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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"
22#include "webrtc/api/mediastreamprovider.h"
23#include "webrtc/api/peerconnectioninterface.h"
24#include "webrtc/api/statstypes.h"
kwiberg4485ffb2016-04-26 08:14:39 -070025#include "webrtc/base/constructormagic.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000026#include "webrtc/base/sigslot.h"
Henrik Boström5e56c592015-08-11 10:33:13 +020027#include "webrtc/base/sslidentity.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000028#include "webrtc/base/thread.h"
kjellandera96e2d72016-02-04 23:52:28 -080029#include "webrtc/media/base/mediachannel.h"
Honghai Zhang7fb69db2016-03-14 11:59:18 -070030#include "webrtc/p2p/base/candidate.h"
Tommif888bb52015-12-12 01:37:01 +010031#include "webrtc/p2p/base/transportcontroller.h"
kjellander@webrtc.org9b8df252016-02-12 06:47:59 +010032#include "webrtc/pc/mediasession.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000033
34namespace cricket {
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000035
henrike@webrtc.org28e20752013-07-10 00:45:36 +000036class ChannelManager;
37class DataChannel;
38class StatsReport;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000039class VideoChannel;
40class VoiceChannel;
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000041
henrike@webrtc.org28e20752013-07-10 00:45:36 +000042} // namespace cricket
43
44namespace webrtc {
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000045
henrike@webrtc.org28e20752013-07-10 00:45:36 +000046class IceRestartAnswerLatch;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000047class JsepIceCandidate;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000048class MediaStreamSignaling;
wu@webrtc.org91053e72013-08-10 07:18:04 +000049class WebRtcSessionDescriptionFactory;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000050
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000051extern const char kBundleWithoutRtcpMux[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000052extern const char kCreateChannelFailed[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000053extern const char kInvalidCandidates[];
54extern const char kInvalidSdp[];
55extern const char kMlineMismatch[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000056extern const char kPushDownTDFailed[];
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +000057extern const char kSdpWithoutDtlsFingerprint[];
58extern const char kSdpWithoutSdesCrypto[];
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000059extern const char kSdpWithoutIceUfragPwd[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000060extern const char kSdpWithoutSdesAndDtlsDisabled[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000061extern const char kSessionError[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000062extern const char kSessionErrorDesc[];
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +000063extern const char kDtlsSetupFailureRtp[];
64extern const char kDtlsSetupFailureRtcp[];
deadbeefcbecd352015-09-23 11:50:27 -070065extern const char kEnableBundleFailed[];
66
buildbot@webrtc.org53df88c2014-08-07 22:46:01 +000067// Maximum number of received video streams that will be processed by webrtc
68// even if they are not signalled beforehand.
69extern const int kMaxUnsignalledRecvStreams;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000070
71// ICE state callback interface.
72class IceObserver {
73 public:
wu@webrtc.org364f2042013-11-20 21:49:41 +000074 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +000075 // Called any time the IceConnectionState changes
Peter Thatcher54360512015-07-08 11:08:35 -070076 // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to
77 // conform to the w3c standard.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000078 virtual void OnIceConnectionChange(
79 PeerConnectionInterface::IceConnectionState new_state) {}
80 // Called any time the IceGatheringState changes
81 virtual void OnIceGatheringChange(
82 PeerConnectionInterface::IceGatheringState new_state) {}
83 // New Ice candidate have been found.
84 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000085
Honghai Zhang7fb69db2016-03-14 11:59:18 -070086 // Some local ICE candidates have been removed.
87 virtual void OnIceCandidatesRemoved(
88 const std::vector<cricket::Candidate>& candidates) = 0;
89
Peter Thatcher54360512015-07-08 11:08:35 -070090 // Called whenever the state changes between receiving and not receiving.
91 virtual void OnIceConnectionReceivingChange(bool receiving) {}
92
henrike@webrtc.org28e20752013-07-10 00:45:36 +000093 protected:
94 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +000095
96 private:
henrikg3c089d72015-09-16 05:37:44 -070097 RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000098};
99
deadbeefd59daf82015-10-14 15:02:44 -0700100// Statistics for all the transports of the session.
101typedef std::map<std::string, cricket::TransportStats> TransportStatsMap;
102typedef std::map<std::string, std::string> ProxyTransportMap;
103
104// TODO(pthatcher): Think of a better name for this. We already have
105// a TransportStats in transport.h. Perhaps TransportsStats?
106struct SessionStats {
107 ProxyTransportMap proxy_to_transport;
108 TransportStatsMap transport_stats;
109};
110
111// A WebRtcSession manages general session state. This includes negotiation
112// of both the application-level and network-level protocols: the former
113// defines what will be sent and the latter defines how it will be sent. Each
114// network-level protocol is represented by a Transport object. Each Transport
115// participates in the network-level negotiation. The individual streams of
116// packets are represented by TransportChannels. The application-level protocol
117// is represented by SessionDecription objects.
118class WebRtcSession : public AudioProviderInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000119 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000120 public DtmfProviderInterface,
deadbeefd59daf82015-10-14 15:02:44 -0700121 public DataChannelProviderInterface,
122 public sigslot::has_slots<> {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000123 public:
deadbeefd59daf82015-10-14 15:02:44 -0700124 enum State {
125 STATE_INIT = 0,
126 STATE_SENTOFFER, // Sent offer, waiting for answer.
127 STATE_RECEIVEDOFFER, // Received an offer. Need to send answer.
128 STATE_SENTPRANSWER, // Sent provisional answer. Need to send answer.
129 STATE_RECEIVEDPRANSWER, // Received provisional answer, waiting for answer.
130 STATE_INPROGRESS, // Offer/answer exchange completed.
131 STATE_CLOSED, // Close() was called.
132 };
133
134 enum Error {
135 ERROR_NONE = 0, // no error
136 ERROR_CONTENT = 1, // channel errors in SetLocalContent/SetRemoteContent
137 ERROR_TRANSPORT = 2, // transport error of some kind
138 };
139
stefanc1aeaf02015-10-15 07:26:07 -0700140 WebRtcSession(webrtc::MediaControllerInterface* media_controller,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000141 rtc::Thread* signaling_thread,
142 rtc::Thread* worker_thread,
deadbeefab9b2d12015-10-14 11:33:11 -0700143 cricket::PortAllocator* port_allocator);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000144 virtual ~WebRtcSession();
145
deadbeefd59daf82015-10-14 15:02:44 -0700146 // These are const to allow them to be called from const methods.
147 rtc::Thread* signaling_thread() const { return signaling_thread_; }
148 rtc::Thread* worker_thread() const { return worker_thread_; }
149 cricket::PortAllocator* port_allocator() const { return port_allocator_; }
150
151 // The ID of this session.
152 const std::string& id() const { return sid_; }
153
Henrik Lundin64dad832015-05-11 12:44:23 +0200154 bool Initialize(
155 const PeerConnectionFactoryInterface::Options& options,
kwibergd1fe2812016-04-27 06:47:29 -0700156 std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
Henrik Lundin64dad832015-05-11 12:44:23 +0200157 const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000158 // Deletes the voice, video and data channel and changes the session state
deadbeefd59daf82015-10-14 15:02:44 -0700159 // to STATE_CLOSED.
160 void Close();
161
162 // Returns true if we were the initial offerer.
163 bool initial_offerer() const { return initial_offerer_; }
164
165 // Returns the current state of the session. See the enum above for details.
166 // Each time the state changes, we will fire this signal.
167 State state() const { return state_; }
168 sigslot::signal2<WebRtcSession*, State> SignalState;
169
170 // Returns the last error in the session. See the enum above for details.
171 Error error() const { return error_; }
172 const std::string& error_desc() const { return error_desc_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000173
174 void RegisterIceObserver(IceObserver* observer) {
175 ice_observer_ = observer;
176 }
177
178 virtual cricket::VoiceChannel* voice_channel() {
179 return voice_channel_.get();
180 }
181 virtual cricket::VideoChannel* video_channel() {
182 return video_channel_.get();
183 }
184 virtual cricket::DataChannel* data_channel() {
185 return data_channel_.get();
186 }
187
deadbeef0ed85b22016-02-23 17:24:52 -0800188 cricket::BaseChannel* GetChannel(const std::string& content_name);
189
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000190 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
191 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000192
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000193 // Get current ssl role from transport.
Taylor Brandstetterf475d362016-01-08 15:35:57 -0800194 bool GetSslRole(const std::string& transport_name, rtc::SSLRole* role);
195
196 // Get current SSL role for this channel's transport.
197 // If |transport| is null, returns false.
198 bool GetSslRole(const cricket::BaseChannel* channel, rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000199
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000200 void CreateOffer(
201 CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 11:33:11 -0700202 const PeerConnectionInterface::RTCOfferAnswerOptions& options,
203 const cricket::MediaSessionOptions& session_options);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000204 void CreateAnswer(CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 11:33:11 -0700205 const cricket::MediaSessionOptions& session_options);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000206 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000207 bool SetLocalDescription(SessionDescriptionInterface* desc,
208 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000209 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000210 bool SetRemoteDescription(SessionDescriptionInterface* desc,
211 std::string* err_desc);
212 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000213
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700214 bool RemoveRemoteIceCandidates(
215 const std::vector<cricket::Candidate>& candidates);
216
mallinath@webrtc.org3d81b1b2014-09-09 14:38:10 +0000217 bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000218
honghaiz1f429e32015-09-28 07:57:34 -0700219 cricket::IceConfig ParseIceConfig(
220 const PeerConnectionInterface::RTCConfiguration& config) const;
221
deadbeefd59daf82015-10-14 15:02:44 -0700222 void SetIceConfig(const cricket::IceConfig& ice_config);
223
224 // Start gathering candidates for any new transports, or transports doing an
225 // ICE restart.
226 void MaybeStartGathering();
227
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000228 const SessionDescriptionInterface* local_description() const {
229 return local_desc_.get();
230 }
231 const SessionDescriptionInterface* remote_description() const {
232 return remote_desc_.get();
233 }
234
235 // Get the id used as a media stream track's "id" field from ssrc.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200236 virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
237 virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000238
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000239 // AudioMediaProviderInterface implementation.
solenbergd4cec0d2015-10-09 08:55:48 -0700240 void SetAudioPlayout(uint32_t ssrc, bool enable) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200241 void SetAudioSend(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000242 bool enable,
243 const cricket::AudioOptions& options,
Taylor Brandstetter1a018dc2016-03-08 12:37:39 -0800244 cricket::AudioSource* source) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200245 void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
deadbeef2d110be2016-01-13 12:00:26 -0800246 void SetRawAudioSink(uint32_t ssrc,
kwibergd1fe2812016-04-27 06:47:29 -0700247 std::unique_ptr<AudioSinkInterface> sink) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000248
skvladdc1c62c2016-03-16 19:07:43 -0700249 RtpParameters GetAudioRtpParameters(uint32_t ssrc) const override;
250 bool SetAudioRtpParameters(uint32_t ssrc,
251 const RtpParameters& parameters) override;
252
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000253 // Implements VideoMediaProviderInterface.
nisse2ded9b12016-04-08 02:23:55 -0700254 bool SetSource(
255 uint32_t ssrc,
256 rtc::VideoSourceInterface<cricket::VideoFrame>* source) override;
nisse08582ff2016-02-04 01:24:52 -0800257 void SetVideoPlayout(
258 uint32_t ssrc,
259 bool enable,
260 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200261 void SetVideoSend(uint32_t ssrc,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000262 bool enable,
263 const cricket::VideoOptions* options) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000264
skvladdc1c62c2016-03-16 19:07:43 -0700265 RtpParameters GetVideoRtpParameters(uint32_t ssrc) const override;
266 bool SetVideoRtpParameters(uint32_t ssrc,
267 const RtpParameters& parameters) override;
268
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000269 // Implements DtmfProviderInterface.
nisseef8b61e2016-04-29 06:09:15 -0700270 bool CanInsertDtmf(const std::string& track_id) override;
271 bool InsertDtmf(const std::string& track_id,
272 int code, int duration) override;
273 sigslot::signal0<>* GetOnDestroyedSignal() override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000274
wu@webrtc.org78187522013-10-07 23:32:02 +0000275 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000276 bool SendData(const cricket::SendDataParams& params,
jbaucheec21bd2016-03-20 06:15:43 -0700277 const rtc::CopyOnWriteBuffer& payload,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000278 cricket::SendDataResult* result) override;
279 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
280 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
281 void AddSctpDataStream(int sid) override;
282 void RemoveSctpDataStream(int sid) override;
283 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02 +0000284
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000285 // Returns stats for all channels of all transports.
286 // This avoids exposing the internal structures used to track them.
deadbeefd59daf82015-10-14 15:02:44 -0700287 virtual bool GetTransportStats(SessionStats* stats);
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000288
deadbeefcbecd352015-09-23 11:50:27 -0700289 // Get stats for a specific channel
deadbeefd59daf82015-10-14 15:02:44 -0700290 bool GetChannelTransportStats(cricket::BaseChannel* ch, SessionStats* stats);
deadbeefcbecd352015-09-23 11:50:27 -0700291
292 // virtual so it can be mocked in unit tests
293 virtual bool GetLocalCertificate(
294 const std::string& transport_name,
295 rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
296
297 // Caller owns returned certificate
jbauch555604a2016-04-26 03:13:22 -0700298 virtual std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
kwibergb4d01c42016-04-06 05:15:06 -0700299 const std::string& transport_name);
deadbeefcbecd352015-09-23 11:50:27 -0700300
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000301 cricket::DataChannelType data_channel_type() const;
302
deadbeef0ed85b22016-02-23 17:24:52 -0800303 bool IceRestartPending(const std::string& content_name) const;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000304
Henrik Boströmd8281982015-08-27 10:12:24 +0200305 // Called when an RTCCertificate is generated or retrieved by
wu@webrtc.org91053e72013-08-10 07:18:04 +0000306 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
Henrik Boströmd8281982015-08-27 10:12:24 +0200307 void OnCertificateReady(
308 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
pthatcher@webrtc.org4eeef582015-03-16 19:34:23 +0000309 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000310
311 // For unit test.
Henrik Boströmd8281982015-08-27 10:12:24 +0200312 bool waiting_for_certificate_for_testing() const;
deadbeefcbecd352015-09-23 11:50:27 -0700313 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000314
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000315 void set_metrics_observer(
316 webrtc::MetricsObserverInterface* metrics_observer) {
317 metrics_observer_ = metrics_observer;
318 }
319
deadbeefab9b2d12015-10-14 11:33:11 -0700320 // Called when voice_channel_, video_channel_ and data_channel_ are created
321 // and destroyed. As a result of, for example, setting a new description.
322 sigslot::signal0<> SignalVoiceChannelCreated;
323 sigslot::signal0<> SignalVoiceChannelDestroyed;
324 sigslot::signal0<> SignalVideoChannelCreated;
325 sigslot::signal0<> SignalVideoChannelDestroyed;
326 sigslot::signal0<> SignalDataChannelCreated;
327 sigslot::signal0<> SignalDataChannelDestroyed;
deadbeef057ecf02016-01-20 14:30:43 -0800328 // Called when the whole session is destroyed.
329 sigslot::signal0<> SignalDestroyed;
deadbeefab9b2d12015-10-14 11:33:11 -0700330
331 // Called when a valid data channel OPEN message is received.
332 // std::string represents the data channel label.
333 sigslot::signal2<const std::string&, const InternalDataChannelInit&>
334 SignalDataChannelOpenMessage;
335
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000336 private:
337 // Indicates the type of SessionDescription in a call to SetLocalDescription
338 // and SetRemoteDescription.
339 enum Action {
340 kOffer,
341 kPrAnswer,
342 kAnswer,
343 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000344
deadbeefd59daf82015-10-14 15:02:44 -0700345 // Log session state.
346 void LogState(State old_state, State new_state);
347
348 // Updates the state, signaling if necessary.
349 virtual void SetState(State state);
350
351 // Updates the error state, signaling if necessary.
352 // TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|.
353 virtual void SetError(Error error, const std::string& error_desc);
354
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000355 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000356 std::string* err_desc);
357 static Action GetAction(const std::string& type);
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000358 // Push the media parts of the local or remote session description
359 // down to all of the channels.
360 bool PushdownMediaDescription(cricket::ContentAction action,
361 cricket::ContentSource source,
362 std::string* error_desc);
363
deadbeefd59daf82015-10-14 15:02:44 -0700364 bool PushdownTransportDescription(cricket::ContentSource source,
365 cricket::ContentAction action,
366 std::string* error_desc);
367
368 // Helper methods to push local and remote transport descriptions.
369 bool PushdownLocalTransportDescription(
370 const cricket::SessionDescription* sdesc,
371 cricket::ContentAction action,
372 std::string* error_desc);
373 bool PushdownRemoteTransportDescription(
374 const cricket::SessionDescription* sdesc,
375 cricket::ContentAction action,
376 std::string* error_desc);
377
378 // Returns true and the TransportInfo of the given |content_name|
379 // from |description|. Returns false if it's not available.
380 static bool GetTransportDescription(
381 const cricket::SessionDescription* description,
382 const std::string& content_name,
383 cricket::TransportDescription* info);
384
deadbeefcbecd352015-09-23 11:50:27 -0700385 // Cause all the BaseChannels in the bundle group to have the same
386 // transport channel.
387 bool EnableBundle(const cricket::ContentGroup& bundle);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000388
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000389 // Enables media channels to allow sending of media.
390 void EnableChannels();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000391 // Returns the media index for a local ice candidate given the content name.
392 // Returns false if the local session description does not have a media
393 // content called |content_name|.
394 bool GetLocalCandidateMediaIndex(const std::string& content_name,
395 int* sdp_mline_index);
396 // Uses all remote candidates in |remote_desc| in this session.
397 bool UseCandidatesInSessionDescription(
398 const SessionDescriptionInterface* remote_desc);
399 // Uses |candidate| in this session.
400 bool UseCandidate(const IceCandidateInterface* candidate);
401 // Deletes the corresponding channel of contents that don't exist in |desc|.
402 // |desc| can be null. This means that all channels are deleted.
deadbeefcbecd352015-09-23 11:50:27 -0700403 void RemoveUnusedChannels(const cricket::SessionDescription* desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000404
405 // Allocates media channels based on the |desc|. If |desc| doesn't have
406 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
407 // This method will also delete any existing media channels before creating.
408 bool CreateChannels(const cricket::SessionDescription* desc);
409
410 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000411 bool CreateVoiceChannel(const cricket::ContentInfo* content);
412 bool CreateVideoChannel(const cricket::ContentInfo* content);
413 bool CreateDataChannel(const cricket::ContentInfo* content);
414
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000415 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
416 // messages.
417 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
418 const cricket::ReceiveDataParams& params,
jbaucheec21bd2016-03-20 06:15:43 -0700419 const rtc::CopyOnWriteBuffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000420
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000421 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000422 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
Peter Thatcher54360512015-07-08 11:08:35 -0700423 void SetIceConnectionReceiving(bool receiving);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000424
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000425 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000426 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000427 // Below methods are helper methods which verifies SDP.
428 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
429 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000430 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000431
432 // Check if a call to SetLocalDescription is acceptable with |action|.
433 bool ExpectSetLocalDescription(Action action);
434 // Check if a call to SetRemoteDescription is acceptable with |action|.
435 bool ExpectSetRemoteDescription(Action action);
436 // Verifies a=setup attribute as per RFC 5763.
437 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
438 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000439
jiayl@webrtc.orge10d28c2014-07-17 17:07:49 +0000440 // Returns true if we are ready to push down the remote candidate.
441 // |remote_desc| is the new remote description, or NULL if the current remote
442 // description should be used. Output |valid| is true if the candidate media
443 // index is valid.
444 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
445 const SessionDescriptionInterface* remote_desc,
446 bool* valid);
447
deadbeefcbecd352015-09-23 11:50:27 -0700448 void OnTransportControllerConnectionState(cricket::IceConnectionState state);
449 void OnTransportControllerReceiving(bool receiving);
450 void OnTransportControllerGatheringState(cricket::IceGatheringState state);
451 void OnTransportControllerCandidatesGathered(
452 const std::string& transport_name,
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700453 const std::vector<cricket::Candidate>& candidates);
454 void OnTransportControllerCandidatesRemoved(
455 const std::vector<cricket::Candidate>& candidates);
deadbeefcbecd352015-09-23 11:50:27 -0700456
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000457 std::string GetSessionErrorMsg();
458
deadbeefcbecd352015-09-23 11:50:27 -0700459 // Invoked when TransportController connection completion is signaled.
460 // Reports stats for all transports in use.
461 void ReportTransportStats();
462
463 // Gather the usage of IPv4/IPv6 as best connection.
jbauchac8869e2015-07-03 01:36:14 -0700464 void ReportBestConnectionState(const cricket::TransportStats& stats);
465
466 void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000467
stefanc1aeaf02015-10-15 07:26:07 -0700468 void OnSentPacket_w(cricket::TransportChannel* channel,
469 const rtc::SentPacket& sent_packet);
470
deadbeefd59daf82015-10-14 15:02:44 -0700471 rtc::Thread* const signaling_thread_;
472 rtc::Thread* const worker_thread_;
473 cricket::PortAllocator* const port_allocator_;
474
475 State state_ = STATE_INIT;
476 Error error_ = ERROR_NONE;
477 std::string error_desc_;
478
479 const std::string sid_;
480 bool initial_offerer_ = false;
481
kwibergd1fe2812016-04-27 06:47:29 -0700482 std::unique_ptr<cricket::TransportController> transport_controller_;
stefanc1aeaf02015-10-15 07:26:07 -0700483 MediaControllerInterface* media_controller_;
kwibergd1fe2812016-04-27 06:47:29 -0700484 std::unique_ptr<cricket::VoiceChannel> voice_channel_;
485 std::unique_ptr<cricket::VideoChannel> video_channel_;
486 std::unique_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000487 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000488 IceObserver* ice_observer_;
489 PeerConnectionInterface::IceConnectionState ice_connection_state_;
Peter Thatcher54360512015-07-08 11:08:35 -0700490 bool ice_connection_receiving_;
kwibergd1fe2812016-04-27 06:47:29 -0700491 std::unique_ptr<SessionDescriptionInterface> local_desc_;
492 std::unique_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000493 // If the remote peer is using a older version of implementation.
494 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000495 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000496 // Specifies which kind of data channel is allowed. This is controlled
497 // by the chrome command-line flag and constraints:
498 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
499 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
500 // not set or false, SCTP is allowed (DCT_SCTP);
501 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
502 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
503 cricket::DataChannelType data_channel_type_;
deadbeef0ed85b22016-02-23 17:24:52 -0800504 // List of content names for which the remote side triggered an ICE restart.
505 std::set<std::string> pending_ice_restarts_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000506
kwibergd1fe2812016-04-27 06:47:29 -0700507 std::unique_ptr<WebRtcSessionDescriptionFactory> webrtc_session_desc_factory_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000508
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000509 // Member variables for caching global options.
510 cricket::AudioOptions audio_options_;
511 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000512 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000513
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000514 // Declares the bundle policy for the WebRTCSession.
515 PeerConnectionInterface::BundlePolicy bundle_policy_;
516
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700517 // Declares the RTCP mux policy for the WebRTCSession.
518 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
519
henrikg3c089d72015-09-16 05:37:44 -0700520 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
wu@webrtc.org364f2042013-11-20 21:49:41 +0000521};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000522} // namespace webrtc
523
Henrik Kjellander15583c12016-02-10 10:53:12 +0100524#endif // WEBRTC_API_WEBRTCSESSION_H_