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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"
deadbeefd59daf82015-10-14 15:02:44 -070041#include "webrtc/p2p/base/transportcontroller.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
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
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000154 WebRtcSession(cricket::ChannelManager* channel_manager,
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;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000253
254 // Implements VideoMediaProviderInterface.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200255 bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override;
256 void SetVideoPlayout(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000257 bool enable,
258 cricket::VideoRenderer* renderer) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200259 void SetVideoSend(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000260 bool enable,
261 const cricket::VideoOptions* options) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000262
263 // Implements DtmfProviderInterface.
264 virtual bool CanInsertDtmf(const std::string& track_id);
265 virtual bool InsertDtmf(const std::string& track_id,
266 int code, int duration);
267 virtual sigslot::signal0<>* GetOnDestroyedSignal();
268
wu@webrtc.org78187522013-10-07 23:32:02 +0000269 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000270 bool SendData(const cricket::SendDataParams& params,
271 const rtc::Buffer& payload,
272 cricket::SendDataResult* result) override;
273 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
274 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
275 void AddSctpDataStream(int sid) override;
276 void RemoveSctpDataStream(int sid) override;
277 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02 +0000278
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000279 // Returns stats for all channels of all transports.
280 // This avoids exposing the internal structures used to track them.
deadbeefd59daf82015-10-14 15:02:44 -0700281 virtual bool GetTransportStats(SessionStats* stats);
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000282
deadbeefcbecd352015-09-23 11:50:27 -0700283 // Get stats for a specific channel
deadbeefd59daf82015-10-14 15:02:44 -0700284 bool GetChannelTransportStats(cricket::BaseChannel* ch, SessionStats* stats);
deadbeefcbecd352015-09-23 11:50:27 -0700285
286 // virtual so it can be mocked in unit tests
287 virtual bool GetLocalCertificate(
288 const std::string& transport_name,
289 rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
290
291 // Caller owns returned certificate
292 virtual bool GetRemoteSSLCertificate(const std::string& transport_name,
293 rtc::SSLCertificate** cert);
294
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000295 cricket::DataChannelType data_channel_type() const;
296
wu@webrtc.org91053e72013-08-10 07:18:04 +0000297 bool IceRestartPending() const;
298
299 void ResetIceRestartLatch();
300
Henrik Boströmd8281982015-08-27 10:12:24 +0200301 // Called when an RTCCertificate is generated or retrieved by
wu@webrtc.org91053e72013-08-10 07:18:04 +0000302 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
Henrik Boströmd8281982015-08-27 10:12:24 +0200303 void OnCertificateReady(
304 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +0000305 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000306
307 // For unit test.
Henrik Boströmd8281982015-08-27 10:12:24 +0200308 bool waiting_for_certificate_for_testing() const;
deadbeefcbecd352015-09-23 11:50:27 -0700309 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000310
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000311 void set_metrics_observer(
312 webrtc::MetricsObserverInterface* metrics_observer) {
313 metrics_observer_ = metrics_observer;
314 }
315
deadbeefab9b2d12015-10-14 11:33:11 -0700316 // Called when voice_channel_, video_channel_ and data_channel_ are created
317 // and destroyed. As a result of, for example, setting a new description.
318 sigslot::signal0<> SignalVoiceChannelCreated;
319 sigslot::signal0<> SignalVoiceChannelDestroyed;
320 sigslot::signal0<> SignalVideoChannelCreated;
321 sigslot::signal0<> SignalVideoChannelDestroyed;
322 sigslot::signal0<> SignalDataChannelCreated;
323 sigslot::signal0<> SignalDataChannelDestroyed;
324
325 // Called when a valid data channel OPEN message is received.
326 // std::string represents the data channel label.
327 sigslot::signal2<const std::string&, const InternalDataChannelInit&>
328 SignalDataChannelOpenMessage;
329
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000330 private:
331 // Indicates the type of SessionDescription in a call to SetLocalDescription
332 // and SetRemoteDescription.
333 enum Action {
334 kOffer,
335 kPrAnswer,
336 kAnswer,
337 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000338
deadbeefd59daf82015-10-14 15:02:44 -0700339 // Log session state.
340 void LogState(State old_state, State new_state);
341
342 // Updates the state, signaling if necessary.
343 virtual void SetState(State state);
344
345 // Updates the error state, signaling if necessary.
346 // TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|.
347 virtual void SetError(Error error, const std::string& error_desc);
348
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000349 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000350 std::string* err_desc);
351 static Action GetAction(const std::string& type);
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000352 // Push the media parts of the local or remote session description
353 // down to all of the channels.
354 bool PushdownMediaDescription(cricket::ContentAction action,
355 cricket::ContentSource source,
356 std::string* error_desc);
357
deadbeefd59daf82015-10-14 15:02:44 -0700358 bool PushdownTransportDescription(cricket::ContentSource source,
359 cricket::ContentAction action,
360 std::string* error_desc);
361
362 // Helper methods to push local and remote transport descriptions.
363 bool PushdownLocalTransportDescription(
364 const cricket::SessionDescription* sdesc,
365 cricket::ContentAction action,
366 std::string* error_desc);
367 bool PushdownRemoteTransportDescription(
368 const cricket::SessionDescription* sdesc,
369 cricket::ContentAction action,
370 std::string* error_desc);
371
372 // Returns true and the TransportInfo of the given |content_name|
373 // from |description|. Returns false if it's not available.
374 static bool GetTransportDescription(
375 const cricket::SessionDescription* description,
376 const std::string& content_name,
377 cricket::TransportDescription* info);
378
deadbeefcbecd352015-09-23 11:50:27 -0700379 cricket::BaseChannel* GetChannel(const std::string& content_name);
380 // Cause all the BaseChannels in the bundle group to have the same
381 // transport channel.
382 bool EnableBundle(const cricket::ContentGroup& bundle);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000383
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000384 // Enables media channels to allow sending of media.
385 void EnableChannels();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000386 // Returns the media index for a local ice candidate given the content name.
387 // Returns false if the local session description does not have a media
388 // content called |content_name|.
389 bool GetLocalCandidateMediaIndex(const std::string& content_name,
390 int* sdp_mline_index);
391 // Uses all remote candidates in |remote_desc| in this session.
392 bool UseCandidatesInSessionDescription(
393 const SessionDescriptionInterface* remote_desc);
394 // Uses |candidate| in this session.
395 bool UseCandidate(const IceCandidateInterface* candidate);
396 // Deletes the corresponding channel of contents that don't exist in |desc|.
397 // |desc| can be null. This means that all channels are deleted.
deadbeefcbecd352015-09-23 11:50:27 -0700398 void RemoveUnusedChannels(const cricket::SessionDescription* desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000399
400 // Allocates media channels based on the |desc|. If |desc| doesn't have
401 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
402 // This method will also delete any existing media channels before creating.
403 bool CreateChannels(const cricket::SessionDescription* desc);
404
405 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000406 bool CreateVoiceChannel(const cricket::ContentInfo* content);
407 bool CreateVideoChannel(const cricket::ContentInfo* content);
408 bool CreateDataChannel(const cricket::ContentInfo* content);
409
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000410 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
411 // messages.
412 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
413 const cricket::ReceiveDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000414 const rtc::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000415
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000416 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000417 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
Peter Thatcher54360512015-07-08 11:08:35 -0700418 void SetIceConnectionReceiving(bool receiving);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000419
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000420 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000421 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000422 // Below methods are helper methods which verifies SDP.
423 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
424 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000425 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000426
427 // Check if a call to SetLocalDescription is acceptable with |action|.
428 bool ExpectSetLocalDescription(Action action);
429 // Check if a call to SetRemoteDescription is acceptable with |action|.
430 bool ExpectSetRemoteDescription(Action action);
431 // Verifies a=setup attribute as per RFC 5763.
432 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
433 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000434
jiayl@webrtc.orge10d28c2014-07-17 17:07:49 +0000435 // Returns true if we are ready to push down the remote candidate.
436 // |remote_desc| is the new remote description, or NULL if the current remote
437 // description should be used. Output |valid| is true if the candidate media
438 // index is valid.
439 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
440 const SessionDescriptionInterface* remote_desc,
441 bool* valid);
442
deadbeefcbecd352015-09-23 11:50:27 -0700443 void OnTransportControllerConnectionState(cricket::IceConnectionState state);
444 void OnTransportControllerReceiving(bool receiving);
445 void OnTransportControllerGatheringState(cricket::IceGatheringState state);
446 void OnTransportControllerCandidatesGathered(
447 const std::string& transport_name,
448 const cricket::Candidates& candidates);
449
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000450 std::string GetSessionErrorMsg();
451
deadbeefcbecd352015-09-23 11:50:27 -0700452 // Invoked when TransportController connection completion is signaled.
453 // Reports stats for all transports in use.
454 void ReportTransportStats();
455
456 // Gather the usage of IPv4/IPv6 as best connection.
jbauchac8869e2015-07-03 01:36:14 -0700457 void ReportBestConnectionState(const cricket::TransportStats& stats);
458
459 void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000460
deadbeefd59daf82015-10-14 15:02:44 -0700461 rtc::Thread* const signaling_thread_;
462 rtc::Thread* const worker_thread_;
463 cricket::PortAllocator* const port_allocator_;
464
465 State state_ = STATE_INIT;
466 Error error_ = ERROR_NONE;
467 std::string error_desc_;
468
469 const std::string sid_;
470 bool initial_offerer_ = false;
471
472 rtc::scoped_ptr<cricket::TransportController> transport_controller_;
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200473 rtc::scoped_ptr<MediaControllerInterface> media_controller_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000474 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
475 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
476 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000477 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000478 IceObserver* ice_observer_;
479 PeerConnectionInterface::IceConnectionState ice_connection_state_;
Peter Thatcher54360512015-07-08 11:08:35 -0700480 bool ice_connection_receiving_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000481 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
482 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000483 // If the remote peer is using a older version of implementation.
484 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000485 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000486 // Specifies which kind of data channel is allowed. This is controlled
487 // by the chrome command-line flag and constraints:
488 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
489 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
490 // not set or false, SCTP is allowed (DCT_SCTP);
491 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
492 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
493 cricket::DataChannelType data_channel_type_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000494 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000495
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000496 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04 +0000497 webrtc_session_desc_factory_;
498
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000499 // Member variables for caching global options.
500 cricket::AudioOptions audio_options_;
501 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000502 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000503
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000504 // Declares the bundle policy for the WebRTCSession.
505 PeerConnectionInterface::BundlePolicy bundle_policy_;
506
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700507 // Declares the RTCP mux policy for the WebRTCSession.
508 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
509
henrikg3c089d72015-09-16 05:37:44 -0700510 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
wu@webrtc.org364f2042013-11-20 21:49:41 +0000511};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000512} // namespace webrtc
513
514#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_