CreateAnswer(CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 11:33:11 -0700214 const MediaConstraintsInterface* constraints,
215 const cricket::MediaSessionOptions& session_options);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000216 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000217 bool SetLocalDescription(SessionDescriptionInterface* desc,
218 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000219 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000220 bool SetRemoteDescription(SessionDescriptionInterface* desc,
221 std::string* err_desc);
222 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000223
mallinath@webrtc.org3d81b1b2014-09-09 14:38:10 +0000224 bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000225
honghaiz1f429e32015-09-28 07:57:34 -0700226 cricket::IceConfig ParseIceConfig(
227 const PeerConnectionInterface::RTCConfiguration& config) const;
228
deadbeefd59daf82015-10-14 15:02:44 -0700229 void SetIceConfig(const cricket::IceConfig& ice_config);
230
231 // Start gathering candidates for any new transports, or transports doing an
232 // ICE restart.
233 void MaybeStartGathering();
234
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000235 const SessionDescriptionInterface* local_description() const {
236 return local_desc_.get();
237 }
238 const SessionDescriptionInterface* remote_description() const {
239 return remote_desc_.get();
240 }
241
242 // Get the id used as a media stream track's "id" field from ssrc.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200243 virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
244 virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000245
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000246 // AudioMediaProviderInterface implementation.
solenbergd4cec0d2015-10-09 08:55:48 -0700247 void SetAudioPlayout(uint32_t ssrc, bool enable) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200248 void SetAudioSend(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000249 bool enable,
250 const cricket::AudioOptions& options,
251 cricket::AudioRenderer* renderer) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200252 void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
Tommif888bb52015-12-12 01:37:01 +0100253 void SetRawAudioSink(uint32_t ssrc,
254 rtc::scoped_ptr<AudioSinkInterface> sink) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000255
256 // Implements VideoMediaProviderInterface.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200257 bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override;
258 void SetVideoPlayout(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000259 bool enable,
260 cricket::VideoRenderer* renderer) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200261 void SetVideoSend(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000262 bool enable,
263 const cricket::VideoOptions* options) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000264
265 // Implements DtmfProviderInterface.
266 virtual bool CanInsertDtmf(const std::string& track_id);
267 virtual bool InsertDtmf(const std::string& track_id,
268 int code, int duration);
269 virtual sigslot::signal0<>* GetOnDestroyedSignal();
270
wu@webrtc.org78187522013-10-07 23:32:02 +0000271 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000272 bool SendData(const cricket::SendDataParams& params,
273 const rtc::Buffer& payload,
274 cricket::SendDataResult* result) override;
275 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
276 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
277 void AddSctpDataStream(int sid) override;
278 void RemoveSctpDataStream(int sid) override;
279 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02 +0000280
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000281 // Returns stats for all channels of all transports.
282 // This avoids exposing the internal structures used to track them.
deadbeefd59daf82015-10-14 15:02:44 -0700283 virtual bool GetTransportStats(SessionStats* stats);
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000284
deadbeefcbecd352015-09-23 11:50:27 -0700285 // Get stats for a specific channel
deadbeefd59daf82015-10-14 15:02:44 -0700286 bool GetChannelTransportStats(cricket::BaseChannel* ch, SessionStats* stats);
deadbeefcbecd352015-09-23 11:50:27 -0700287
288 // virtual so it can be mocked in unit tests
289 virtual bool GetLocalCertificate(
290 const std::string& transport_name,
291 rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
292
293 // Caller owns returned certificate
294 virtual bool GetRemoteSSLCertificate(const std::string& transport_name,
295 rtc::SSLCertificate** cert);
296
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000297 cricket::DataChannelType data_channel_type() const;
298
wu@webrtc.org91053e72013-08-10 07:18:04 +0000299 bool IceRestartPending() const;
300
301 void ResetIceRestartLatch();
302
Henrik Boströmd8281982015-08-27 10:12:24 +0200303 // Called when an RTCCertificate is generated or retrieved by
wu@webrtc.org91053e72013-08-10 07:18:04 +0000304 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
Henrik Boströmd8281982015-08-27 10:12:24 +0200305 void OnCertificateReady(
306 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +0000307 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000308
309 // For unit test.
Henrik Boströmd8281982015-08-27 10:12:24 +0200310 bool waiting_for_certificate_for_testing() const;
deadbeefcbecd352015-09-23 11:50:27 -0700311 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000312
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000313 void set_metrics_observer(
314 webrtc::MetricsObserverInterface* metrics_observer) {
315 metrics_observer_ = metrics_observer;
316 }
317
deadbeefab9b2d12015-10-14 11:33:11 -0700318 // Called when voice_channel_, video_channel_ and data_channel_ are created
319 // and destroyed. As a result of, for example, setting a new description.
320 sigslot::signal0<> SignalVoiceChannelCreated;
321 sigslot::signal0<> SignalVoiceChannelDestroyed;
322 sigslot::signal0<> SignalVideoChannelCreated;
323 sigslot::signal0<> SignalVideoChannelDestroyed;
324 sigslot::signal0<> SignalDataChannelCreated;
325 sigslot::signal0<> SignalDataChannelDestroyed;
326
327 // Called when a valid data channel OPEN message is received.
328 // std::string represents the data channel label.
329 sigslot::signal2<const std::string&, const InternalDataChannelInit&>
330 SignalDataChannelOpenMessage;
331
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000332 private:
333 // Indicates the type of SessionDescription in a call to SetLocalDescription
334 // and SetRemoteDescription.
335 enum Action {
336 kOffer,
337 kPrAnswer,
338 kAnswer,
339 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000340
deadbeefd59daf82015-10-14 15:02:44 -0700341 // Log session state.
342 void LogState(State old_state, State new_state);
343
344 // Updates the state, signaling if necessary.
345 virtual void SetState(State state);
346
347 // Updates the error state, signaling if necessary.
348 // TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|.
349 virtual void SetError(Error error, const std::string& error_desc);
350
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000351 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000352 std::string* err_desc);
353 static Action GetAction(const std::string& type);
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000354 // Push the media parts of the local or remote session description
355 // down to all of the channels.
356 bool PushdownMediaDescription(cricket::ContentAction action,
357 cricket::ContentSource source,
358 std::string* error_desc);
359
deadbeefd59daf82015-10-14 15:02:44 -0700360 bool PushdownTransportDescription(cricket::ContentSource source,
361 cricket::ContentAction action,
362 std::string* error_desc);
363
364 // Helper methods to push local and remote transport descriptions.
365 bool PushdownLocalTransportDescription(
366 const cricket::SessionDescription* sdesc,
367 cricket::ContentAction action,
368 std::string* error_desc);
369 bool PushdownRemoteTransportDescription(
370 const cricket::SessionDescription* sdesc,
371 cricket::ContentAction action,
372 std::string* error_desc);
373
374 // Returns true and the TransportInfo of the given |content_name|
375 // from |description|. Returns false if it's not available.
376 static bool GetTransportDescription(
377 const cricket::SessionDescription* description,
378 const std::string& content_name,
379 cricket::TransportDescription* info);
380
deadbeefcbecd352015-09-23 11:50:27 -0700381 cricket::BaseChannel* GetChannel(const std::string& content_name);
382 // Cause all the BaseChannels in the bundle group to have the same
383 // transport channel.
384 bool EnableBundle(const cricket::ContentGroup& bundle);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000385
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000386 // Enables media channels to allow sending of media.
387 void EnableChannels();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000388 // Returns the media index for a local ice candidate given the content name.
389 // Returns false if the local session description does not have a media
390 // content called |content_name|.
391 bool GetLocalCandidateMediaIndex(const std::string& content_name,
392 int* sdp_mline_index);
393 // Uses all remote candidates in |remote_desc| in this session.
394 bool UseCandidatesInSessionDescription(
395 const SessionDescriptionInterface* remote_desc);
396 // Uses |candidate| in this session.
397 bool UseCandidate(const IceCandidateInterface* candidate);
398 // Deletes the corresponding channel of contents that don't exist in |desc|.
399 // |desc| can be null. This means that all channels are deleted.
deadbeefcbecd352015-09-23 11:50:27 -0700400 void RemoveUnusedChannels(const cricket::SessionDescription* desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000401
402 // Allocates media channels based on the |desc|. If |desc| doesn't have
403 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
404 // This method will also delete any existing media channels before creating.
405 bool CreateChannels(const cricket::SessionDescription* desc);
406
407 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000408 bool CreateVoiceChannel(const cricket::ContentInfo* content);
409 bool CreateVideoChannel(const cricket::ContentInfo* content);
410 bool CreateDataChannel(const cricket::ContentInfo* content);
411
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000412 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
413 // messages.
414 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
415 const cricket::ReceiveDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000416 const rtc::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000417
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000418 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000419 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
Peter Thatcher54360512015-07-08 11:08:35 -0700420 void SetIceConnectionReceiving(bool receiving);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000421
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000422 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000423 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000424 // Below methods are helper methods which verifies SDP.
425 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
426 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000427 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000428
429 // Check if a call to SetLocalDescription is acceptable with |action|.
430 bool ExpectSetLocalDescription(Action action);
431 // Check if a call to SetRemoteDescription is acceptable with |action|.
432 bool ExpectSetRemoteDescription(Action action);
433 // Verifies a=setup attribute as per RFC 5763.
434 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
435 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000436
jiayl@webrtc.orge10d28c2014-07-17 17:07:49 +0000437 // Returns true if we are ready to push down the remote candidate.
438 // |remote_desc| is the new remote description, or NULL if the current remote
439 // description should be used. Output |valid| is true if the candidate media
440 // index is valid.
441 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
442 const SessionDescriptionInterface* remote_desc,
443 bool* valid);
444
deadbeefcbecd352015-09-23 11:50:27 -0700445 void OnTransportControllerConnectionState(cricket::IceConnectionState state);
446 void OnTransportControllerReceiving(bool receiving);
447 void OnTransportControllerGatheringState(cricket::IceGatheringState state);
448 void OnTransportControllerCandidatesGathered(
449 const std::string& transport_name,
450 const cricket::Candidates& candidates);
451
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000452 std::string GetSessionErrorMsg();
453
deadbeefcbecd352015-09-23 11:50:27 -0700454 // Invoked when TransportController connection completion is signaled.
455 // Reports stats for all transports in use.
456 void ReportTransportStats();
457
458 // Gather the usage of IPv4/IPv6 as best connection.
jbauchac8869e2015-07-03 01:36:14 -0700459 void ReportBestConnectionState(const cricket::TransportStats& stats);
460
461 void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000462
stefanc1aeaf02015-10-15 07:26:07 -0700463 void OnSentPacket_w(cricket::TransportChannel* channel,
464 const rtc::SentPacket& sent_packet);
465
deadbeefd59daf82015-10-14 15:02:44 -0700466 rtc::Thread* const signaling_thread_;
467 rtc::Thread* const worker_thread_;
468 cricket::PortAllocator* const port_allocator_;
469
470 State state_ = STATE_INIT;
471 Error error_ = ERROR_NONE;
472 std::string error_desc_;
473
474 const std::string sid_;
475 bool initial_offerer_ = false;
476
477 rtc::scoped_ptr<cricket::TransportController> transport_controller_;
stefanc1aeaf02015-10-15 07:26:07 -0700478 MediaControllerInterface* media_controller_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000479 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
480 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
481 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000482 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000483 IceObserver* ice_observer_;
484 PeerConnectionInterface::IceConnectionState ice_connection_state_;
Peter Thatcher54360512015-07-08 11:08:35 -0700485 bool ice_connection_receiving_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000486 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
487 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000488 // If the remote peer is using a older version of implementation.
489 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000490 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000491 // Specifies which kind of data channel is allowed. This is controlled
492 // by the chrome command-line flag and constraints:
493 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
494 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
495 // not set or false, SCTP is allowed (DCT_SCTP);
496 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
497 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
498 cricket::DataChannelType data_channel_type_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000499 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000500
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000501 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04 +0000502 webrtc_session_desc_factory_;
503
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000504 // Member variables for caching global options.
505 cricket::AudioOptions audio_options_;
506 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000507 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000508
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000509 // Declares the bundle policy for the WebRTCSession.
510 PeerConnectionInterface::BundlePolicy bundle_policy_;
511
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700512 // Declares the RTCP mux policy for the WebRTCSession.
513 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
514
henrikg3c089d72015-09-16 05:37:44 -0700515 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
wu@webrtc.org364f2042013-11-20 21:49:41 +0000516};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000517} // namespace webrtc
518
519#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_