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henrike@webrtc.org28e20752013-07-10 00:45:361/*
2 * libjingle
jlmiller@webrtc.org5f93d0a2015-01-20 21:36:133 * Copyright 2012 Google Inc.
henrike@webrtc.org28e20752013-07-10 00:45:364 *
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 18:50:2732#include <vector>
henrike@webrtc.org28e20752013-07-10 00:45:3633
buildbot@webrtc.orga09a9992014-08-13 17:26:0834#include "talk/app/webrtc/datachannel.h"
henrike@webrtc.org28e20752013-07-10 00:45:3635#include "talk/app/webrtc/dtmfsender.h"
Fredrik Solenberg709ed672015-09-15 10:26:3336#include "talk/app/webrtc/mediacontroller.h"
henrike@webrtc.org28e20752013-07-10 00:45:3637#include "talk/app/webrtc/mediastreamprovider.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:0838#include "talk/app/webrtc/peerconnectioninterface.h"
henrike@webrtc.org28e20752013-07-10 00:45:3639#include "talk/app/webrtc/statstypes.h"
henrike@webrtc.org28e20752013-07-10 00:45:3640#include "talk/media/base/mediachannel.h"
henrike@webrtc.org28e20752013-07-10 00:45:3641#include "talk/session/media/mediasession.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:0842#include "webrtc/base/sigslot.h"
Henrik Boström5e56c592015-08-11 08:33:1343#include "webrtc/base/sslidentity.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:0844#include "webrtc/base/thread.h"
Tommif888bb52015-12-12 00:37:0145#include "webrtc/p2p/base/transportcontroller.h"
henrike@webrtc.org28e20752013-07-10 00:45:3646
47namespace cricket {
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:2848
henrike@webrtc.org28e20752013-07-10 00:45:3649class ChannelManager;
50class DataChannel;
51class StatsReport;
henrike@webrtc.org28e20752013-07-10 00:45:3652class VideoCapturer;
henrike@webrtc.org28e20752013-07-10 00:45:3653class VideoChannel;
54class VoiceChannel;
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:2855
henrike@webrtc.org28e20752013-07-10 00:45:3656} // namespace cricket
57
58namespace webrtc {
buildbot@webrtc.org41451d42014-05-03 05:39:4559
henrike@webrtc.org28e20752013-07-10 00:45:3660class IceRestartAnswerLatch;
buildbot@webrtc.org41451d42014-05-03 05:39:4561class JsepIceCandidate;
henrike@webrtc.org28e20752013-07-10 00:45:3662class MediaStreamSignaling;
wu@webrtc.org91053e72013-08-10 07:18:0463class WebRtcSessionDescriptionFactory;
henrike@webrtc.org28e20752013-07-10 00:45:3664
henrike@webrtc.org1e09a712013-07-26 19:17:5965extern const char kBundleWithoutRtcpMux[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:5466extern const char kCreateChannelFailed[];
henrike@webrtc.org28e20752013-07-10 00:45:3667extern const char kInvalidCandidates[];
68extern const char kInvalidSdp[];
69extern const char kMlineMismatch[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:5470extern const char kPushDownTDFailed[];
henrike@webrtc.orgb90991d2014-03-04 19:54:5771extern const char kSdpWithoutDtlsFingerprint[];
72extern const char kSdpWithoutSdesCrypto[];
mallinath@webrtc.org19f27e62013-10-13 17:18:2773extern const char kSdpWithoutIceUfragPwd[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:5474extern const char kSdpWithoutSdesAndDtlsDisabled[];
henrike@webrtc.org28e20752013-07-10 00:45:3675extern const char kSessionError[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:5476extern const char kSessionErrorDesc[];
pthatcher@webrtc.org4eeef582015-03-16 19:34:2377extern const char kDtlsSetupFailureRtp[];
78extern const char kDtlsSetupFailureRtcp[];
deadbeefcbecd352015-09-23 18:50:2779extern const char kEnableBundleFailed[];
80
buildbot@webrtc.org53df88c2014-08-07 22:46:0181// 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 00:45:3684
85// ICE state callback interface.
86class IceObserver {
87 public:
wu@webrtc.org364f2042013-11-20 21:49:4188 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:3689 // Called any time the IceConnectionState changes
Peter Thatcher54360512015-07-08 18:08:3590 // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to
91 // conform to the w3c standard.
henrike@webrtc.org28e20752013-07-10 00:45:3692 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 18:08:35104 // Called whenever the state changes between receiving and not receiving.
105 virtual void OnIceConnectionReceivingChange(bool receiving) {}
106
henrike@webrtc.org28e20752013-07-10 00:45:36107 protected:
108 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41109
110 private:
henrikg3c089d72015-09-16 12:37:44111 RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36112};
113
deadbeefd59daf82015-10-14 22:02:44114// 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 00:45:36133 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02134 public DtmfProviderInterface,
deadbeefd59daf82015-10-14 22:02:44135 public DataChannelProviderInterface,
136 public sigslot::has_slots<> {
henrike@webrtc.org28e20752013-07-10 00:45:36137 public:
deadbeefd59daf82015-10-14 22:02:44138 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 14:26:07154 WebRtcSession(webrtc::MediaControllerInterface* media_controller,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52155 rtc::Thread* signaling_thread,
156 rtc::Thread* worker_thread,
deadbeefab9b2d12015-10-14 18:33:11157 cricket::PortAllocator* port_allocator);
henrike@webrtc.org28e20752013-07-10 00:45:36158 virtual ~WebRtcSession();
159
deadbeefd59daf82015-10-14 22:02:44160 // 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 10:44:23168 bool Initialize(
169 const PeerConnectionFactoryInterface::Options& options,
170 const MediaConstraintsInterface* constraints,
Henrik Boström5e56c592015-08-11 08:33:13171 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
Henrik Lundin64dad832015-05-11 10:44:23172 const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
henrike@webrtc.org28e20752013-07-10 00:45:36173 // Deletes the voice, video and data channel and changes the session state
deadbeefd59daf82015-10-14 22:02:44174 // 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 00:45:36188
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 19:54:57203 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
204 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36205
sergeyu@chromium.org0be6aa02013-08-23 23:21:25206 // Get current ssl role from transport.
Taylor Brandstetterf475d3652016-01-08 23:35:57207 bool GetSslRole(const std::string& transport_name, rtc::SSLRole* role);
208
209 // Get current SSL role for this channel's transport.
210 // If |transport| is null, returns false.
211 bool GetSslRole(const cricket::BaseChannel* channel, rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25212
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16213 void CreateOffer(
214 CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 18:33:11215 const PeerConnectionInterface::RTCOfferAnswerOptions& options,
216 const cricket::MediaSessionOptions& session_options);
wu@webrtc.org91053e72013-08-10 07:18:04217 void CreateAnswer(CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 18:33:11218 const MediaConstraintsInterface* constraints,
219 const cricket::MediaSessionOptions& session_options);
henrike@webrtc.org28654cb2013-07-22 21:07:49220 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36221 bool SetLocalDescription(SessionDescriptionInterface* desc,
222 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49223 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36224 bool SetRemoteDescription(SessionDescriptionInterface* desc,
225 std::string* err_desc);
226 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45227
mallinath@webrtc.org3d81b1b2014-09-09 14:38:10228 bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
buildbot@webrtc.org41451d42014-05-03 05:39:45229
honghaiz1f429e32015-09-28 14:57:34230 cricket::IceConfig ParseIceConfig(
231 const PeerConnectionInterface::RTCConfiguration& config) const;
232
deadbeefd59daf82015-10-14 22:02:44233 void SetIceConfig(const cricket::IceConfig& ice_config);
234
235 // Start gathering candidates for any new transports, or transports doing an
236 // ICE restart.
237 void MaybeStartGathering();
238
henrike@webrtc.org28e20752013-07-10 00:45:36239 const SessionDescriptionInterface* local_description() const {
240 return local_desc_.get();
241 }
242 const SessionDescriptionInterface* remote_description() const {
243 return remote_desc_.get();
244 }
245
246 // Get the id used as a media stream track's "id" field from ssrc.
Peter Boström0c4e06b2015-10-07 10:23:21247 virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
248 virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
xians@webrtc.org4cb01282014-06-12 14:57:05249
henrike@webrtc.org28e20752013-07-10 00:45:36250 // AudioMediaProviderInterface implementation.
solenbergd4cec0d2015-10-09 15:55:48251 void SetAudioPlayout(uint32_t ssrc, bool enable) override;
Peter Boström0c4e06b2015-10-07 10:23:21252 void SetAudioSend(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35253 bool enable,
254 const cricket::AudioOptions& options,
255 cricket::AudioRenderer* renderer) override;
Peter Boström0c4e06b2015-10-07 10:23:21256 void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
Tommif888bb52015-12-12 00:37:01257 void SetRawAudioSink(uint32_t ssrc,
258 rtc::scoped_ptr<AudioSinkInterface> sink) override;
henrike@webrtc.org28e20752013-07-10 00:45:36259
260 // Implements VideoMediaProviderInterface.
Peter Boström0c4e06b2015-10-07 10:23:21261 bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override;
262 void SetVideoPlayout(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35263 bool enable,
264 cricket::VideoRenderer* renderer) override;
Peter Boström0c4e06b2015-10-07 10:23:21265 void SetVideoSend(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35266 bool enable,
267 const cricket::VideoOptions* options) override;
henrike@webrtc.org28e20752013-07-10 00:45:36268
269 // Implements DtmfProviderInterface.
270 virtual bool CanInsertDtmf(const std::string& track_id);
271 virtual bool InsertDtmf(const std::string& track_id,
272 int code, int duration);
273 virtual sigslot::signal0<>* GetOnDestroyedSignal();
274
wu@webrtc.org78187522013-10-07 23:32:02275 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35276 bool SendData(const cricket::SendDataParams& params,
277 const rtc::Buffer& payload,
278 cricket::SendDataResult* result) override;
279 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
280 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
281 void AddSctpDataStream(int sid) override;
282 void RemoveSctpDataStream(int sid) override;
283 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02284
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40285 // Returns stats for all channels of all transports.
286 // This avoids exposing the internal structures used to track them.
deadbeefd59daf82015-10-14 22:02:44287 virtual bool GetTransportStats(SessionStats* stats);
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40288
deadbeefcbecd352015-09-23 18:50:27289 // Get stats for a specific channel
deadbeefd59daf82015-10-14 22:02:44290 bool GetChannelTransportStats(cricket::BaseChannel* ch, SessionStats* stats);
deadbeefcbecd352015-09-23 18:50:27291
292 // virtual so it can be mocked in unit tests
293 virtual bool GetLocalCertificate(
294 const std::string& transport_name,
295 rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
296
297 // Caller owns returned certificate
298 virtual bool GetRemoteSSLCertificate(const std::string& transport_name,
299 rtc::SSLCertificate** cert);
300
henrike@webrtc.org28e20752013-07-10 00:45:36301 cricket::DataChannelType data_channel_type() const;
302
wu@webrtc.org91053e72013-08-10 07:18:04303 bool IceRestartPending() const;
304
305 void ResetIceRestartLatch();
306
Henrik Boströmd8281982015-08-27 08:12:24307 // Called when an RTCCertificate is generated or retrieved by
wu@webrtc.org91053e72013-08-10 07:18:04308 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
Henrik Boströmd8281982015-08-27 08:12:24309 void OnCertificateReady(
310 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
pthatcher@webrtc.org4eeef582015-03-16 19:34:23311 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
wu@webrtc.org91053e72013-08-10 07:18:04312
313 // For unit test.
Henrik Boströmd8281982015-08-27 08:12:24314 bool waiting_for_certificate_for_testing() const;
deadbeefcbecd352015-09-23 18:50:27315 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
wu@webrtc.org91053e72013-08-10 07:18:04316
guoweis@webrtc.org7169afd2014-12-04 17:59:29317 void set_metrics_observer(
318 webrtc::MetricsObserverInterface* metrics_observer) {
319 metrics_observer_ = metrics_observer;
320 }
321
deadbeefab9b2d12015-10-14 18:33:11322 // Called when voice_channel_, video_channel_ and data_channel_ are created
323 // and destroyed. As a result of, for example, setting a new description.
324 sigslot::signal0<> SignalVoiceChannelCreated;
325 sigslot::signal0<> SignalVoiceChannelDestroyed;
326 sigslot::signal0<> SignalVideoChannelCreated;
327 sigslot::signal0<> SignalVideoChannelDestroyed;
328 sigslot::signal0<> SignalDataChannelCreated;
329 sigslot::signal0<> SignalDataChannelDestroyed;
330
331 // Called when a valid data channel OPEN message is received.
332 // std::string represents the data channel label.
333 sigslot::signal2<const std::string&, const InternalDataChannelInit&>
334 SignalDataChannelOpenMessage;
335
henrike@webrtc.org28e20752013-07-10 00:45:36336 private:
337 // Indicates the type of SessionDescription in a call to SetLocalDescription
338 // and SetRemoteDescription.
339 enum Action {
340 kOffer,
341 kPrAnswer,
342 kAnswer,
343 };
wu@webrtc.org91053e72013-08-10 07:18:04344
deadbeefd59daf82015-10-14 22:02:44345 // Log session state.
346 void LogState(State old_state, State new_state);
347
348 // Updates the state, signaling if necessary.
349 virtual void SetState(State state);
350
351 // Updates the error state, signaling if necessary.
352 // TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|.
353 virtual void SetError(Error error, const std::string& error_desc);
354
henrike@webrtc.org28e20752013-07-10 00:45:36355 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36356 std::string* err_desc);
357 static Action GetAction(const std::string& type);
pthatcher@webrtc.org592470b2015-03-16 21:15:37358 // Push the media parts of the local or remote session description
359 // down to all of the channels.
360 bool PushdownMediaDescription(cricket::ContentAction action,
361 cricket::ContentSource source,
362 std::string* error_desc);
363
deadbeefd59daf82015-10-14 22:02:44364 bool PushdownTransportDescription(cricket::ContentSource source,
365 cricket::ContentAction action,
366 std::string* error_desc);
367
368 // Helper methods to push local and remote transport descriptions.
369 bool PushdownLocalTransportDescription(
370 const cricket::SessionDescription* sdesc,
371 cricket::ContentAction action,
372 std::string* error_desc);
373 bool PushdownRemoteTransportDescription(
374 const cricket::SessionDescription* sdesc,
375 cricket::ContentAction action,
376 std::string* error_desc);
377
378 // Returns true and the TransportInfo of the given |content_name|
379 // from |description|. Returns false if it's not available.
380 static bool GetTransportDescription(
381 const cricket::SessionDescription* description,
382 const std::string& content_name,
383 cricket::TransportDescription* info);
384
deadbeefcbecd352015-09-23 18:50:27385 cricket::BaseChannel* GetChannel(const std::string& content_name);
386 // Cause all the BaseChannels in the bundle group to have the same
387 // transport channel.
388 bool EnableBundle(const cricket::ContentGroup& bundle);
henrike@webrtc.org28e20752013-07-10 00:45:36389
henrike@webrtc.org28e20752013-07-10 00:45:36390 // Enables media channels to allow sending of media.
391 void EnableChannels();
henrike@webrtc.org28e20752013-07-10 00:45:36392 // Returns the media index for a local ice candidate given the content name.
393 // Returns false if the local session description does not have a media
394 // content called |content_name|.
395 bool GetLocalCandidateMediaIndex(const std::string& content_name,
396 int* sdp_mline_index);
397 // Uses all remote candidates in |remote_desc| in this session.
398 bool UseCandidatesInSessionDescription(
399 const SessionDescriptionInterface* remote_desc);
400 // Uses |candidate| in this session.
401 bool UseCandidate(const IceCandidateInterface* candidate);
402 // Deletes the corresponding channel of contents that don't exist in |desc|.
403 // |desc| can be null. This means that all channels are deleted.
deadbeefcbecd352015-09-23 18:50:27404 void RemoveUnusedChannels(const cricket::SessionDescription* desc);
henrike@webrtc.org28e20752013-07-10 00:45:36405
406 // Allocates media channels based on the |desc|. If |desc| doesn't have
407 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
408 // This method will also delete any existing media channels before creating.
409 bool CreateChannels(const cricket::SessionDescription* desc);
410
411 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59412 bool CreateVoiceChannel(const cricket::ContentInfo* content);
413 bool CreateVideoChannel(const cricket::ContentInfo* content);
414 bool CreateDataChannel(const cricket::ContentInfo* content);
415
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58416 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
417 // messages.
418 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
419 const cricket::ReceiveDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52420 const rtc::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36421
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54422 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36423 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
Peter Thatcher54360512015-07-08 18:08:35424 void SetIceConnectionReceiving(bool receiving);
henrike@webrtc.org28e20752013-07-10 00:45:36425
sergeyu@chromium.org0be6aa02013-08-23 23:21:25426 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59427 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25428 // Below methods are helper methods which verifies SDP.
429 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
430 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54431 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25432
433 // Check if a call to SetLocalDescription is acceptable with |action|.
434 bool ExpectSetLocalDescription(Action action);
435 // Check if a call to SetRemoteDescription is acceptable with |action|.
436 bool ExpectSetRemoteDescription(Action action);
437 // Verifies a=setup attribute as per RFC 5763.
438 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
439 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59440
jiayl@webrtc.orge10d28c2014-07-17 17:07:49441 // Returns true if we are ready to push down the remote candidate.
442 // |remote_desc| is the new remote description, or NULL if the current remote
443 // description should be used. Output |valid| is true if the candidate media
444 // index is valid.
445 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
446 const SessionDescriptionInterface* remote_desc,
447 bool* valid);
448
deadbeefcbecd352015-09-23 18:50:27449 void OnTransportControllerConnectionState(cricket::IceConnectionState state);
450 void OnTransportControllerReceiving(bool receiving);
451 void OnTransportControllerGatheringState(cricket::IceGatheringState state);
452 void OnTransportControllerCandidatesGathered(
453 const std::string& transport_name,
454 const cricket::Candidates& candidates);
455
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54456 std::string GetSessionErrorMsg();
457
deadbeefcbecd352015-09-23 18:50:27458 // Invoked when TransportController connection completion is signaled.
459 // Reports stats for all transports in use.
460 void ReportTransportStats();
461
462 // Gather the usage of IPv4/IPv6 as best connection.
jbauchac8869e2015-07-03 08:36:14463 void ReportBestConnectionState(const cricket::TransportStats& stats);
464
465 void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
guoweis@webrtc.org7169afd2014-12-04 17:59:29466
stefanc1aeaf02015-10-15 14:26:07467 void OnSentPacket_w(cricket::TransportChannel* channel,
468 const rtc::SentPacket& sent_packet);
469
deadbeefd59daf82015-10-14 22:02:44470 rtc::Thread* const signaling_thread_;
471 rtc::Thread* const worker_thread_;
472 cricket::PortAllocator* const port_allocator_;
473
474 State state_ = STATE_INIT;
475 Error error_ = ERROR_NONE;
476 std::string error_desc_;
477
478 const std::string sid_;
479 bool initial_offerer_ = false;
480
481 rtc::scoped_ptr<cricket::TransportController> transport_controller_;
stefanc1aeaf02015-10-15 14:26:07482 MediaControllerInterface* media_controller_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52483 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
484 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
485 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36486 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36487 IceObserver* ice_observer_;
488 PeerConnectionInterface::IceConnectionState ice_connection_state_;
Peter Thatcher54360512015-07-08 18:08:35489 bool ice_connection_receiving_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52490 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
491 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36492 // If the remote peer is using a older version of implementation.
493 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10494 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36495 // Specifies which kind of data channel is allowed. This is controlled
496 // by the chrome command-line flag and constraints:
497 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
498 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
499 // not set or false, SCTP is allowed (DCT_SCTP);
500 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
501 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
502 cricket::DataChannelType data_channel_type_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52503 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04504
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52505 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04506 webrtc_session_desc_factory_;
507
henrike@webrtc.org6e3dbc22014-03-25 17:09:47508 // Member variables for caching global options.
509 cricket::AudioOptions audio_options_;
510 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29511 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47512
pthatcher@webrtc.org877ac762015-02-04 22:03:09513 // Declares the bundle policy for the WebRTCSession.
514 PeerConnectionInterface::BundlePolicy bundle_policy_;
515
Peter Thatcheraf55ccc2015-05-21 14:48:41516 // Declares the RTCP mux policy for the WebRTCSession.
517 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
518
henrikg3c089d72015-09-16 12:37:44519 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
wu@webrtc.org364f2042013-11-20 21:49:41520};
henrike@webrtc.org28e20752013-07-10 00:45:36521} // namespace webrtc
522
523#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_