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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
jbauch555604a2016-04-26 03:13:22 -070014#include <memory>
deadbeef0ed85b22016-02-23 17:24:52 -080015#include <set>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000016#include <string>
deadbeefcbecd352015-09-23 11:50:27 -070017#include <vector>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000018
Henrik Kjellander15583c12016-02-10 10:53:12 +010019#include "webrtc/api/datachannel.h"
20#include "webrtc/api/dtmfsender.h"
21#include "webrtc/api/mediacontroller.h"
Henrik Kjellander15583c12016-02-10 10:53:12 +010022#include "webrtc/api/peerconnectioninterface.h"
23#include "webrtc/api/statstypes.h"
kwiberg4485ffb2016-04-26 08:14:39 -070024#include "webrtc/base/constructormagic.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000025#include "webrtc/base/sigslot.h"
Henrik Boström5e56c592015-08-11 10:33:13 +020026#include "webrtc/base/sslidentity.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000027#include "webrtc/base/thread.h"
kjellandera96e2d72016-02-04 23:52:28 -080028#include "webrtc/media/base/mediachannel.h"
Honghai Zhang7fb69db2016-03-14 11:59:18 -070029#include "webrtc/p2p/base/candidate.h"
Tommif888bb52015-12-12 01:37:01 +010030#include "webrtc/p2p/base/transportcontroller.h"
kjellander@webrtc.org9b8df252016-02-12 06:47:59 +010031#include "webrtc/pc/mediasession.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000032
zhihuang9763d562016-08-05 11:14:50 -070033#ifdef HAVE_QUIC
34#include "webrtc/api/quicdatatransport.h"
35#endif // HAVE_QUIC
36
henrike@webrtc.org28e20752013-07-10 00:45:36 +000037namespace cricket {
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000038
henrike@webrtc.org28e20752013-07-10 00:45:36 +000039class ChannelManager;
40class DataChannel;
41class StatsReport;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000042class VideoChannel;
43class VoiceChannel;
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000044
zhihuang9763d562016-08-05 11:14:50 -070045#ifdef HAVE_QUIC
46class QuicTransportChannel;
47#endif // HAVE_QUIC
48
henrike@webrtc.org28e20752013-07-10 00:45:36 +000049} // namespace cricket
50
51namespace webrtc {
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000052
henrike@webrtc.org28e20752013-07-10 00:45:36 +000053class IceRestartAnswerLatch;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000054class JsepIceCandidate;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000055class MediaStreamSignaling;
wu@webrtc.org91053e72013-08-10 07:18:04 +000056class WebRtcSessionDescriptionFactory;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000057
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000058extern const char kBundleWithoutRtcpMux[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000059extern const char kCreateChannelFailed[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000060extern const char kInvalidCandidates[];
61extern const char kInvalidSdp[];
62extern const char kMlineMismatch[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000063extern const char kPushDownTDFailed[];
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +000064extern const char kSdpWithoutDtlsFingerprint[];
65extern const char kSdpWithoutSdesCrypto[];
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000066extern const char kSdpWithoutIceUfragPwd[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000067extern const char kSdpWithoutSdesAndDtlsDisabled[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000068extern const char kSessionError[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000069extern const char kSessionErrorDesc[];
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +000070extern const char kDtlsSetupFailureRtp[];
71extern const char kDtlsSetupFailureRtcp[];
deadbeefcbecd352015-09-23 11:50:27 -070072extern const char kEnableBundleFailed[];
73
buildbot@webrtc.org53df88c2014-08-07 22:46:01 +000074// Maximum number of received video streams that will be processed by webrtc
75// even if they are not signalled beforehand.
76extern const int kMaxUnsignalledRecvStreams;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000077
78// ICE state callback interface.
79class IceObserver {
80 public:
wu@webrtc.org364f2042013-11-20 21:49:41 +000081 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +000082 // Called any time the IceConnectionState changes
Peter Thatcher54360512015-07-08 11:08:35 -070083 // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to
84 // conform to the w3c standard.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000085 virtual void OnIceConnectionChange(
86 PeerConnectionInterface::IceConnectionState new_state) {}
87 // Called any time the IceGatheringState changes
88 virtual void OnIceGatheringChange(
89 PeerConnectionInterface::IceGatheringState new_state) {}
90 // New Ice candidate have been found.
91 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000092
Honghai Zhang7fb69db2016-03-14 11:59:18 -070093 // Some local ICE candidates have been removed.
94 virtual void OnIceCandidatesRemoved(
95 const std::vector<cricket::Candidate>& candidates) = 0;
96
Peter Thatcher54360512015-07-08 11:08:35 -070097 // Called whenever the state changes between receiving and not receiving.
98 virtual void OnIceConnectionReceivingChange(bool receiving) {}
99
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000100 protected:
101 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +0000102
103 private:
henrikg3c089d72015-09-16 05:37:44 -0700104 RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000105};
106
deadbeefd59daf82015-10-14 15:02:44 -0700107// Statistics for all the transports of the session.
108typedef std::map<std::string, cricket::TransportStats> TransportStatsMap;
109typedef std::map<std::string, std::string> ProxyTransportMap;
110
111// TODO(pthatcher): Think of a better name for this. We already have
112// a TransportStats in transport.h. Perhaps TransportsStats?
113struct SessionStats {
114 ProxyTransportMap proxy_to_transport;
115 TransportStatsMap transport_stats;
116};
117
hbosdf6075a2016-12-19 04:58:02 -0800118struct ChannelNamePair {
119 ChannelNamePair(
120 const std::string& content_name, const std::string& transport_name)
121 : content_name(content_name), transport_name(transport_name) {}
122 std::string content_name;
123 std::string transport_name;
124};
125
126struct ChannelNamePairs {
127 rtc::Optional<ChannelNamePair> voice;
128 rtc::Optional<ChannelNamePair> video;
129 rtc::Optional<ChannelNamePair> data;
130};
131
deadbeefd59daf82015-10-14 15:02:44 -0700132// A WebRtcSession manages general session state. This includes negotiation
133// of both the application-level and network-level protocols: the former
134// defines what will be sent and the latter defines how it will be sent. Each
135// network-level protocol is represented by a Transport object. Each Transport
136// participates in the network-level negotiation. The individual streams of
137// packets are represented by TransportChannels. The application-level protocol
138// is represented by SessionDecription objects.
Taylor Brandstetterba29c6a2016-06-27 16:30:35 -0700139class WebRtcSession :
140
141 public DtmfProviderInterface,
142 public DataChannelProviderInterface,
143 public sigslot::has_slots<> {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000144 public:
deadbeefd59daf82015-10-14 15:02:44 -0700145 enum State {
146 STATE_INIT = 0,
147 STATE_SENTOFFER, // Sent offer, waiting for answer.
148 STATE_RECEIVEDOFFER, // Received an offer. Need to send answer.
149 STATE_SENTPRANSWER, // Sent provisional answer. Need to send answer.
150 STATE_RECEIVEDPRANSWER, // Received provisional answer, waiting for answer.
151 STATE_INPROGRESS, // Offer/answer exchange completed.
152 STATE_CLOSED, // Close() was called.
153 };
154
155 enum Error {
156 ERROR_NONE = 0, // no error
157 ERROR_CONTENT = 1, // channel errors in SetLocalContent/SetRemoteContent
158 ERROR_TRANSPORT = 2, // transport error of some kind
159 };
160
zhihuang29ff8442016-07-27 11:07:25 -0700161 WebRtcSession(
162 webrtc::MediaControllerInterface* media_controller,
163 rtc::Thread* network_thread,
164 rtc::Thread* worker_thread,
165 rtc::Thread* signaling_thread,
166 cricket::PortAllocator* port_allocator,
167 std::unique_ptr<cricket::TransportController> transport_controller);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000168 virtual ~WebRtcSession();
169
deadbeefd59daf82015-10-14 15:02:44 -0700170 // These are const to allow them to be called from const methods.
zhihuang9763d562016-08-05 11:14:50 -0700171 rtc::Thread* network_thread() const { return network_thread_; }
deadbeefd59daf82015-10-14 15:02:44 -0700172 rtc::Thread* worker_thread() const { return worker_thread_; }
danilchape9021a32016-05-17 01:52:02 -0700173 rtc::Thread* signaling_thread() const { return signaling_thread_; }
deadbeefd59daf82015-10-14 15:02:44 -0700174
175 // The ID of this session.
176 const std::string& id() const { return sid_; }
177
Henrik Lundin64dad832015-05-11 12:44:23 +0200178 bool Initialize(
179 const PeerConnectionFactoryInterface::Options& options,
Henrik Boströmd03c23b2016-06-01 11:44:18 +0200180 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
Henrik Lundin64dad832015-05-11 12:44:23 +0200181 const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000182 // Deletes the voice, video and data channel and changes the session state
deadbeefd59daf82015-10-14 15:02:44 -0700183 // to STATE_CLOSED.
184 void Close();
185
186 // Returns true if we were the initial offerer.
187 bool initial_offerer() const { return initial_offerer_; }
188
189 // Returns the current state of the session. See the enum above for details.
190 // Each time the state changes, we will fire this signal.
191 State state() const { return state_; }
192 sigslot::signal2<WebRtcSession*, State> SignalState;
193
194 // Returns the last error in the session. See the enum above for details.
195 Error error() const { return error_; }
196 const std::string& error_desc() const { return error_desc_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000197
198 void RegisterIceObserver(IceObserver* observer) {
199 ice_observer_ = observer;
200 }
201
202 virtual cricket::VoiceChannel* voice_channel() {
203 return voice_channel_.get();
204 }
205 virtual cricket::VideoChannel* video_channel() {
206 return video_channel_.get();
207 }
208 virtual cricket::DataChannel* data_channel() {
209 return data_channel_.get();
210 }
hbosdf6075a2016-12-19 04:58:02 -0800211 std::unique_ptr<ChannelNamePairs> GetChannelNamePairs();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000212
deadbeef0ed85b22016-02-23 17:24:52 -0800213 cricket::BaseChannel* GetChannel(const std::string& content_name);
214
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000215 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000216
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000217 // Get current ssl role from transport.
Taylor Brandstetterf475d362016-01-08 15:35:57 -0800218 bool GetSslRole(const std::string& transport_name, rtc::SSLRole* role);
219
220 // Get current SSL role for this channel's transport.
221 // If |transport| is null, returns false.
222 bool GetSslRole(const cricket::BaseChannel* channel, rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000223
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000224 void CreateOffer(
225 CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 11:33:11 -0700226 const PeerConnectionInterface::RTCOfferAnswerOptions& options,
227 const cricket::MediaSessionOptions& session_options);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000228 void CreateAnswer(CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 11:33:11 -0700229 const cricket::MediaSessionOptions& session_options);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000230 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000231 bool SetLocalDescription(SessionDescriptionInterface* desc,
232 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000233 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000234 bool SetRemoteDescription(SessionDescriptionInterface* desc,
235 std::string* err_desc);
236 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000237
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700238 bool RemoveRemoteIceCandidates(
239 const std::vector<cricket::Candidate>& candidates);
240
honghaiz1f429e32015-09-28 07:57:34 -0700241 cricket::IceConfig ParseIceConfig(
242 const PeerConnectionInterface::RTCConfiguration& config) const;
243
deadbeefd59daf82015-10-14 15:02:44 -0700244 void SetIceConfig(const cricket::IceConfig& ice_config);
245
246 // Start gathering candidates for any new transports, or transports doing an
247 // ICE restart.
248 void MaybeStartGathering();
249
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000250 const SessionDescriptionInterface* local_description() const {
251 return local_desc_.get();
252 }
253 const SessionDescriptionInterface* remote_description() const {
254 return remote_desc_.get();
255 }
256
257 // Get the id used as a media stream track's "id" field from ssrc.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200258 virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
259 virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000260
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000261 // Implements DtmfProviderInterface.
nisseef8b61e2016-04-29 06:09:15 -0700262 bool CanInsertDtmf(const std::string& track_id) override;
263 bool InsertDtmf(const std::string& track_id,
264 int code, int duration) override;
265 sigslot::signal0<>* GetOnDestroyedSignal() override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000266
wu@webrtc.org78187522013-10-07 23:32:02 +0000267 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000268 bool SendData(const cricket::SendDataParams& params,
jbaucheec21bd2016-03-20 06:15:43 -0700269 const rtc::CopyOnWriteBuffer& payload,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000270 cricket::SendDataResult* result) override;
271 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
272 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
273 void AddSctpDataStream(int sid) override;
274 void RemoveSctpDataStream(int sid) override;
275 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02 +0000276
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000277 // Returns stats for all channels of all transports.
278 // This avoids exposing the internal structures used to track them.
hbosdf6075a2016-12-19 04:58:02 -0800279 // The parameterless version creates |ChannelNamePairs| from |voice_channel|,
280 // |video_channel| and |voice_channel| if available - this requires it to be
281 // called on the signaling thread - and invokes the other |GetStats|. The
282 // other |GetStats| can be invoked on any thread; if not invoked on the
283 // network thread a thread hop will happen.
284 std::unique_ptr<SessionStats> GetStats_s();
285 virtual std::unique_ptr<SessionStats> GetStats(
286 const ChannelNamePairs& channel_name_pairs);
deadbeefcbecd352015-09-23 11:50:27 -0700287
288 // virtual so it can be mocked in unit tests
289 virtual bool GetLocalCertificate(
290 const std::string& transport_name,
291 rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
292
293 // Caller owns returned certificate
jbauch555604a2016-04-26 03:13:22 -0700294 virtual std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
kwibergb4d01c42016-04-06 05:15:06 -0700295 const std::string& transport_name);
deadbeefcbecd352015-09-23 11:50:27 -0700296
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000297 cricket::DataChannelType data_channel_type() const;
298
deadbeefd1a38b52016-12-10 13:15:33 -0800299 // Returns true if there was an ICE restart initiated by the remote offer.
deadbeef0ed85b22016-02-23 17:24:52 -0800300 bool IceRestartPending(const std::string& content_name) const;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000301
deadbeefd1a38b52016-12-10 13:15:33 -0800302 // Set the "needs-ice-restart" flag as described in JSEP. After the flag is
303 // set, offers should generate new ufrags/passwords until an ICE restart
304 // occurs.
305 void SetNeedsIceRestartFlag();
306 // Returns true if the ICE restart flag above was set, and no ICE restart has
307 // occurred yet for this transport (by applying a local description with
308 // changed ufrag/password). If the transport has been deleted as a result of
309 // bundling, returns false.
310 bool NeedsIceRestart(const std::string& content_name) const;
311
Henrik Boströmd8281982015-08-27 10:12:24 +0200312 // Called when an RTCCertificate is generated or retrieved by
wu@webrtc.org91053e72013-08-10 07:18:04 +0000313 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
Henrik Boströmd8281982015-08-27 10:12:24 +0200314 void OnCertificateReady(
315 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +0000316 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000317
318 // For unit test.
Henrik Boströmd8281982015-08-27 10:12:24 +0200319 bool waiting_for_certificate_for_testing() const;
deadbeefcbecd352015-09-23 11:50:27 -0700320 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000321
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000322 void set_metrics_observer(
323 webrtc::MetricsObserverInterface* metrics_observer) {
324 metrics_observer_ = metrics_observer;
Honghai Zhangd93f50c2016-10-05 11:47:22 -0700325 transport_controller_->SetMetricsObserver(metrics_observer);
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000326 }
327
deadbeefab9b2d12015-10-14 11:33:11 -0700328 // Called when voice_channel_, video_channel_ and data_channel_ are created
329 // and destroyed. As a result of, for example, setting a new description.
330 sigslot::signal0<> SignalVoiceChannelCreated;
331 sigslot::signal0<> SignalVoiceChannelDestroyed;
332 sigslot::signal0<> SignalVideoChannelCreated;
333 sigslot::signal0<> SignalVideoChannelDestroyed;
334 sigslot::signal0<> SignalDataChannelCreated;
335 sigslot::signal0<> SignalDataChannelDestroyed;
deadbeef057ecf02016-01-20 14:30:43 -0800336 // Called when the whole session is destroyed.
337 sigslot::signal0<> SignalDestroyed;
deadbeefab9b2d12015-10-14 11:33:11 -0700338
339 // Called when a valid data channel OPEN message is received.
340 // std::string represents the data channel label.
341 sigslot::signal2<const std::string&, const InternalDataChannelInit&>
342 SignalDataChannelOpenMessage;
zhihuang9763d562016-08-05 11:14:50 -0700343#ifdef HAVE_QUIC
344 QuicDataTransport* quic_data_transport() {
345 return quic_data_transport_.get();
346 }
347#endif // HAVE_QUIC
deadbeefab9b2d12015-10-14 11:33:11 -0700348
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000349 private:
350 // Indicates the type of SessionDescription in a call to SetLocalDescription
351 // and SetRemoteDescription.
352 enum Action {
353 kOffer,
354 kPrAnswer,
355 kAnswer,
356 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000357
deadbeefd59daf82015-10-14 15:02:44 -0700358 // Log session state.
359 void LogState(State old_state, State new_state);
360
361 // Updates the state, signaling if necessary.
362 virtual void SetState(State state);
363
364 // Updates the error state, signaling if necessary.
365 // TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|.
366 virtual void SetError(Error error, const std::string& error_desc);
367
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000368 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000369 std::string* err_desc);
370 static Action GetAction(const std::string& type);
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000371 // Push the media parts of the local or remote session description
372 // down to all of the channels.
373 bool PushdownMediaDescription(cricket::ContentAction action,
374 cricket::ContentSource source,
375 std::string* error_desc);
376
deadbeefd59daf82015-10-14 15:02:44 -0700377 bool PushdownTransportDescription(cricket::ContentSource source,
378 cricket::ContentAction action,
379 std::string* error_desc);
380
381 // Helper methods to push local and remote transport descriptions.
382 bool PushdownLocalTransportDescription(
383 const cricket::SessionDescription* sdesc,
384 cricket::ContentAction action,
385 std::string* error_desc);
386 bool PushdownRemoteTransportDescription(
387 const cricket::SessionDescription* sdesc,
388 cricket::ContentAction action,
389 std::string* error_desc);
390
391 // Returns true and the TransportInfo of the given |content_name|
392 // from |description|. Returns false if it's not available.
393 static bool GetTransportDescription(
394 const cricket::SessionDescription* description,
395 const std::string& content_name,
396 cricket::TransportDescription* info);
397
skvlad6c87a672016-05-17 17:49:52 -0700398 // Returns the name of the transport channel when BUNDLE is enabled, or
399 // nullptr if the channel is not part of any bundle.
400 const std::string* GetBundleTransportName(
401 const cricket::ContentInfo* content,
402 const cricket::ContentGroup* bundle);
403
deadbeefcbecd352015-09-23 11:50:27 -0700404 // Cause all the BaseChannels in the bundle group to have the same
405 // transport channel.
406 bool EnableBundle(const cricket::ContentGroup& bundle);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000407
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000408 // Enables media channels to allow sending of media.
409 void EnableChannels();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000410 // Returns the media index for a local ice candidate given the content name.
411 // Returns false if the local session description does not have a media
412 // content called |content_name|.
413 bool GetLocalCandidateMediaIndex(const std::string& content_name,
414 int* sdp_mline_index);
415 // Uses all remote candidates in |remote_desc| in this session.
416 bool UseCandidatesInSessionDescription(
417 const SessionDescriptionInterface* remote_desc);
418 // Uses |candidate| in this session.
419 bool UseCandidate(const IceCandidateInterface* candidate);
420 // Deletes the corresponding channel of contents that don't exist in |desc|.
421 // |desc| can be null. This means that all channels are deleted.
deadbeefcbecd352015-09-23 11:50:27 -0700422 void RemoveUnusedChannels(const cricket::SessionDescription* desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000423
424 // Allocates media channels based on the |desc|. If |desc| doesn't have
425 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
426 // This method will also delete any existing media channels before creating.
427 bool CreateChannels(const cricket::SessionDescription* desc);
428
429 // Helper methods to create media channels.
skvlad6c87a672016-05-17 17:49:52 -0700430 bool CreateVoiceChannel(const cricket::ContentInfo* content,
431 const std::string* bundle_transport);
432 bool CreateVideoChannel(const cricket::ContentInfo* content,
433 const std::string* bundle_transport);
434 bool CreateDataChannel(const cricket::ContentInfo* content,
435 const std::string* bundle_transport);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000436
hbosdf6075a2016-12-19 04:58:02 -0800437 std::unique_ptr<SessionStats> GetStats_n(
438 const ChannelNamePairs& channel_name_pairs);
439
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000440 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
441 // messages.
442 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
443 const cricket::ReceiveDataParams& params,
jbaucheec21bd2016-03-20 06:15:43 -0700444 const rtc::CopyOnWriteBuffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000445
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000446 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000447 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
Peter Thatcher54360512015-07-08 11:08:35 -0700448 void SetIceConnectionReceiving(bool receiving);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000449
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000450 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000451 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000452 // Below methods are helper methods which verifies SDP.
453 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
454 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000455 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000456
457 // Check if a call to SetLocalDescription is acceptable with |action|.
458 bool ExpectSetLocalDescription(Action action);
459 // Check if a call to SetRemoteDescription is acceptable with |action|.
460 bool ExpectSetRemoteDescription(Action action);
461 // Verifies a=setup attribute as per RFC 5763.
462 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
463 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000464
jiayl@webrtc.orge10d28c2014-07-17 17:07:49 +0000465 // Returns true if we are ready to push down the remote candidate.
466 // |remote_desc| is the new remote description, or NULL if the current remote
467 // description should be used. Output |valid| is true if the candidate media
468 // index is valid.
469 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
470 const SessionDescriptionInterface* remote_desc,
471 bool* valid);
472
deadbeef7af91dd2016-12-13 11:29:11 -0800473 // Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
474 // this session.
475 bool SrtpRequired() const;
476
deadbeefcbecd352015-09-23 11:50:27 -0700477 void OnTransportControllerConnectionState(cricket::IceConnectionState state);
478 void OnTransportControllerReceiving(bool receiving);
479 void OnTransportControllerGatheringState(cricket::IceGatheringState state);
480 void OnTransportControllerCandidatesGathered(
481 const std::string& transport_name,
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700482 const std::vector<cricket::Candidate>& candidates);
483 void OnTransportControllerCandidatesRemoved(
484 const std::vector<cricket::Candidate>& candidates);
deadbeefcbecd352015-09-23 11:50:27 -0700485
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000486 std::string GetSessionErrorMsg();
487
deadbeefcbecd352015-09-23 11:50:27 -0700488 // Invoked when TransportController connection completion is signaled.
489 // Reports stats for all transports in use.
490 void ReportTransportStats();
491
492 // Gather the usage of IPv4/IPv6 as best connection.
jbauchac8869e2015-07-03 01:36:14 -0700493 void ReportBestConnectionState(const cricket::TransportStats& stats);
494
495 void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000496
Danil Chapovalov33b01f22016-05-11 19:55:27 +0200497 void OnSentPacket_w(const rtc::SentPacket& sent_packet);
stefanc1aeaf02015-10-15 07:26:07 -0700498
zhihuang9763d562016-08-05 11:14:50 -0700499 const std::string GetTransportName(const std::string& content_name);
500
zhihuangd82eee02016-08-26 11:25:05 -0700501 void OnDtlsHandshakeError(rtc::SSLHandshakeError error);
502
zhihuang9763d562016-08-05 11:14:50 -0700503 rtc::Thread* const network_thread_;
deadbeefd59daf82015-10-14 15:02:44 -0700504 rtc::Thread* const worker_thread_;
danilchape9021a32016-05-17 01:52:02 -0700505 rtc::Thread* const signaling_thread_;
deadbeefd59daf82015-10-14 15:02:44 -0700506
507 State state_ = STATE_INIT;
508 Error error_ = ERROR_NONE;
509 std::string error_desc_;
510
511 const std::string sid_;
512 bool initial_offerer_ = false;
513
hbosdf6075a2016-12-19 04:58:02 -0800514 const std::unique_ptr<cricket::TransportController> transport_controller_;
stefanc1aeaf02015-10-15 07:26:07 -0700515 MediaControllerInterface* media_controller_;
kwibergd1fe2812016-04-27 06:47:29 -0700516 std::unique_ptr<cricket::VoiceChannel> voice_channel_;
517 std::unique_ptr<cricket::VideoChannel> video_channel_;
518 std::unique_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000519 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000520 IceObserver* ice_observer_;
521 PeerConnectionInterface::IceConnectionState ice_connection_state_;
Peter Thatcher54360512015-07-08 11:08:35 -0700522 bool ice_connection_receiving_;
kwibergd1fe2812016-04-27 06:47:29 -0700523 std::unique_ptr<SessionDescriptionInterface> local_desc_;
524 std::unique_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000525 // If the remote peer is using a older version of implementation.
526 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000527 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000528 // Specifies which kind of data channel is allowed. This is controlled
529 // by the chrome command-line flag and constraints:
530 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
531 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
532 // not set or false, SCTP is allowed (DCT_SCTP);
533 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
534 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
zhihuang9763d562016-08-05 11:14:50 -0700535 // The data channel type could be DCT_QUIC if the QUIC data channel is
536 // enabled.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000537 cricket::DataChannelType data_channel_type_;
deadbeef0ed85b22016-02-23 17:24:52 -0800538 // List of content names for which the remote side triggered an ICE restart.
539 std::set<std::string> pending_ice_restarts_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000540
kwibergd1fe2812016-04-27 06:47:29 -0700541 std::unique_ptr<WebRtcSessionDescriptionFactory> webrtc_session_desc_factory_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000542
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000543 // Member variables for caching global options.
544 cricket::AudioOptions audio_options_;
545 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000546 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000547
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000548 // Declares the bundle policy for the WebRTCSession.
549 PeerConnectionInterface::BundlePolicy bundle_policy_;
550
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700551 // Declares the RTCP mux policy for the WebRTCSession.
552 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
553
zhihuang184a3fd2016-06-14 11:47:14 -0700554 bool received_first_video_packet_ = false;
555 bool received_first_audio_packet_ = false;
556
zhihuang9763d562016-08-05 11:14:50 -0700557#ifdef HAVE_QUIC
558 std::unique_ptr<QuicDataTransport> quic_data_transport_;
559#endif // HAVE_QUIC
560
henrikg3c089d72015-09-16 05:37:44 -0700561 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
wu@webrtc.org364f2042013-11-20 21:49:41 +0000562};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000563} // namespace webrtc
564
Henrik Kjellander15583c12016-02-10 10:53:12 +0100565#endif // WEBRTC_API_WEBRTCSESSION_H_