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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
kjellanderb24317b2016-02-10 07:54:43 -08002 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00003 *
kjellanderb24317b2016-02-10 07:54:43 -08004 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00009 */
10
Henrik Kjellander15583c12016-02-10 10:53:12 +010011#ifndef WEBRTC_API_WEBRTCSESSION_H_
12#define WEBRTC_API_WEBRTCSESSION_H_
henrike@webrtc.org28e20752013-07-10 00:45:36 +000013
deadbeef0ed85b22016-02-23 17:24:52 -080014#include <set>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000015#include <string>
deadbeefcbecd352015-09-23 11:50:27 -070016#include <vector>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000017
Henrik Kjellander15583c12016-02-10 10:53:12 +010018#include "webrtc/api/datachannel.h"
19#include "webrtc/api/dtmfsender.h"
20#include "webrtc/api/mediacontroller.h"
21#include "webrtc/api/mediastreamprovider.h"
22#include "webrtc/api/peerconnectioninterface.h"
23#include "webrtc/api/statstypes.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000024#include "webrtc/base/sigslot.h"
Henrik Boström5e56c592015-08-11 10:33:13 +020025#include "webrtc/base/sslidentity.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000026#include "webrtc/base/thread.h"
kjellandera96e2d72016-02-04 23:52:28 -080027#include "webrtc/media/base/mediachannel.h"
Tommif888bb52015-12-12 01:37:01 +010028#include "webrtc/p2p/base/transportcontroller.h"
kjellander@webrtc.org9b8df252016-02-12 06:47:59 +010029#include "webrtc/pc/mediasession.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000030
31namespace cricket {
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000032
henrike@webrtc.org28e20752013-07-10 00:45:36 +000033class ChannelManager;
34class DataChannel;
35class StatsReport;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000036class VideoCapturer;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000037class VideoChannel;
38class VoiceChannel;
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000039
henrike@webrtc.org28e20752013-07-10 00:45:36 +000040} // namespace cricket
41
42namespace webrtc {
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000043
henrike@webrtc.org28e20752013-07-10 00:45:36 +000044class IceRestartAnswerLatch;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000045class JsepIceCandidate;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000046class MediaStreamSignaling;
wu@webrtc.org91053e72013-08-10 07:18:04 +000047class WebRtcSessionDescriptionFactory;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000048
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000049extern const char kBundleWithoutRtcpMux[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000050extern const char kCreateChannelFailed[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000051extern const char kInvalidCandidates[];
52extern const char kInvalidSdp[];
53extern const char kMlineMismatch[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000054extern const char kPushDownTDFailed[];
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +000055extern const char kSdpWithoutDtlsFingerprint[];
56extern const char kSdpWithoutSdesCrypto[];
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000057extern const char kSdpWithoutIceUfragPwd[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000058extern const char kSdpWithoutSdesAndDtlsDisabled[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000059extern const char kSessionError[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000060extern const char kSessionErrorDesc[];
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +000061extern const char kDtlsSetupFailureRtp[];
62extern const char kDtlsSetupFailureRtcp[];
deadbeefcbecd352015-09-23 11:50:27 -070063extern const char kEnableBundleFailed[];
64
buildbot@webrtc.org53df88c2014-08-07 22:46:01 +000065// Maximum number of received video streams that will be processed by webrtc
66// even if they are not signalled beforehand.
67extern const int kMaxUnsignalledRecvStreams;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000068
69// ICE state callback interface.
70class IceObserver {
71 public:
wu@webrtc.org364f2042013-11-20 21:49:41 +000072 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +000073 // Called any time the IceConnectionState changes
Peter Thatcher54360512015-07-08 11:08:35 -070074 // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to
75 // conform to the w3c standard.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000076 virtual void OnIceConnectionChange(
77 PeerConnectionInterface::IceConnectionState new_state) {}
78 // Called any time the IceGatheringState changes
79 virtual void OnIceGatheringChange(
80 PeerConnectionInterface::IceGatheringState new_state) {}
81 // New Ice candidate have been found.
82 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000083
Peter Thatcher54360512015-07-08 11:08:35 -070084 // Called whenever the state changes between receiving and not receiving.
85 virtual void OnIceConnectionReceivingChange(bool receiving) {}
86
henrike@webrtc.org28e20752013-07-10 00:45:36 +000087 protected:
88 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +000089
90 private:
henrikg3c089d72015-09-16 05:37:44 -070091 RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000092};
93
deadbeefd59daf82015-10-14 15:02:44 -070094// Statistics for all the transports of the session.
95typedef std::map<std::string, cricket::TransportStats> TransportStatsMap;
96typedef std::map<std::string, std::string> ProxyTransportMap;
97
98// TODO(pthatcher): Think of a better name for this. We already have
99// a TransportStats in transport.h. Perhaps TransportsStats?
100struct SessionStats {
101 ProxyTransportMap proxy_to_transport;
102 TransportStatsMap transport_stats;
103};
104
105// A WebRtcSession manages general session state. This includes negotiation
106// of both the application-level and network-level protocols: the former
107// defines what will be sent and the latter defines how it will be sent. Each
108// network-level protocol is represented by a Transport object. Each Transport
109// participates in the network-level negotiation. The individual streams of
110// packets are represented by TransportChannels. The application-level protocol
111// is represented by SessionDecription objects.
112class WebRtcSession : public AudioProviderInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000113 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000114 public DtmfProviderInterface,
deadbeefd59daf82015-10-14 15:02:44 -0700115 public DataChannelProviderInterface,
116 public sigslot::has_slots<> {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000117 public:
deadbeefd59daf82015-10-14 15:02:44 -0700118 enum State {
119 STATE_INIT = 0,
120 STATE_SENTOFFER, // Sent offer, waiting for answer.
121 STATE_RECEIVEDOFFER, // Received an offer. Need to send answer.
122 STATE_SENTPRANSWER, // Sent provisional answer. Need to send answer.
123 STATE_RECEIVEDPRANSWER, // Received provisional answer, waiting for answer.
124 STATE_INPROGRESS, // Offer/answer exchange completed.
125 STATE_CLOSED, // Close() was called.
126 };
127
128 enum Error {
129 ERROR_NONE = 0, // no error
130 ERROR_CONTENT = 1, // channel errors in SetLocalContent/SetRemoteContent
131 ERROR_TRANSPORT = 2, // transport error of some kind
132 };
133
stefanc1aeaf02015-10-15 07:26:07 -0700134 WebRtcSession(webrtc::MediaControllerInterface* media_controller,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000135 rtc::Thread* signaling_thread,
136 rtc::Thread* worker_thread,
deadbeefab9b2d12015-10-14 11:33:11 -0700137 cricket::PortAllocator* port_allocator);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000138 virtual ~WebRtcSession();
139
deadbeefd59daf82015-10-14 15:02:44 -0700140 // These are const to allow them to be called from const methods.
141 rtc::Thread* signaling_thread() const { return signaling_thread_; }
142 rtc::Thread* worker_thread() const { return worker_thread_; }
143 cricket::PortAllocator* port_allocator() const { return port_allocator_; }
144
145 // The ID of this session.
146 const std::string& id() const { return sid_; }
147
Henrik Lundin64dad832015-05-11 12:44:23 +0200148 bool Initialize(
149 const PeerConnectionFactoryInterface::Options& options,
Henrik Boström5e56c592015-08-11 10:33:13 +0200150 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
Henrik Lundin64dad832015-05-11 12:44:23 +0200151 const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000152 // Deletes the voice, video and data channel and changes the session state
deadbeefd59daf82015-10-14 15:02:44 -0700153 // to STATE_CLOSED.
154 void Close();
155
156 // Returns true if we were the initial offerer.
157 bool initial_offerer() const { return initial_offerer_; }
158
159 // Returns the current state of the session. See the enum above for details.
160 // Each time the state changes, we will fire this signal.
161 State state() const { return state_; }
162 sigslot::signal2<WebRtcSession*, State> SignalState;
163
164 // Returns the last error in the session. See the enum above for details.
165 Error error() const { return error_; }
166 const std::string& error_desc() const { return error_desc_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000167
168 void RegisterIceObserver(IceObserver* observer) {
169 ice_observer_ = observer;
170 }
171
172 virtual cricket::VoiceChannel* voice_channel() {
173 return voice_channel_.get();
174 }
175 virtual cricket::VideoChannel* video_channel() {
176 return video_channel_.get();
177 }
178 virtual cricket::DataChannel* data_channel() {
179 return data_channel_.get();
180 }
181
deadbeef0ed85b22016-02-23 17:24:52 -0800182 cricket::BaseChannel* GetChannel(const std::string& content_name);
183
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000184 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
185 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000186
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000187 // Get current ssl role from transport.
Taylor Brandstetterf475d362016-01-08 15:35:57 -0800188 bool GetSslRole(const std::string& transport_name, rtc::SSLRole* role);
189
190 // Get current SSL role for this channel's transport.
191 // If |transport| is null, returns false.
192 bool GetSslRole(const cricket::BaseChannel* channel, rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000193
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000194 void CreateOffer(
195 CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 11:33:11 -0700196 const PeerConnectionInterface::RTCOfferAnswerOptions& options,
197 const cricket::MediaSessionOptions& session_options);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000198 void CreateAnswer(CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 11:33:11 -0700199 const cricket::MediaSessionOptions& session_options);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000200 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000201 bool SetLocalDescription(SessionDescriptionInterface* desc,
202 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000203 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000204 bool SetRemoteDescription(SessionDescriptionInterface* desc,
205 std::string* err_desc);
206 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000207
mallinath@webrtc.org3d81b1b2014-09-09 14:38:10 +0000208 bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000209
honghaiz1f429e32015-09-28 07:57:34 -0700210 cricket::IceConfig ParseIceConfig(
211 const PeerConnectionInterface::RTCConfiguration& config) const;
212
deadbeefd59daf82015-10-14 15:02:44 -0700213 void SetIceConfig(const cricket::IceConfig& ice_config);
214
215 // Start gathering candidates for any new transports, or transports doing an
216 // ICE restart.
217 void MaybeStartGathering();
218
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000219 const SessionDescriptionInterface* local_description() const {
220 return local_desc_.get();
221 }
222 const SessionDescriptionInterface* remote_description() const {
223 return remote_desc_.get();
224 }
225
226 // Get the id used as a media stream track's "id" field from ssrc.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200227 virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
228 virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000229
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000230 // AudioMediaProviderInterface implementation.
solenbergd4cec0d2015-10-09 08:55:48 -0700231 void SetAudioPlayout(uint32_t ssrc, bool enable) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200232 void SetAudioSend(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000233 bool enable,
234 const cricket::AudioOptions& options,
Taylor Brandstetter1a018dc2016-03-08 12:37:39 -0800235 cricket::AudioSource* source) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200236 void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
deadbeef2d110be2016-01-13 12:00:26 -0800237 void SetRawAudioSink(uint32_t ssrc,
238 rtc::scoped_ptr<AudioSinkInterface> sink) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000239
240 // Implements VideoMediaProviderInterface.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200241 bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override;
nisse08582ff2016-02-04 01:24:52 -0800242 void SetVideoPlayout(
243 uint32_t ssrc,
244 bool enable,
245 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200246 void SetVideoSend(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000247 bool enable,
248 const cricket::VideoOptions* options) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000249
250 // Implements DtmfProviderInterface.
251 virtual bool CanInsertDtmf(const std::string& track_id);
252 virtual bool InsertDtmf(const std::string& track_id,
253 int code, int duration);
254 virtual sigslot::signal0<>* GetOnDestroyedSignal();
255
wu@webrtc.org78187522013-10-07 23:32:02 +0000256 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000257 bool SendData(const cricket::SendDataParams& params,
258 const rtc::Buffer& payload,
259 cricket::SendDataResult* result) override;
260 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
261 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
262 void AddSctpDataStream(int sid) override;
263 void RemoveSctpDataStream(int sid) override;
264 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02 +0000265
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000266 // Returns stats for all channels of all transports.
267 // This avoids exposing the internal structures used to track them.
deadbeefd59daf82015-10-14 15:02:44 -0700268 virtual bool GetTransportStats(SessionStats* stats);
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000269
deadbeefcbecd352015-09-23 11:50:27 -0700270 // Get stats for a specific channel
deadbeefd59daf82015-10-14 15:02:44 -0700271 bool GetChannelTransportStats(cricket::BaseChannel* ch, SessionStats* stats);
deadbeefcbecd352015-09-23 11:50:27 -0700272
273 // virtual so it can be mocked in unit tests
274 virtual bool GetLocalCertificate(
275 const std::string& transport_name,
276 rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
277
278 // Caller owns returned certificate
279 virtual bool GetRemoteSSLCertificate(const std::string& transport_name,
280 rtc::SSLCertificate** cert);
281
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000282 cricket::DataChannelType data_channel_type() const;
283
deadbeef0ed85b22016-02-23 17:24:52 -0800284 bool IceRestartPending(const std::string& content_name) const;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000285
Henrik Boströmd8281982015-08-27 10:12:24 +0200286 // Called when an RTCCertificate is generated or retrieved by
wu@webrtc.org91053e72013-08-10 07:18:04 +0000287 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
Henrik Boströmd8281982015-08-27 10:12:24 +0200288 void OnCertificateReady(
289 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +0000290 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000291
292 // For unit test.
Henrik Boströmd8281982015-08-27 10:12:24 +0200293 bool waiting_for_certificate_for_testing() const;
deadbeefcbecd352015-09-23 11:50:27 -0700294 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000295
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000296 void set_metrics_observer(
297 webrtc::MetricsObserverInterface* metrics_observer) {
298 metrics_observer_ = metrics_observer;
299 }
300
deadbeefab9b2d12015-10-14 11:33:11 -0700301 // Called when voice_channel_, video_channel_ and data_channel_ are created
302 // and destroyed. As a result of, for example, setting a new description.
303 sigslot::signal0<> SignalVoiceChannelCreated;
304 sigslot::signal0<> SignalVoiceChannelDestroyed;
305 sigslot::signal0<> SignalVideoChannelCreated;
306 sigslot::signal0<> SignalVideoChannelDestroyed;
307 sigslot::signal0<> SignalDataChannelCreated;
308 sigslot::signal0<> SignalDataChannelDestroyed;
deadbeef057ecf02016-01-20 14:30:43 -0800309 // Called when the whole session is destroyed.
310 sigslot::signal0<> SignalDestroyed;
deadbeefab9b2d12015-10-14 11:33:11 -0700311
312 // Called when a valid data channel OPEN message is received.
313 // std::string represents the data channel label.
314 sigslot::signal2<const std::string&, const InternalDataChannelInit&>
315 SignalDataChannelOpenMessage;
316
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000317 private:
318 // Indicates the type of SessionDescription in a call to SetLocalDescription
319 // and SetRemoteDescription.
320 enum Action {
321 kOffer,
322 kPrAnswer,
323 kAnswer,
324 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000325
deadbeefd59daf82015-10-14 15:02:44 -0700326 // Log session state.
327 void LogState(State old_state, State new_state);
328
329 // Updates the state, signaling if necessary.
330 virtual void SetState(State state);
331
332 // Updates the error state, signaling if necessary.
333 // TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|.
334 virtual void SetError(Error error, const std::string& error_desc);
335
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000336 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000337 std::string* err_desc);
338 static Action GetAction(const std::string& type);
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000339 // Push the media parts of the local or remote session description
340 // down to all of the channels.
341 bool PushdownMediaDescription(cricket::ContentAction action,
342 cricket::ContentSource source,
343 std::string* error_desc);
344
deadbeefd59daf82015-10-14 15:02:44 -0700345 bool PushdownTransportDescription(cricket::ContentSource source,
346 cricket::ContentAction action,
347 std::string* error_desc);
348
349 // Helper methods to push local and remote transport descriptions.
350 bool PushdownLocalTransportDescription(
351 const cricket::SessionDescription* sdesc,
352 cricket::ContentAction action,
353 std::string* error_desc);
354 bool PushdownRemoteTransportDescription(
355 const cricket::SessionDescription* sdesc,
356 cricket::ContentAction action,
357 std::string* error_desc);
358
359 // Returns true and the TransportInfo of the given |content_name|
360 // from |description|. Returns false if it's not available.
361 static bool GetTransportDescription(
362 const cricket::SessionDescription* description,
363 const std::string& content_name,
364 cricket::TransportDescription* info);
365
deadbeefcbecd352015-09-23 11:50:27 -0700366 // Cause all the BaseChannels in the bundle group to have the same
367 // transport channel.
368 bool EnableBundle(const cricket::ContentGroup& bundle);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000369
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000370 // Enables media channels to allow sending of media.
371 void EnableChannels();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000372 // Returns the media index for a local ice candidate given the content name.
373 // Returns false if the local session description does not have a media
374 // content called |content_name|.
375 bool GetLocalCandidateMediaIndex(const std::string& content_name,
376 int* sdp_mline_index);
377 // Uses all remote candidates in |remote_desc| in this session.
378 bool UseCandidatesInSessionDescription(
379 const SessionDescriptionInterface* remote_desc);
380 // Uses |candidate| in this session.
381 bool UseCandidate(const IceCandidateInterface* candidate);
382 // Deletes the corresponding channel of contents that don't exist in |desc|.
383 // |desc| can be null. This means that all channels are deleted.
deadbeefcbecd352015-09-23 11:50:27 -0700384 void RemoveUnusedChannels(const cricket::SessionDescription* desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000385
386 // Allocates media channels based on the |desc|. If |desc| doesn't have
387 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
388 // This method will also delete any existing media channels before creating.
389 bool CreateChannels(const cricket::SessionDescription* desc);
390
391 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000392 bool CreateVoiceChannel(const cricket::ContentInfo* content);
393 bool CreateVideoChannel(const cricket::ContentInfo* content);
394 bool CreateDataChannel(const cricket::ContentInfo* content);
395
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000396 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
397 // messages.
398 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
399 const cricket::ReceiveDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000400 const rtc::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000401
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000402 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000403 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
Peter Thatcher54360512015-07-08 11:08:35 -0700404 void SetIceConnectionReceiving(bool receiving);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000405
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000406 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000407 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000408 // Below methods are helper methods which verifies SDP.
409 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
410 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000411 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000412
413 // Check if a call to SetLocalDescription is acceptable with |action|.
414 bool ExpectSetLocalDescription(Action action);
415 // Check if a call to SetRemoteDescription is acceptable with |action|.
416 bool ExpectSetRemoteDescription(Action action);
417 // Verifies a=setup attribute as per RFC 5763.
418 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
419 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000420
jiayl@webrtc.orge10d28c2014-07-17 17:07:49 +0000421 // Returns true if we are ready to push down the remote candidate.
422 // |remote_desc| is the new remote description, or NULL if the current remote
423 // description should be used. Output |valid| is true if the candidate media
424 // index is valid.
425 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
426 const SessionDescriptionInterface* remote_desc,
427 bool* valid);
428
deadbeefcbecd352015-09-23 11:50:27 -0700429 void OnTransportControllerConnectionState(cricket::IceConnectionState state);
430 void OnTransportControllerReceiving(bool receiving);
431 void OnTransportControllerGatheringState(cricket::IceGatheringState state);
432 void OnTransportControllerCandidatesGathered(
433 const std::string& transport_name,
tommi6f59a4f2016-03-11 14:05:09 -0800434 const cricket::Candidates& candidates);
deadbeefcbecd352015-09-23 11:50:27 -0700435
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000436 std::string GetSessionErrorMsg();
437
deadbeefcbecd352015-09-23 11:50:27 -0700438 // Invoked when TransportController connection completion is signaled.
439 // Reports stats for all transports in use.
440 void ReportTransportStats();
441
442 // Gather the usage of IPv4/IPv6 as best connection.
jbauchac8869e2015-07-03 01:36:14 -0700443 void ReportBestConnectionState(const cricket::TransportStats& stats);
444
445 void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000446
stefanc1aeaf02015-10-15 07:26:07 -0700447 void OnSentPacket_w(cricket::TransportChannel* channel,
448 const rtc::SentPacket& sent_packet);
449
deadbeefd59daf82015-10-14 15:02:44 -0700450 rtc::Thread* const signaling_thread_;
451 rtc::Thread* const worker_thread_;
452 cricket::PortAllocator* const port_allocator_;
453
454 State state_ = STATE_INIT;
455 Error error_ = ERROR_NONE;
456 std::string error_desc_;
457
458 const std::string sid_;
459 bool initial_offerer_ = false;
460
461 rtc::scoped_ptr<cricket::TransportController> transport_controller_;
stefanc1aeaf02015-10-15 07:26:07 -0700462 MediaControllerInterface* media_controller_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000463 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
464 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
465 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000466 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000467 IceObserver* ice_observer_;
468 PeerConnectionInterface::IceConnectionState ice_connection_state_;
Peter Thatcher54360512015-07-08 11:08:35 -0700469 bool ice_connection_receiving_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000470 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
471 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000472 // If the remote peer is using a older version of implementation.
473 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000474 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000475 // Specifies which kind of data channel is allowed. This is controlled
476 // by the chrome command-line flag and constraints:
477 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
478 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
479 // not set or false, SCTP is allowed (DCT_SCTP);
480 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
481 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
482 cricket::DataChannelType data_channel_type_;
deadbeef0ed85b22016-02-23 17:24:52 -0800483 // List of content names for which the remote side triggered an ICE restart.
484 std::set<std::string> pending_ice_restarts_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000485
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000486 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04 +0000487 webrtc_session_desc_factory_;
488
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000489 // Member variables for caching global options.
490 cricket::AudioOptions audio_options_;
491 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000492 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000493
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000494 // Declares the bundle policy for the WebRTCSession.
495 PeerConnectionInterface::BundlePolicy bundle_policy_;
496
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700497 // Declares the RTCP mux policy for the WebRTCSession.
498 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
499
henrikg3c089d72015-09-16 05:37:44 -0700500 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
wu@webrtc.org364f2042013-11-20 21:49:41 +0000501};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000502} // namespace webrtc
503
Henrik Kjellander15583c12016-02-10 10:53:12 +0100504#endif // WEBRTC_API_WEBRTCSESSION_H_