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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"
Honghai Zhang7fb69db2016-03-14 11:59:18 -070028#include "webrtc/p2p/base/candidate.h"
Tommif888bb52015-12-12 01:37:01 +010029#include "webrtc/p2p/base/transportcontroller.h"
kjellander@webrtc.org9b8df252016-02-12 06:47:59 +010030#include "webrtc/pc/mediasession.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000031
32namespace cricket {
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000033
henrike@webrtc.org28e20752013-07-10 00:45:36 +000034class ChannelManager;
35class DataChannel;
36class StatsReport;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000037class VideoCapturer;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000038class VideoChannel;
39class VoiceChannel;
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000040
henrike@webrtc.org28e20752013-07-10 00:45:36 +000041} // namespace cricket
42
43namespace webrtc {
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000044
henrike@webrtc.org28e20752013-07-10 00:45:36 +000045class IceRestartAnswerLatch;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000046class JsepIceCandidate;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000047class MediaStreamSignaling;
wu@webrtc.org91053e72013-08-10 07:18:04 +000048class WebRtcSessionDescriptionFactory;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000049
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000050extern const char kBundleWithoutRtcpMux[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000051extern const char kCreateChannelFailed[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000052extern const char kInvalidCandidates[];
53extern const char kInvalidSdp[];
54extern const char kMlineMismatch[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000055extern const char kPushDownTDFailed[];
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +000056extern const char kSdpWithoutDtlsFingerprint[];
57extern const char kSdpWithoutSdesCrypto[];
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000058extern const char kSdpWithoutIceUfragPwd[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000059extern const char kSdpWithoutSdesAndDtlsDisabled[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000060extern const char kSessionError[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000061extern const char kSessionErrorDesc[];
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +000062extern const char kDtlsSetupFailureRtp[];
63extern const char kDtlsSetupFailureRtcp[];
deadbeefcbecd352015-09-23 11:50:27 -070064extern const char kEnableBundleFailed[];
65
buildbot@webrtc.org53df88c2014-08-07 22:46:01 +000066// Maximum number of received video streams that will be processed by webrtc
67// even if they are not signalled beforehand.
68extern const int kMaxUnsignalledRecvStreams;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000069
70// ICE state callback interface.
71class IceObserver {
72 public:
wu@webrtc.org364f2042013-11-20 21:49:41 +000073 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +000074 // Called any time the IceConnectionState changes
Peter Thatcher54360512015-07-08 11:08:35 -070075 // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to
76 // conform to the w3c standard.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000077 virtual void OnIceConnectionChange(
78 PeerConnectionInterface::IceConnectionState new_state) {}
79 // Called any time the IceGatheringState changes
80 virtual void OnIceGatheringChange(
81 PeerConnectionInterface::IceGatheringState new_state) {}
82 // New Ice candidate have been found.
83 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000084
Honghai Zhang7fb69db2016-03-14 11:59:18 -070085 // Some local ICE candidates have been removed.
86 virtual void OnIceCandidatesRemoved(
87 const std::vector<cricket::Candidate>& candidates) = 0;
88
Peter Thatcher54360512015-07-08 11:08:35 -070089 // Called whenever the state changes between receiving and not receiving.
90 virtual void OnIceConnectionReceivingChange(bool receiving) {}
91
henrike@webrtc.org28e20752013-07-10 00:45:36 +000092 protected:
93 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +000094
95 private:
henrikg3c089d72015-09-16 05:37:44 -070096 RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000097};
98
deadbeefd59daf82015-10-14 15:02:44 -070099// Statistics for all the transports of the session.
100typedef std::map<std::string, cricket::TransportStats> TransportStatsMap;
101typedef std::map<std::string, std::string> ProxyTransportMap;
102
103// TODO(pthatcher): Think of a better name for this. We already have
104// a TransportStats in transport.h. Perhaps TransportsStats?
105struct SessionStats {
106 ProxyTransportMap proxy_to_transport;
107 TransportStatsMap transport_stats;
108};
109
110// A WebRtcSession manages general session state. This includes negotiation
111// of both the application-level and network-level protocols: the former
112// defines what will be sent and the latter defines how it will be sent. Each
113// network-level protocol is represented by a Transport object. Each Transport
114// participates in the network-level negotiation. The individual streams of
115// packets are represented by TransportChannels. The application-level protocol
116// is represented by SessionDecription objects.
117class WebRtcSession : public AudioProviderInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000118 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000119 public DtmfProviderInterface,
deadbeefd59daf82015-10-14 15:02:44 -0700120 public DataChannelProviderInterface,
121 public sigslot::has_slots<> {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000122 public:
deadbeefd59daf82015-10-14 15:02:44 -0700123 enum State {
124 STATE_INIT = 0,
125 STATE_SENTOFFER, // Sent offer, waiting for answer.
126 STATE_RECEIVEDOFFER, // Received an offer. Need to send answer.
127 STATE_SENTPRANSWER, // Sent provisional answer. Need to send answer.
128 STATE_RECEIVEDPRANSWER, // Received provisional answer, waiting for answer.
129 STATE_INPROGRESS, // Offer/answer exchange completed.
130 STATE_CLOSED, // Close() was called.
131 };
132
133 enum Error {
134 ERROR_NONE = 0, // no error
135 ERROR_CONTENT = 1, // channel errors in SetLocalContent/SetRemoteContent
136 ERROR_TRANSPORT = 2, // transport error of some kind
137 };
138
stefanc1aeaf02015-10-15 07:26:07 -0700139 WebRtcSession(webrtc::MediaControllerInterface* media_controller,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000140 rtc::Thread* signaling_thread,
141 rtc::Thread* worker_thread,
deadbeefab9b2d12015-10-14 11:33:11 -0700142 cricket::PortAllocator* port_allocator);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000143 virtual ~WebRtcSession();
144
deadbeefd59daf82015-10-14 15:02:44 -0700145 // These are const to allow them to be called from const methods.
146 rtc::Thread* signaling_thread() const { return signaling_thread_; }
147 rtc::Thread* worker_thread() const { return worker_thread_; }
148 cricket::PortAllocator* port_allocator() const { return port_allocator_; }
149
150 // The ID of this session.
151 const std::string& id() const { return sid_; }
152
Henrik Lundin64dad832015-05-11 12:44:23 +0200153 bool Initialize(
154 const PeerConnectionFactoryInterface::Options& options,
Henrik Boström5e56c592015-08-11 10:33:13 +0200155 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
Henrik Lundin64dad832015-05-11 12:44:23 +0200156 const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000157 // Deletes the voice, video and data channel and changes the session state
deadbeefd59daf82015-10-14 15:02:44 -0700158 // to STATE_CLOSED.
159 void Close();
160
161 // Returns true if we were the initial offerer.
162 bool initial_offerer() const { return initial_offerer_; }
163
164 // Returns the current state of the session. See the enum above for details.
165 // Each time the state changes, we will fire this signal.
166 State state() const { return state_; }
167 sigslot::signal2<WebRtcSession*, State> SignalState;
168
169 // Returns the last error in the session. See the enum above for details.
170 Error error() const { return error_; }
171 const std::string& error_desc() const { return error_desc_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000172
173 void RegisterIceObserver(IceObserver* observer) {
174 ice_observer_ = observer;
175 }
176
177 virtual cricket::VoiceChannel* voice_channel() {
178 return voice_channel_.get();
179 }
180 virtual cricket::VideoChannel* video_channel() {
181 return video_channel_.get();
182 }
183 virtual cricket::DataChannel* data_channel() {
184 return data_channel_.get();
185 }
186
deadbeef0ed85b22016-02-23 17:24:52 -0800187 cricket::BaseChannel* GetChannel(const std::string& content_name);
188
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000189 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
190 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000191
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000192 // Get current ssl role from transport.
Taylor Brandstetterf475d362016-01-08 15:35:57 -0800193 bool GetSslRole(const std::string& transport_name, rtc::SSLRole* role);
194
195 // Get current SSL role for this channel's transport.
196 // If |transport| is null, returns false.
197 bool GetSslRole(const cricket::BaseChannel* channel, rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000198
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000199 void CreateOffer(
200 CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 11:33:11 -0700201 const PeerConnectionInterface::RTCOfferAnswerOptions& options,
202 const cricket::MediaSessionOptions& session_options);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000203 void CreateAnswer(CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 11:33:11 -0700204 const cricket::MediaSessionOptions& session_options);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000205 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000206 bool SetLocalDescription(SessionDescriptionInterface* desc,
207 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000208 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000209 bool SetRemoteDescription(SessionDescriptionInterface* desc,
210 std::string* err_desc);
211 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000212
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700213 bool RemoveRemoteIceCandidates(
214 const std::vector<cricket::Candidate>& candidates);
215
mallinath@webrtc.org3d81b1b2014-09-09 14:38:10 +0000216 bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000217
honghaiz1f429e32015-09-28 07:57:34 -0700218 cricket::IceConfig ParseIceConfig(
219 const PeerConnectionInterface::RTCConfiguration& config) const;
220
deadbeefd59daf82015-10-14 15:02:44 -0700221 void SetIceConfig(const cricket::IceConfig& ice_config);
222
223 // Start gathering candidates for any new transports, or transports doing an
224 // ICE restart.
225 void MaybeStartGathering();
226
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000227 const SessionDescriptionInterface* local_description() const {
228 return local_desc_.get();
229 }
230 const SessionDescriptionInterface* remote_description() const {
231 return remote_desc_.get();
232 }
233
234 // Get the id used as a media stream track's "id" field from ssrc.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200235 virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
236 virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000237
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000238 // AudioMediaProviderInterface implementation.
solenbergd4cec0d2015-10-09 08:55:48 -0700239 void SetAudioPlayout(uint32_t ssrc, bool enable) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200240 void SetAudioSend(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000241 bool enable,
242 const cricket::AudioOptions& options,
Taylor Brandstetter1a018dc2016-03-08 12:37:39 -0800243 cricket::AudioSource* source) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200244 void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
deadbeef2d110be2016-01-13 12:00:26 -0800245 void SetRawAudioSink(uint32_t ssrc,
246 rtc::scoped_ptr<AudioSinkInterface> sink) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000247
skvladdc1c62c2016-03-16 19:07:43 -0700248 RtpParameters GetAudioRtpParameters(uint32_t ssrc) const override;
249 bool SetAudioRtpParameters(uint32_t ssrc,
250 const RtpParameters& parameters) override;
251
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000252 // Implements VideoMediaProviderInterface.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200253 bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override;
nisse08582ff2016-02-04 01:24:52 -0800254 void SetVideoPlayout(
255 uint32_t ssrc,
256 bool enable,
257 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200258 void SetVideoSend(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000259 bool enable,
260 const cricket::VideoOptions* options) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000261
skvladdc1c62c2016-03-16 19:07:43 -0700262 RtpParameters GetVideoRtpParameters(uint32_t ssrc) const override;
263 bool SetVideoRtpParameters(uint32_t ssrc,
264 const RtpParameters& parameters) override;
265
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000266 // Implements DtmfProviderInterface.
267 virtual bool CanInsertDtmf(const std::string& track_id);
268 virtual bool InsertDtmf(const std::string& track_id,
269 int code, int duration);
270 virtual sigslot::signal0<>* GetOnDestroyedSignal();
271
wu@webrtc.org78187522013-10-07 23:32:02 +0000272 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000273 bool SendData(const cricket::SendDataParams& params,
jbauch944c3902016-03-19 01:57:31 -0700274 const rtc::CopyOnWriteBuffer& payload,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000275 cricket::SendDataResult* result) override;
276 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
277 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
278 void AddSctpDataStream(int sid) override;
279 void RemoveSctpDataStream(int sid) override;
280 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02 +0000281
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000282 // Returns stats for all channels of all transports.
283 // This avoids exposing the internal structures used to track them.
deadbeefd59daf82015-10-14 15:02:44 -0700284 virtual bool GetTransportStats(SessionStats* stats);
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000285
deadbeefcbecd352015-09-23 11:50:27 -0700286 // Get stats for a specific channel
deadbeefd59daf82015-10-14 15:02:44 -0700287 bool GetChannelTransportStats(cricket::BaseChannel* ch, SessionStats* stats);
deadbeefcbecd352015-09-23 11:50:27 -0700288
289 // virtual so it can be mocked in unit tests
290 virtual bool GetLocalCertificate(
291 const std::string& transport_name,
292 rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
293
294 // Caller owns returned certificate
295 virtual bool GetRemoteSSLCertificate(const std::string& transport_name,
296 rtc::SSLCertificate** cert);
297
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000298 cricket::DataChannelType data_channel_type() const;
299
deadbeef0ed85b22016-02-23 17:24:52 -0800300 bool IceRestartPending(const std::string& content_name) const;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000301
Henrik Boströmd8281982015-08-27 10:12:24 +0200302 // Called when an RTCCertificate is generated or retrieved by
wu@webrtc.org91053e72013-08-10 07:18:04 +0000303 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
Henrik Boströmd8281982015-08-27 10:12:24 +0200304 void OnCertificateReady(
305 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +0000306 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000307
308 // For unit test.
Henrik Boströmd8281982015-08-27 10:12:24 +0200309 bool waiting_for_certificate_for_testing() const;
deadbeefcbecd352015-09-23 11:50:27 -0700310 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000311
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000312 void set_metrics_observer(
313 webrtc::MetricsObserverInterface* metrics_observer) {
314 metrics_observer_ = metrics_observer;
315 }
316
deadbeefab9b2d12015-10-14 11:33:11 -0700317 // Called when voice_channel_, video_channel_ and data_channel_ are created
318 // and destroyed. As a result of, for example, setting a new description.
319 sigslot::signal0<> SignalVoiceChannelCreated;
320 sigslot::signal0<> SignalVoiceChannelDestroyed;
321 sigslot::signal0<> SignalVideoChannelCreated;
322 sigslot::signal0<> SignalVideoChannelDestroyed;
323 sigslot::signal0<> SignalDataChannelCreated;
324 sigslot::signal0<> SignalDataChannelDestroyed;
deadbeef057ecf02016-01-20 14:30:43 -0800325 // Called when the whole session is destroyed.
326 sigslot::signal0<> SignalDestroyed;
deadbeefab9b2d12015-10-14 11:33:11 -0700327
328 // Called when a valid data channel OPEN message is received.
329 // std::string represents the data channel label.
330 sigslot::signal2<const std::string&, const InternalDataChannelInit&>
331 SignalDataChannelOpenMessage;
332
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000333 private:
334 // Indicates the type of SessionDescription in a call to SetLocalDescription
335 // and SetRemoteDescription.
336 enum Action {
337 kOffer,
338 kPrAnswer,
339 kAnswer,
340 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000341
deadbeefd59daf82015-10-14 15:02:44 -0700342 // Log session state.
343 void LogState(State old_state, State new_state);
344
345 // Updates the state, signaling if necessary.
346 virtual void SetState(State state);
347
348 // Updates the error state, signaling if necessary.
349 // TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|.
350 virtual void SetError(Error error, const std::string& error_desc);
351
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000352 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000353 std::string* err_desc);
354 static Action GetAction(const std::string& type);
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000355 // Push the media parts of the local or remote session description
356 // down to all of the channels.
357 bool PushdownMediaDescription(cricket::ContentAction action,
358 cricket::ContentSource source,
359 std::string* error_desc);
360
deadbeefd59daf82015-10-14 15:02:44 -0700361 bool PushdownTransportDescription(cricket::ContentSource source,
362 cricket::ContentAction action,
363 std::string* error_desc);
364
365 // Helper methods to push local and remote transport descriptions.
366 bool PushdownLocalTransportDescription(
367 const cricket::SessionDescription* sdesc,
368 cricket::ContentAction action,
369 std::string* error_desc);
370 bool PushdownRemoteTransportDescription(
371 const cricket::SessionDescription* sdesc,
372 cricket::ContentAction action,
373 std::string* error_desc);
374
375 // Returns true and the TransportInfo of the given |content_name|
376 // from |description|. Returns false if it's not available.
377 static bool GetTransportDescription(
378 const cricket::SessionDescription* description,
379 const std::string& content_name,
380 cricket::TransportDescription* info);
381
deadbeefcbecd352015-09-23 11:50:27 -0700382 // 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,
jbauch944c3902016-03-19 01:57:31 -0700416 const rtc::CopyOnWriteBuffer& 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,
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700450 const std::vector<cricket::Candidate>& candidates);
451 void OnTransportControllerCandidatesRemoved(
452 const std::vector<cricket::Candidate>& candidates);
deadbeefcbecd352015-09-23 11:50:27 -0700453
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000454 std::string GetSessionErrorMsg();
455
deadbeefcbecd352015-09-23 11:50:27 -0700456 // Invoked when TransportController connection completion is signaled.
457 // Reports stats for all transports in use.
458 void ReportTransportStats();
459
460 // Gather the usage of IPv4/IPv6 as best connection.
jbauchac8869e2015-07-03 01:36:14 -0700461 void ReportBestConnectionState(const cricket::TransportStats& stats);
462
463 void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000464
stefanc1aeaf02015-10-15 07:26:07 -0700465 void OnSentPacket_w(cricket::TransportChannel* channel,
466 const rtc::SentPacket& sent_packet);
467
deadbeefd59daf82015-10-14 15:02:44 -0700468 rtc::Thread* const signaling_thread_;
469 rtc::Thread* const worker_thread_;
470 cricket::PortAllocator* const port_allocator_;
471
472 State state_ = STATE_INIT;
473 Error error_ = ERROR_NONE;
474 std::string error_desc_;
475
476 const std::string sid_;
477 bool initial_offerer_ = false;
478
479 rtc::scoped_ptr<cricket::TransportController> transport_controller_;
stefanc1aeaf02015-10-15 07:26:07 -0700480 MediaControllerInterface* media_controller_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000481 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
482 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
483 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000484 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000485 IceObserver* ice_observer_;
486 PeerConnectionInterface::IceConnectionState ice_connection_state_;
Peter Thatcher54360512015-07-08 11:08:35 -0700487 bool ice_connection_receiving_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000488 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
489 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000490 // If the remote peer is using a older version of implementation.
491 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000492 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000493 // Specifies which kind of data channel is allowed. This is controlled
494 // by the chrome command-line flag and constraints:
495 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
496 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
497 // not set or false, SCTP is allowed (DCT_SCTP);
498 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
499 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
500 cricket::DataChannelType data_channel_type_;
deadbeef0ed85b22016-02-23 17:24:52 -0800501 // List of content names for which the remote side triggered an ICE restart.
502 std::set<std::string> pending_ice_restarts_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000503
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000504 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04 +0000505 webrtc_session_desc_factory_;
506
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000507 // Member variables for caching global options.
508 cricket::AudioOptions audio_options_;
509 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000510 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000511
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000512 // Declares the bundle policy for the WebRTCSession.
513 PeerConnectionInterface::BundlePolicy bundle_policy_;
514
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700515 // Declares the RTCP mux policy for the WebRTCSession.
516 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
517
henrikg3c089d72015-09-16 05:37:44 -0700518 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
wu@webrtc.org364f2042013-11-20 21:49:41 +0000519};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000520} // namespace webrtc
521
Henrik Kjellander15583c12016-02-10 10:53:12 +0100522#endif // WEBRTC_API_WEBRTCSESSION_H_