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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 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
Honghai Zhang7fb69db2016-03-14 11:59:18 -070084 // Some local ICE candidates have been removed.
85 virtual void OnIceCandidatesRemoved(
86 const std::vector<cricket::Candidate>& candidates) = 0;
87
Peter Thatcher54360512015-07-08 11:08:35 -070088 // Called whenever the state changes between receiving and not receiving.
89 virtual void OnIceConnectionReceivingChange(bool receiving) {}
90
henrike@webrtc.org28e20752013-07-10 00:45:36 +000091 protected:
92 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +000093
94 private:
henrikg3c089d72015-09-16 05:37:44 -070095 RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000096};
97
deadbeefd59daf82015-10-14 15:02:44 -070098// Statistics for all the transports of the session.
99typedef std::map<std::string, cricket::TransportStats> TransportStatsMap;
100typedef std::map<std::string, std::string> ProxyTransportMap;
101
102// TODO(pthatcher): Think of a better name for this. We already have
103// a TransportStats in transport.h. Perhaps TransportsStats?
104struct SessionStats {
105 ProxyTransportMap proxy_to_transport;
106 TransportStatsMap transport_stats;
107};
108
109// A WebRtcSession manages general session state. This includes negotiation
110// of both the application-level and network-level protocols: the former
111// defines what will be sent and the latter defines how it will be sent. Each
112// network-level protocol is represented by a Transport object. Each Transport
113// participates in the network-level negotiation. The individual streams of
114// packets are represented by TransportChannels. The application-level protocol
115// is represented by SessionDecription objects.
116class WebRtcSession : public AudioProviderInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000117 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000118 public DtmfProviderInterface,
deadbeefd59daf82015-10-14 15:02:44 -0700119 public DataChannelProviderInterface,
120 public sigslot::has_slots<> {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000121 public:
deadbeefd59daf82015-10-14 15:02:44 -0700122 enum State {
123 STATE_INIT = 0,
124 STATE_SENTOFFER, // Sent offer, waiting for answer.
125 STATE_RECEIVEDOFFER, // Received an offer. Need to send answer.
126 STATE_SENTPRANSWER, // Sent provisional answer. Need to send answer.
127 STATE_RECEIVEDPRANSWER, // Received provisional answer, waiting for answer.
128 STATE_INPROGRESS, // Offer/answer exchange completed.
129 STATE_CLOSED, // Close() was called.
130 };
131
132 enum Error {
133 ERROR_NONE = 0, // no error
134 ERROR_CONTENT = 1, // channel errors in SetLocalContent/SetRemoteContent
135 ERROR_TRANSPORT = 2, // transport error of some kind
136 };
137
stefanc1aeaf02015-10-15 07:26:07 -0700138 WebRtcSession(webrtc::MediaControllerInterface* media_controller,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000139 rtc::Thread* signaling_thread,
140 rtc::Thread* worker_thread,
deadbeefab9b2d12015-10-14 11:33:11 -0700141 cricket::PortAllocator* port_allocator);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000142 virtual ~WebRtcSession();
143
deadbeefd59daf82015-10-14 15:02:44 -0700144 // These are const to allow them to be called from const methods.
145 rtc::Thread* signaling_thread() const { return signaling_thread_; }
146 rtc::Thread* worker_thread() const { return worker_thread_; }
147 cricket::PortAllocator* port_allocator() const { return port_allocator_; }
148
149 // The ID of this session.
150 const std::string& id() const { return sid_; }
151
Henrik Lundin64dad832015-05-11 12:44:23 +0200152 bool Initialize(
153 const PeerConnectionFactoryInterface::Options& options,
Henrik Boström5e56c592015-08-11 10:33:13 +0200154 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
Henrik Lundin64dad832015-05-11 12:44:23 +0200155 const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000156 // Deletes the voice, video and data channel and changes the session state
deadbeefd59daf82015-10-14 15:02:44 -0700157 // to STATE_CLOSED.
158 void Close();
159
160 // Returns true if we were the initial offerer.
161 bool initial_offerer() const { return initial_offerer_; }
162
163 // Returns the current state of the session. See the enum above for details.
164 // Each time the state changes, we will fire this signal.
165 State state() const { return state_; }
166 sigslot::signal2<WebRtcSession*, State> SignalState;
167
168 // Returns the last error in the session. See the enum above for details.
169 Error error() const { return error_; }
170 const std::string& error_desc() const { return error_desc_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000171
172 void RegisterIceObserver(IceObserver* observer) {
173 ice_observer_ = observer;
174 }
175
176 virtual cricket::VoiceChannel* voice_channel() {
177 return voice_channel_.get();
178 }
179 virtual cricket::VideoChannel* video_channel() {
180 return video_channel_.get();
181 }
182 virtual cricket::DataChannel* data_channel() {
183 return data_channel_.get();
184 }
185
deadbeef0ed85b22016-02-23 17:24:52 -0800186 cricket::BaseChannel* GetChannel(const std::string& content_name);
187
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000188 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
189 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000190
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000191 // Get current ssl role from transport.
Taylor Brandstetterf475d362016-01-08 15:35:57 -0800192 bool GetSslRole(const std::string& transport_name, rtc::SSLRole* role);
193
194 // Get current SSL role for this channel's transport.
195 // If |transport| is null, returns false.
196 bool GetSslRole(const cricket::BaseChannel* channel, rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000197
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000198 void CreateOffer(
199 CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 11:33:11 -0700200 const PeerConnectionInterface::RTCOfferAnswerOptions& options,
201 const cricket::MediaSessionOptions& session_options);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000202 void CreateAnswer(CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 11:33:11 -0700203 const cricket::MediaSessionOptions& session_options);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000204 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000205 bool SetLocalDescription(SessionDescriptionInterface* desc,
206 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000207 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000208 bool SetRemoteDescription(SessionDescriptionInterface* desc,
209 std::string* err_desc);
210 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000211
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700212 bool RemoveRemoteIceCandidates(
213 const std::vector<cricket::Candidate>& candidates);
214
mallinath@webrtc.org3d81b1b2014-09-09 14:38:10 +0000215 bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000216
honghaiz1f429e32015-09-28 07:57:34 -0700217 cricket::IceConfig ParseIceConfig(
218 const PeerConnectionInterface::RTCConfiguration& config) const;
219
deadbeefd59daf82015-10-14 15:02:44 -0700220 void SetIceConfig(const cricket::IceConfig& ice_config);
221
222 // Start gathering candidates for any new transports, or transports doing an
223 // ICE restart.
224 void MaybeStartGathering();
225
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000226 const SessionDescriptionInterface* local_description() const {
227 return local_desc_.get();
228 }
229 const SessionDescriptionInterface* remote_description() const {
230 return remote_desc_.get();
231 }
232
233 // Get the id used as a media stream track's "id" field from ssrc.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200234 virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
235 virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000236
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000237 // AudioMediaProviderInterface implementation.
solenbergd4cec0d2015-10-09 08:55:48 -0700238 void SetAudioPlayout(uint32_t ssrc, bool enable) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200239 void SetAudioSend(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000240 bool enable,
241 const cricket::AudioOptions& options,
Taylor Brandstetter1a018dc2016-03-08 12:37:39 -0800242 cricket::AudioSource* source) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200243 void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
deadbeef2d110be2016-01-13 12:00:26 -0800244 void SetRawAudioSink(uint32_t ssrc,
245 rtc::scoped_ptr<AudioSinkInterface> sink) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000246
skvladdc1c62c2016-03-16 19:07:43 -0700247 RtpParameters GetAudioRtpParameters(uint32_t ssrc) const override;
248 bool SetAudioRtpParameters(uint32_t ssrc,
249 const RtpParameters& parameters) override;
250
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000251 // Implements VideoMediaProviderInterface.
nisse2ded9b12016-04-08 02:23:55 -0700252 bool SetSource(
253 uint32_t ssrc,
254 rtc::VideoSourceInterface<cricket::VideoFrame>* source) override;
nisse08582ff2016-02-04 01:24:52 -0800255 void SetVideoPlayout(
256 uint32_t ssrc,
257 bool enable,
258 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) 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
skvladdc1c62c2016-03-16 19:07:43 -0700263 RtpParameters GetVideoRtpParameters(uint32_t ssrc) const override;
264 bool SetVideoRtpParameters(uint32_t ssrc,
265 const RtpParameters& parameters) override;
266
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000267 // Implements DtmfProviderInterface.
268 virtual bool CanInsertDtmf(const std::string& track_id);
269 virtual bool InsertDtmf(const std::string& track_id,
270 int code, int duration);
271 virtual sigslot::signal0<>* GetOnDestroyedSignal();
272
wu@webrtc.org78187522013-10-07 23:32:02 +0000273 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000274 bool SendData(const cricket::SendDataParams& params,
jbaucheec21bd2016-03-20 06:15:43 -0700275 const rtc::CopyOnWriteBuffer& payload,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000276 cricket::SendDataResult* result) override;
277 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
278 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
279 void AddSctpDataStream(int sid) override;
280 void RemoveSctpDataStream(int sid) override;
281 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02 +0000282
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000283 // Returns stats for all channels of all transports.
284 // This avoids exposing the internal structures used to track them.
deadbeefd59daf82015-10-14 15:02:44 -0700285 virtual bool GetTransportStats(SessionStats* stats);
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000286
deadbeefcbecd352015-09-23 11:50:27 -0700287 // Get stats for a specific channel
deadbeefd59daf82015-10-14 15:02:44 -0700288 bool GetChannelTransportStats(cricket::BaseChannel* ch, SessionStats* stats);
deadbeefcbecd352015-09-23 11:50:27 -0700289
290 // virtual so it can be mocked in unit tests
291 virtual bool GetLocalCertificate(
292 const std::string& transport_name,
293 rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
294
295 // Caller owns returned certificate
kwibergb4d01c42016-04-06 05:15:06 -0700296 virtual rtc::scoped_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
297 const std::string& transport_name);
deadbeefcbecd352015-09-23 11:50:27 -0700298
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000299 cricket::DataChannelType data_channel_type() const;
300
deadbeef0ed85b22016-02-23 17:24:52 -0800301 bool IceRestartPending(const std::string& content_name) const;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000302
Henrik Boströmd8281982015-08-27 10:12:24 +0200303 // Called when an RTCCertificate is generated or retrieved by
wu@webrtc.org91053e72013-08-10 07:18:04 +0000304 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
Henrik Boströmd8281982015-08-27 10:12:24 +0200305 void OnCertificateReady(
306 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +0000307 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000308
309 // For unit test.
Henrik Boströmd8281982015-08-27 10:12:24 +0200310 bool waiting_for_certificate_for_testing() const;
deadbeefcbecd352015-09-23 11:50:27 -0700311 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000312
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000313 void set_metrics_observer(
314 webrtc::MetricsObserverInterface* metrics_observer) {
315 metrics_observer_ = metrics_observer;
316 }
317
deadbeefab9b2d12015-10-14 11:33:11 -0700318 // Called when voice_channel_, video_channel_ and data_channel_ are created
319 // and destroyed. As a result of, for example, setting a new description.
320 sigslot::signal0<> SignalVoiceChannelCreated;
321 sigslot::signal0<> SignalVoiceChannelDestroyed;
322 sigslot::signal0<> SignalVideoChannelCreated;
323 sigslot::signal0<> SignalVideoChannelDestroyed;
324 sigslot::signal0<> SignalDataChannelCreated;
325 sigslot::signal0<> SignalDataChannelDestroyed;
deadbeef057ecf02016-01-20 14:30:43 -0800326 // Called when the whole session is destroyed.
327 sigslot::signal0<> SignalDestroyed;
deadbeefab9b2d12015-10-14 11:33:11 -0700328
329 // Called when a valid data channel OPEN message is received.
330 // std::string represents the data channel label.
331 sigslot::signal2<const std::string&, const InternalDataChannelInit&>
332 SignalDataChannelOpenMessage;
333
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000334 private:
335 // Indicates the type of SessionDescription in a call to SetLocalDescription
336 // and SetRemoteDescription.
337 enum Action {
338 kOffer,
339 kPrAnswer,
340 kAnswer,
341 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000342
deadbeefd59daf82015-10-14 15:02:44 -0700343 // Log session state.
344 void LogState(State old_state, State new_state);
345
346 // Updates the state, signaling if necessary.
347 virtual void SetState(State state);
348
349 // Updates the error state, signaling if necessary.
350 // TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|.
351 virtual void SetError(Error error, const std::string& error_desc);
352
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000353 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000354 std::string* err_desc);
355 static Action GetAction(const std::string& type);
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000356 // Push the media parts of the local or remote session description
357 // down to all of the channels.
358 bool PushdownMediaDescription(cricket::ContentAction action,
359 cricket::ContentSource source,
360 std::string* error_desc);
361
deadbeefd59daf82015-10-14 15:02:44 -0700362 bool PushdownTransportDescription(cricket::ContentSource source,
363 cricket::ContentAction action,
364 std::string* error_desc);
365
366 // Helper methods to push local and remote transport descriptions.
367 bool PushdownLocalTransportDescription(
368 const cricket::SessionDescription* sdesc,
369 cricket::ContentAction action,
370 std::string* error_desc);
371 bool PushdownRemoteTransportDescription(
372 const cricket::SessionDescription* sdesc,
373 cricket::ContentAction action,
374 std::string* error_desc);
375
376 // Returns true and the TransportInfo of the given |content_name|
377 // from |description|. Returns false if it's not available.
378 static bool GetTransportDescription(
379 const cricket::SessionDescription* description,
380 const std::string& content_name,
381 cricket::TransportDescription* info);
382
deadbeefcbecd352015-09-23 11:50:27 -0700383 // Cause all the BaseChannels in the bundle group to have the same
384 // transport channel.
385 bool EnableBundle(const cricket::ContentGroup& bundle);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000386
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000387 // Enables media channels to allow sending of media.
388 void EnableChannels();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000389 // Returns the media index for a local ice candidate given the content name.
390 // Returns false if the local session description does not have a media
391 // content called |content_name|.
392 bool GetLocalCandidateMediaIndex(const std::string& content_name,
393 int* sdp_mline_index);
394 // Uses all remote candidates in |remote_desc| in this session.
395 bool UseCandidatesInSessionDescription(
396 const SessionDescriptionInterface* remote_desc);
397 // Uses |candidate| in this session.
398 bool UseCandidate(const IceCandidateInterface* candidate);
399 // Deletes the corresponding channel of contents that don't exist in |desc|.
400 // |desc| can be null. This means that all channels are deleted.
deadbeefcbecd352015-09-23 11:50:27 -0700401 void RemoveUnusedChannels(const cricket::SessionDescription* desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000402
403 // Allocates media channels based on the |desc|. If |desc| doesn't have
404 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
405 // This method will also delete any existing media channels before creating.
406 bool CreateChannels(const cricket::SessionDescription* desc);
407
408 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000409 bool CreateVoiceChannel(const cricket::ContentInfo* content);
410 bool CreateVideoChannel(const cricket::ContentInfo* content);
411 bool CreateDataChannel(const cricket::ContentInfo* content);
412
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000413 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
414 // messages.
415 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
416 const cricket::ReceiveDataParams& params,
jbaucheec21bd2016-03-20 06:15:43 -0700417 const rtc::CopyOnWriteBuffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000418
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000419 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000420 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
Peter Thatcher54360512015-07-08 11:08:35 -0700421 void SetIceConnectionReceiving(bool receiving);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000422
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000423 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000424 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000425 // Below methods are helper methods which verifies SDP.
426 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
427 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000428 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000429
430 // Check if a call to SetLocalDescription is acceptable with |action|.
431 bool ExpectSetLocalDescription(Action action);
432 // Check if a call to SetRemoteDescription is acceptable with |action|.
433 bool ExpectSetRemoteDescription(Action action);
434 // Verifies a=setup attribute as per RFC 5763.
435 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
436 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000437
jiayl@webrtc.orge10d28c2014-07-17 17:07:49 +0000438 // Returns true if we are ready to push down the remote candidate.
439 // |remote_desc| is the new remote description, or NULL if the current remote
440 // description should be used. Output |valid| is true if the candidate media
441 // index is valid.
442 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
443 const SessionDescriptionInterface* remote_desc,
444 bool* valid);
445
deadbeefcbecd352015-09-23 11:50:27 -0700446 void OnTransportControllerConnectionState(cricket::IceConnectionState state);
447 void OnTransportControllerReceiving(bool receiving);
448 void OnTransportControllerGatheringState(cricket::IceGatheringState state);
449 void OnTransportControllerCandidatesGathered(
450 const std::string& transport_name,
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700451 const std::vector<cricket::Candidate>& candidates);
452 void OnTransportControllerCandidatesRemoved(
453 const std::vector<cricket::Candidate>& candidates);
deadbeefcbecd352015-09-23 11:50:27 -0700454
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000455 std::string GetSessionErrorMsg();
456
deadbeefcbecd352015-09-23 11:50:27 -0700457 // Invoked when TransportController connection completion is signaled.
458 // Reports stats for all transports in use.
459 void ReportTransportStats();
460
461 // Gather the usage of IPv4/IPv6 as best connection.
jbauchac8869e2015-07-03 01:36:14 -0700462 void ReportBestConnectionState(const cricket::TransportStats& stats);
463
464 void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000465
stefanc1aeaf02015-10-15 07:26:07 -0700466 void OnSentPacket_w(cricket::TransportChannel* channel,
467 const rtc::SentPacket& sent_packet);
468
deadbeefd59daf82015-10-14 15:02:44 -0700469 rtc::Thread* const signaling_thread_;
470 rtc::Thread* const worker_thread_;
471 cricket::PortAllocator* const port_allocator_;
472
473 State state_ = STATE_INIT;
474 Error error_ = ERROR_NONE;
475 std::string error_desc_;
476
477 const std::string sid_;
478 bool initial_offerer_ = false;
479
480 rtc::scoped_ptr<cricket::TransportController> transport_controller_;
stefanc1aeaf02015-10-15 07:26:07 -0700481 MediaControllerInterface* media_controller_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000482 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
483 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
484 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000485 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000486 IceObserver* ice_observer_;
487 PeerConnectionInterface::IceConnectionState ice_connection_state_;
Peter Thatcher54360512015-07-08 11:08:35 -0700488 bool ice_connection_receiving_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000489 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
490 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000491 // If the remote peer is using a older version of implementation.
492 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000493 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000494 // Specifies which kind of data channel is allowed. This is controlled
495 // by the chrome command-line flag and constraints:
496 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
497 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
498 // not set or false, SCTP is allowed (DCT_SCTP);
499 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
500 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
501 cricket::DataChannelType data_channel_type_;
deadbeef0ed85b22016-02-23 17:24:52 -0800502 // List of content names for which the remote side triggered an ICE restart.
503 std::set<std::string> pending_ice_restarts_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000504
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000505 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04 +0000506 webrtc_session_desc_factory_;
507
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000508 // Member variables for caching global options.
509 cricket::AudioOptions audio_options_;
510 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000511 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000512
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000513 // Declares the bundle policy for the WebRTCSession.
514 PeerConnectionInterface::BundlePolicy bundle_policy_;
515
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700516 // Declares the RTCP mux policy for the WebRTCSession.
517 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
518
henrikg3c089d72015-09-16 05:37:44 -0700519 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
wu@webrtc.org364f2042013-11-20 21:49:41 +0000520};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000521} // namespace webrtc
522
Henrik Kjellander15583c12016-02-10 10:53:12 +0100523#endif // WEBRTC_API_WEBRTCSESSION_H_