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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"
deadbeef67b3bbe2017-01-04 18:38:02 -080025#include "webrtc/base/optional.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
zhihuang9763d562016-08-05 11:14:50 -070034#ifdef HAVE_QUIC
35#include "webrtc/api/quicdatatransport.h"
36#endif // HAVE_QUIC
37
henrike@webrtc.org28e20752013-07-10 00:45:36 +000038namespace cricket {
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000039
henrike@webrtc.org28e20752013-07-10 00:45:36 +000040class ChannelManager;
deadbeef67b3bbe2017-01-04 18:38:02 -080041class RtpDataChannel;
42class SctpTransportInternal;
43class SctpTransportInternalFactory;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000044class StatsReport;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000045class VideoChannel;
46class VoiceChannel;
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000047
zhihuang9763d562016-08-05 11:14:50 -070048#ifdef HAVE_QUIC
49class QuicTransportChannel;
50#endif // HAVE_QUIC
51
henrike@webrtc.org28e20752013-07-10 00:45:36 +000052} // namespace cricket
53
54namespace webrtc {
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000055
henrike@webrtc.org28e20752013-07-10 00:45:36 +000056class IceRestartAnswerLatch;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000057class JsepIceCandidate;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000058class MediaStreamSignaling;
wu@webrtc.org91053e72013-08-10 07:18:04 +000059class WebRtcSessionDescriptionFactory;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000060
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000061extern const char kBundleWithoutRtcpMux[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000062extern const char kCreateChannelFailed[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000063extern const char kInvalidCandidates[];
64extern const char kInvalidSdp[];
65extern const char kMlineMismatch[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000066extern const char kPushDownTDFailed[];
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +000067extern const char kSdpWithoutDtlsFingerprint[];
68extern const char kSdpWithoutSdesCrypto[];
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000069extern const char kSdpWithoutIceUfragPwd[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000070extern const char kSdpWithoutSdesAndDtlsDisabled[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000071extern const char kSessionError[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000072extern const char kSessionErrorDesc[];
deadbeef67b3bbe2017-01-04 18:38:02 -080073extern const char kDtlsSrtpSetupFailureRtp[];
74extern const char kDtlsSrtpSetupFailureRtcp[];
deadbeefcbecd352015-09-23 11:50:27 -070075extern const char kEnableBundleFailed[];
76
buildbot@webrtc.org53df88c2014-08-07 22:46:01 +000077// Maximum number of received video streams that will be processed by webrtc
78// even if they are not signalled beforehand.
79extern const int kMaxUnsignalledRecvStreams;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000080
81// ICE state callback interface.
82class IceObserver {
83 public:
wu@webrtc.org364f2042013-11-20 21:49:41 +000084 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +000085 // Called any time the IceConnectionState changes
Peter Thatcher54360512015-07-08 11:08:35 -070086 // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to
87 // conform to the w3c standard.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000088 virtual void OnIceConnectionChange(
89 PeerConnectionInterface::IceConnectionState new_state) {}
90 // Called any time the IceGatheringState changes
91 virtual void OnIceGatheringChange(
92 PeerConnectionInterface::IceGatheringState new_state) {}
93 // New Ice candidate have been found.
94 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000095
Honghai Zhang7fb69db2016-03-14 11:59:18 -070096 // Some local ICE candidates have been removed.
97 virtual void OnIceCandidatesRemoved(
98 const std::vector<cricket::Candidate>& candidates) = 0;
99
Peter Thatcher54360512015-07-08 11:08:35 -0700100 // Called whenever the state changes between receiving and not receiving.
101 virtual void OnIceConnectionReceivingChange(bool receiving) {}
102
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000103 protected:
104 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +0000105
106 private:
henrikg3c089d72015-09-16 05:37:44 -0700107 RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000108};
109
deadbeefd59daf82015-10-14 15:02:44 -0700110// Statistics for all the transports of the session.
111typedef std::map<std::string, cricket::TransportStats> TransportStatsMap;
112typedef std::map<std::string, std::string> ProxyTransportMap;
113
114// TODO(pthatcher): Think of a better name for this. We already have
115// a TransportStats in transport.h. Perhaps TransportsStats?
116struct SessionStats {
117 ProxyTransportMap proxy_to_transport;
118 TransportStatsMap transport_stats;
119};
120
hbosdf6075a2016-12-19 04:58:02 -0800121struct ChannelNamePair {
122 ChannelNamePair(
123 const std::string& content_name, const std::string& transport_name)
124 : content_name(content_name), transport_name(transport_name) {}
125 std::string content_name;
126 std::string transport_name;
127};
128
129struct ChannelNamePairs {
130 rtc::Optional<ChannelNamePair> voice;
131 rtc::Optional<ChannelNamePair> video;
132 rtc::Optional<ChannelNamePair> data;
133};
134
deadbeefd59daf82015-10-14 15:02:44 -0700135// A WebRtcSession manages general session state. This includes negotiation
136// of both the application-level and network-level protocols: the former
137// defines what will be sent and the latter defines how it will be sent. Each
138// network-level protocol is represented by a Transport object. Each Transport
139// participates in the network-level negotiation. The individual streams of
140// packets are represented by TransportChannels. The application-level protocol
141// is represented by SessionDecription objects.
Taylor Brandstetterba29c6a2016-06-27 16:30:35 -0700142class WebRtcSession :
143
144 public DtmfProviderInterface,
145 public DataChannelProviderInterface,
146 public sigslot::has_slots<> {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000147 public:
deadbeefd59daf82015-10-14 15:02:44 -0700148 enum State {
149 STATE_INIT = 0,
150 STATE_SENTOFFER, // Sent offer, waiting for answer.
151 STATE_RECEIVEDOFFER, // Received an offer. Need to send answer.
152 STATE_SENTPRANSWER, // Sent provisional answer. Need to send answer.
153 STATE_RECEIVEDPRANSWER, // Received provisional answer, waiting for answer.
154 STATE_INPROGRESS, // Offer/answer exchange completed.
155 STATE_CLOSED, // Close() was called.
156 };
157
158 enum Error {
159 ERROR_NONE = 0, // no error
160 ERROR_CONTENT = 1, // channel errors in SetLocalContent/SetRemoteContent
161 ERROR_TRANSPORT = 2, // transport error of some kind
162 };
163
deadbeef67b3bbe2017-01-04 18:38:02 -0800164 // |sctp_factory| may be null, in which case SCTP is treated as unsupported.
zhihuang29ff8442016-07-27 11:07:25 -0700165 WebRtcSession(
166 webrtc::MediaControllerInterface* media_controller,
167 rtc::Thread* network_thread,
168 rtc::Thread* worker_thread,
169 rtc::Thread* signaling_thread,
170 cricket::PortAllocator* port_allocator,
deadbeef67b3bbe2017-01-04 18:38:02 -0800171 std::unique_ptr<cricket::TransportController> transport_controller,
172 std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000173 virtual ~WebRtcSession();
174
deadbeefd59daf82015-10-14 15:02:44 -0700175 // These are const to allow them to be called from const methods.
zhihuang9763d562016-08-05 11:14:50 -0700176 rtc::Thread* network_thread() const { return network_thread_; }
deadbeefd59daf82015-10-14 15:02:44 -0700177 rtc::Thread* worker_thread() const { return worker_thread_; }
danilchape9021a32016-05-17 01:52:02 -0700178 rtc::Thread* signaling_thread() const { return signaling_thread_; }
deadbeefd59daf82015-10-14 15:02:44 -0700179
180 // The ID of this session.
181 const std::string& id() const { return sid_; }
182
Henrik Lundin64dad832015-05-11 12:44:23 +0200183 bool Initialize(
184 const PeerConnectionFactoryInterface::Options& options,
Henrik Boströmd03c23b2016-06-01 11:44:18 +0200185 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
Henrik Lundin64dad832015-05-11 12:44:23 +0200186 const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000187 // Deletes the voice, video and data channel and changes the session state
deadbeefd59daf82015-10-14 15:02:44 -0700188 // to STATE_CLOSED.
189 void Close();
190
191 // Returns true if we were the initial offerer.
192 bool initial_offerer() const { return initial_offerer_; }
193
194 // Returns the current state of the session. See the enum above for details.
195 // Each time the state changes, we will fire this signal.
196 State state() const { return state_; }
197 sigslot::signal2<WebRtcSession*, State> SignalState;
198
199 // Returns the last error in the session. See the enum above for details.
200 Error error() const { return error_; }
201 const std::string& error_desc() const { return error_desc_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000202
203 void RegisterIceObserver(IceObserver* observer) {
204 ice_observer_ = observer;
205 }
206
deadbeef67b3bbe2017-01-04 18:38:02 -0800207 // Exposed for stats collecting.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000208 virtual cricket::VoiceChannel* voice_channel() {
209 return voice_channel_.get();
210 }
211 virtual cricket::VideoChannel* video_channel() {
212 return video_channel_.get();
213 }
deadbeef67b3bbe2017-01-04 18:38:02 -0800214 // Only valid when using deprecated RTP data channels.
215 virtual cricket::RtpDataChannel* rtp_data_channel() {
216 return rtp_data_channel_.get();
217 }
218 virtual rtc::Optional<std::string> sctp_content_name() const {
219 return sctp_content_name_;
220 }
221 virtual rtc::Optional<std::string> sctp_transport_name() const {
222 return sctp_transport_name_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000223 }
224
deadbeef0ed85b22016-02-23 17:24:52 -0800225 cricket::BaseChannel* GetChannel(const std::string& content_name);
226
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000227 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000228
deadbeef67b3bbe2017-01-04 18:38:02 -0800229 // Get current SSL role used by SCTP's underlying transport.
230 bool GetSctpSslRole(rtc::SSLRole* role);
231 // Get SSL role for an arbitrary m= section (handles bundling correctly).
232 // TODO(deadbeef): This is only used internally by the session description
233 // factory, it shouldn't really be public).
234 bool GetSslRole(const std::string& content_name, rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000235
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000236 void CreateOffer(
237 CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 11:33:11 -0700238 const PeerConnectionInterface::RTCOfferAnswerOptions& options,
239 const cricket::MediaSessionOptions& session_options);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000240 void CreateAnswer(CreateSessionDescriptionObserver* observer,
deadbeefab9b2d12015-10-14 11:33:11 -0700241 const cricket::MediaSessionOptions& session_options);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000242 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000243 bool SetLocalDescription(SessionDescriptionInterface* desc,
244 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000245 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000246 bool SetRemoteDescription(SessionDescriptionInterface* desc,
247 std::string* err_desc);
deadbeef67b3bbe2017-01-04 18:38:02 -0800248
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000249 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000250
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700251 bool RemoveRemoteIceCandidates(
252 const std::vector<cricket::Candidate>& candidates);
253
honghaiz1f429e32015-09-28 07:57:34 -0700254 cricket::IceConfig ParseIceConfig(
255 const PeerConnectionInterface::RTCConfiguration& config) const;
256
deadbeefd59daf82015-10-14 15:02:44 -0700257 void SetIceConfig(const cricket::IceConfig& ice_config);
258
259 // Start gathering candidates for any new transports, or transports doing an
260 // ICE restart.
261 void MaybeStartGathering();
262
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000263 const SessionDescriptionInterface* local_description() const {
deadbeeffe4a8a42016-12-20 17:56:17 -0800264 return pending_local_description_ ? pending_local_description_.get()
265 : current_local_description_.get();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000266 }
267 const SessionDescriptionInterface* remote_description() const {
deadbeeffe4a8a42016-12-20 17:56:17 -0800268 return pending_remote_description_ ? pending_remote_description_.get()
269 : current_remote_description_.get();
270 }
271 const SessionDescriptionInterface* current_local_description() const {
272 return current_local_description_.get();
273 }
274 const SessionDescriptionInterface* current_remote_description() const {
275 return current_remote_description_.get();
276 }
277 const SessionDescriptionInterface* pending_local_description() const {
278 return pending_local_description_.get();
279 }
280 const SessionDescriptionInterface* pending_remote_description() const {
281 return pending_remote_description_.get();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000282 }
283
284 // Get the id used as a media stream track's "id" field from ssrc.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200285 virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
286 virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000287
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000288 // Implements DtmfProviderInterface.
nisseef8b61e2016-04-29 06:09:15 -0700289 bool CanInsertDtmf(const std::string& track_id) override;
290 bool InsertDtmf(const std::string& track_id,
291 int code, int duration) override;
292 sigslot::signal0<>* GetOnDestroyedSignal() override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000293
wu@webrtc.org78187522013-10-07 23:32:02 +0000294 // Implements DataChannelProviderInterface.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000295 bool SendData(const cricket::SendDataParams& params,
jbaucheec21bd2016-03-20 06:15:43 -0700296 const rtc::CopyOnWriteBuffer& payload,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000297 cricket::SendDataResult* result) override;
298 bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
299 void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
300 void AddSctpDataStream(int sid) override;
301 void RemoveSctpDataStream(int sid) override;
302 bool ReadyToSendData() const override;
wu@webrtc.org78187522013-10-07 23:32:02 +0000303
pthatcher@webrtc.orgc04a97f2015-03-16 19:31:40 +0000304 // Returns stats for all channels of all transports.
305 // This avoids exposing the internal structures used to track them.
hbosdf6075a2016-12-19 04:58:02 -0800306 // The parameterless version creates |ChannelNamePairs| from |voice_channel|,
307 // |video_channel| and |voice_channel| if available - this requires it to be
308 // called on the signaling thread - and invokes the other |GetStats|. The
309 // other |GetStats| can be invoked on any thread; if not invoked on the
310 // network thread a thread hop will happen.
311 std::unique_ptr<SessionStats> GetStats_s();
312 virtual std::unique_ptr<SessionStats> GetStats(
313 const ChannelNamePairs& channel_name_pairs);
deadbeefcbecd352015-09-23 11:50:27 -0700314
315 // virtual so it can be mocked in unit tests
316 virtual bool GetLocalCertificate(
317 const std::string& transport_name,
318 rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
319
320 // Caller owns returned certificate
jbauch555604a2016-04-26 03:13:22 -0700321 virtual std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
kwibergb4d01c42016-04-06 05:15:06 -0700322 const std::string& transport_name);
deadbeefcbecd352015-09-23 11:50:27 -0700323
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000324 cricket::DataChannelType data_channel_type() const;
325
deadbeefd1a38b52016-12-10 13:15:33 -0800326 // Returns true if there was an ICE restart initiated by the remote offer.
deadbeef0ed85b22016-02-23 17:24:52 -0800327 bool IceRestartPending(const std::string& content_name) const;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000328
deadbeefd1a38b52016-12-10 13:15:33 -0800329 // Set the "needs-ice-restart" flag as described in JSEP. After the flag is
330 // set, offers should generate new ufrags/passwords until an ICE restart
331 // occurs.
332 void SetNeedsIceRestartFlag();
333 // Returns true if the ICE restart flag above was set, and no ICE restart has
334 // occurred yet for this transport (by applying a local description with
335 // changed ufrag/password). If the transport has been deleted as a result of
336 // bundling, returns false.
337 bool NeedsIceRestart(const std::string& content_name) const;
338
Henrik Boströmd8281982015-08-27 10:12:24 +0200339 // Called when an RTCCertificate is generated or retrieved by
wu@webrtc.org91053e72013-08-10 07:18:04 +0000340 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
Henrik Boströmd8281982015-08-27 10:12:24 +0200341 void OnCertificateReady(
342 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
deadbeef67b3bbe2017-01-04 18:38:02 -0800343 void OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000344
345 // For unit test.
Henrik Boströmd8281982015-08-27 10:12:24 +0200346 bool waiting_for_certificate_for_testing() const;
deadbeefcbecd352015-09-23 11:50:27 -0700347 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
wu@webrtc.org91053e72013-08-10 07:18:04 +0000348
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000349 void set_metrics_observer(
350 webrtc::MetricsObserverInterface* metrics_observer) {
351 metrics_observer_ = metrics_observer;
Honghai Zhangd93f50c2016-10-05 11:47:22 -0700352 transport_controller_->SetMetricsObserver(metrics_observer);
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000353 }
354
deadbeef67b3bbe2017-01-04 18:38:02 -0800355 // Called when voice_channel_, video_channel_ and
356 // rtp_data_channel_/sctp_transport_ are created and destroyed. As a result
357 // of, for example, setting a new description.
deadbeefab9b2d12015-10-14 11:33:11 -0700358 sigslot::signal0<> SignalVoiceChannelCreated;
359 sigslot::signal0<> SignalVoiceChannelDestroyed;
360 sigslot::signal0<> SignalVideoChannelCreated;
361 sigslot::signal0<> SignalVideoChannelDestroyed;
362 sigslot::signal0<> SignalDataChannelCreated;
363 sigslot::signal0<> SignalDataChannelDestroyed;
deadbeef057ecf02016-01-20 14:30:43 -0800364 // Called when the whole session is destroyed.
365 sigslot::signal0<> SignalDestroyed;
deadbeefab9b2d12015-10-14 11:33:11 -0700366
367 // Called when a valid data channel OPEN message is received.
368 // std::string represents the data channel label.
369 sigslot::signal2<const std::string&, const InternalDataChannelInit&>
370 SignalDataChannelOpenMessage;
zhihuang9763d562016-08-05 11:14:50 -0700371#ifdef HAVE_QUIC
372 QuicDataTransport* quic_data_transport() {
373 return quic_data_transport_.get();
374 }
375#endif // HAVE_QUIC
deadbeefab9b2d12015-10-14 11:33:11 -0700376
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000377 private:
378 // Indicates the type of SessionDescription in a call to SetLocalDescription
379 // and SetRemoteDescription.
380 enum Action {
381 kOffer,
382 kPrAnswer,
383 kAnswer,
384 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000385
deadbeeffe4a8a42016-12-20 17:56:17 -0800386 // Non-const versions of local_description()/remote_description(), for use
387 // internally.
388 SessionDescriptionInterface* mutable_local_description() {
389 return pending_local_description_ ? pending_local_description_.get()
390 : current_local_description_.get();
391 }
392 SessionDescriptionInterface* mutable_remote_description() {
393 return pending_remote_description_ ? pending_remote_description_.get()
394 : current_remote_description_.get();
395 }
396
deadbeefd59daf82015-10-14 15:02:44 -0700397 // Log session state.
398 void LogState(State old_state, State new_state);
399
400 // Updates the state, signaling if necessary.
401 virtual void SetState(State state);
402
403 // Updates the error state, signaling if necessary.
404 // TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|.
405 virtual void SetError(Error error, const std::string& error_desc);
406
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000407 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000408 std::string* err_desc);
409 static Action GetAction(const std::string& type);
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000410 // Push the media parts of the local or remote session description
411 // down to all of the channels.
412 bool PushdownMediaDescription(cricket::ContentAction action,
413 cricket::ContentSource source,
414 std::string* error_desc);
deadbeef67b3bbe2017-01-04 18:38:02 -0800415 bool PushdownSctpParameters_n(cricket::ContentSource source);
pthatcher@webrtc.org592470b2015-03-16 21:15:37 +0000416
deadbeefd59daf82015-10-14 15:02:44 -0700417 bool PushdownTransportDescription(cricket::ContentSource source,
418 cricket::ContentAction action,
419 std::string* error_desc);
420
421 // Helper methods to push local and remote transport descriptions.
422 bool PushdownLocalTransportDescription(
423 const cricket::SessionDescription* sdesc,
424 cricket::ContentAction action,
425 std::string* error_desc);
426 bool PushdownRemoteTransportDescription(
427 const cricket::SessionDescription* sdesc,
428 cricket::ContentAction action,
429 std::string* error_desc);
430
431 // Returns true and the TransportInfo of the given |content_name|
432 // from |description|. Returns false if it's not available.
433 static bool GetTransportDescription(
434 const cricket::SessionDescription* description,
435 const std::string& content_name,
436 cricket::TransportDescription* info);
437
skvlad6c87a672016-05-17 17:49:52 -0700438 // Returns the name of the transport channel when BUNDLE is enabled, or
439 // nullptr if the channel is not part of any bundle.
440 const std::string* GetBundleTransportName(
441 const cricket::ContentInfo* content,
442 const cricket::ContentGroup* bundle);
443
deadbeefcbecd352015-09-23 11:50:27 -0700444 // Cause all the BaseChannels in the bundle group to have the same
445 // transport channel.
446 bool EnableBundle(const cricket::ContentGroup& bundle);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000447
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000448 // Enables media channels to allow sending of media.
449 void EnableChannels();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000450 // Returns the media index for a local ice candidate given the content name.
451 // Returns false if the local session description does not have a media
452 // content called |content_name|.
453 bool GetLocalCandidateMediaIndex(const std::string& content_name,
454 int* sdp_mline_index);
455 // Uses all remote candidates in |remote_desc| in this session.
456 bool UseCandidatesInSessionDescription(
457 const SessionDescriptionInterface* remote_desc);
458 // Uses |candidate| in this session.
459 bool UseCandidate(const IceCandidateInterface* candidate);
460 // Deletes the corresponding channel of contents that don't exist in |desc|.
461 // |desc| can be null. This means that all channels are deleted.
deadbeefcbecd352015-09-23 11:50:27 -0700462 void RemoveUnusedChannels(const cricket::SessionDescription* desc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000463
464 // Allocates media channels based on the |desc|. If |desc| doesn't have
465 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
466 // This method will also delete any existing media channels before creating.
467 bool CreateChannels(const cricket::SessionDescription* desc);
468
469 // Helper methods to create media channels.
skvlad6c87a672016-05-17 17:49:52 -0700470 bool CreateVoiceChannel(const cricket::ContentInfo* content,
471 const std::string* bundle_transport);
472 bool CreateVideoChannel(const cricket::ContentInfo* content,
473 const std::string* bundle_transport);
474 bool CreateDataChannel(const cricket::ContentInfo* content,
475 const std::string* bundle_transport);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000476
hbosdf6075a2016-12-19 04:58:02 -0800477 std::unique_ptr<SessionStats> GetStats_n(
478 const ChannelNamePairs& channel_name_pairs);
479
deadbeef67b3bbe2017-01-04 18:38:02 -0800480 bool CreateSctpTransport_n(const std::string& content_name,
481 const std::string& transport_name);
482 // For bundling.
483 void ChangeSctpTransport_n(const std::string& transport_name);
484 void DestroySctpTransport_n();
485 // SctpTransport signal handlers. Needed to marshal signals from the network
486 // to signaling thread.
487 void OnSctpTransportReadyToSendData_n();
488 // This may be called with "false" if the direction of the m= section causes
489 // us to tear down the SCTP connection.
490 void OnSctpTransportReadyToSendData_s(bool ready);
491 void OnSctpTransportDataReceived_n(const cricket::ReceiveDataParams& params,
492 const rtc::CopyOnWriteBuffer& payload);
493 // Beyond just firing the signal to the signaling thread, listens to SCTP
494 // CONTROL messages on unused SIDs and processes them as OPEN messages.
495 void OnSctpTransportDataReceived_s(const cricket::ReceiveDataParams& params,
496 const rtc::CopyOnWriteBuffer& payload);
497 void OnSctpStreamClosedRemotely_n(int sid);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000498
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000499 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000500 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
Peter Thatcher54360512015-07-08 11:08:35 -0700501 void SetIceConnectionReceiving(bool receiving);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000502
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000503 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000504 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000505 // Below methods are helper methods which verifies SDP.
506 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
507 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000508 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000509
510 // Check if a call to SetLocalDescription is acceptable with |action|.
511 bool ExpectSetLocalDescription(Action action);
512 // Check if a call to SetRemoteDescription is acceptable with |action|.
513 bool ExpectSetRemoteDescription(Action action);
514 // Verifies a=setup attribute as per RFC 5763.
515 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
516 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000517
jiayl@webrtc.orge10d28c2014-07-17 17:07:49 +0000518 // Returns true if we are ready to push down the remote candidate.
519 // |remote_desc| is the new remote description, or NULL if the current remote
520 // description should be used. Output |valid| is true if the candidate media
521 // index is valid.
522 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
523 const SessionDescriptionInterface* remote_desc,
524 bool* valid);
525
deadbeef7af91dd2016-12-13 11:29:11 -0800526 // Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
527 // this session.
528 bool SrtpRequired() const;
529
deadbeef67b3bbe2017-01-04 18:38:02 -0800530 // TransportController signal handlers.
deadbeefcbecd352015-09-23 11:50:27 -0700531 void OnTransportControllerConnectionState(cricket::IceConnectionState state);
532 void OnTransportControllerReceiving(bool receiving);
533 void OnTransportControllerGatheringState(cricket::IceGatheringState state);
534 void OnTransportControllerCandidatesGathered(
535 const std::string& transport_name,
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700536 const std::vector<cricket::Candidate>& candidates);
537 void OnTransportControllerCandidatesRemoved(
538 const std::vector<cricket::Candidate>& candidates);
deadbeef67b3bbe2017-01-04 18:38:02 -0800539 void OnTransportControllerDtlsHandshakeError(rtc::SSLHandshakeError error);
deadbeefcbecd352015-09-23 11:50:27 -0700540
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000541 std::string GetSessionErrorMsg();
542
deadbeefcbecd352015-09-23 11:50:27 -0700543 // Invoked when TransportController connection completion is signaled.
544 // Reports stats for all transports in use.
545 void ReportTransportStats();
546
547 // Gather the usage of IPv4/IPv6 as best connection.
jbauchac8869e2015-07-03 01:36:14 -0700548 void ReportBestConnectionState(const cricket::TransportStats& stats);
549
550 void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000551
Danil Chapovalov33b01f22016-05-11 19:55:27 +0200552 void OnSentPacket_w(const rtc::SentPacket& sent_packet);
stefanc1aeaf02015-10-15 07:26:07 -0700553
zhihuang9763d562016-08-05 11:14:50 -0700554 const std::string GetTransportName(const std::string& content_name);
555
556 rtc::Thread* const network_thread_;
deadbeefd59daf82015-10-14 15:02:44 -0700557 rtc::Thread* const worker_thread_;
danilchape9021a32016-05-17 01:52:02 -0700558 rtc::Thread* const signaling_thread_;
deadbeefd59daf82015-10-14 15:02:44 -0700559
560 State state_ = STATE_INIT;
561 Error error_ = ERROR_NONE;
562 std::string error_desc_;
563
564 const std::string sid_;
565 bool initial_offerer_ = false;
566
hbosdf6075a2016-12-19 04:58:02 -0800567 const std::unique_ptr<cricket::TransportController> transport_controller_;
deadbeef67b3bbe2017-01-04 18:38:02 -0800568 const std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory_;
stefanc1aeaf02015-10-15 07:26:07 -0700569 MediaControllerInterface* media_controller_;
kwibergd1fe2812016-04-27 06:47:29 -0700570 std::unique_ptr<cricket::VoiceChannel> voice_channel_;
571 std::unique_ptr<cricket::VideoChannel> video_channel_;
deadbeef67b3bbe2017-01-04 18:38:02 -0800572 // |rtp_data_channel_| is used if in RTP data channel mode, |sctp_transport_|
573 // when using SCTP.
574 std::unique_ptr<cricket::RtpDataChannel> rtp_data_channel_;
575
576 std::unique_ptr<cricket::SctpTransportInternal> sctp_transport_;
577 // |sctp_transport_name_| keeps track of what DTLS transport the SCTP
578 // transport is using (which can change due to bundling).
579 rtc::Optional<std::string> sctp_transport_name_;
580 // |sctp_content_name_| is the content name (MID) in SDP.
581 rtc::Optional<std::string> sctp_content_name_;
582 // Value cached on signaling thread. Only updated when SctpReadyToSendData
583 // fires on the signaling thread.
584 bool sctp_ready_to_send_data_ = false;
585 // Same as signals provided by SctpTransport, but these are guaranteed to
586 // fire on the signaling thread, whereas SctpTransport fires on the networking
587 // thread.
588 // |sctp_invoker_| is used so that any signals queued on the signaling thread
589 // from the network thread are immediately discarded if the SctpTransport is
590 // destroyed (due to m= section being rejected).
591 // TODO(deadbeef): Use a proxy object to ensure that method calls/signals
592 // are marshalled to the right thread. Could almost use proxy.h for this,
593 // but it doesn't have a mechanism for marshalling sigslot::signals
594 std::unique_ptr<rtc::AsyncInvoker> sctp_invoker_;
595 sigslot::signal1<bool> SignalSctpReadyToSendData;
596 sigslot::signal2<const cricket::ReceiveDataParams&,
597 const rtc::CopyOnWriteBuffer&>
598 SignalSctpDataReceived;
599 sigslot::signal1<int> SignalSctpStreamClosedRemotely;
600
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000601 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000602 IceObserver* ice_observer_;
603 PeerConnectionInterface::IceConnectionState ice_connection_state_;
Peter Thatcher54360512015-07-08 11:08:35 -0700604 bool ice_connection_receiving_;
deadbeeffe4a8a42016-12-20 17:56:17 -0800605 std::unique_ptr<SessionDescriptionInterface> current_local_description_;
606 std::unique_ptr<SessionDescriptionInterface> pending_local_description_;
607 std::unique_ptr<SessionDescriptionInterface> current_remote_description_;
608 std::unique_ptr<SessionDescriptionInterface> pending_remote_description_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000609 // If the remote peer is using a older version of implementation.
610 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000611 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000612 // Specifies which kind of data channel is allowed. This is controlled
613 // by the chrome command-line flag and constraints:
614 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
615 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
616 // not set or false, SCTP is allowed (DCT_SCTP);
617 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
618 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
zhihuang9763d562016-08-05 11:14:50 -0700619 // The data channel type could be DCT_QUIC if the QUIC data channel is
620 // enabled.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000621 cricket::DataChannelType data_channel_type_;
deadbeef0ed85b22016-02-23 17:24:52 -0800622 // List of content names for which the remote side triggered an ICE restart.
623 std::set<std::string> pending_ice_restarts_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000624
kwibergd1fe2812016-04-27 06:47:29 -0700625 std::unique_ptr<WebRtcSessionDescriptionFactory> webrtc_session_desc_factory_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000626
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000627 // Member variables for caching global options.
628 cricket::AudioOptions audio_options_;
629 cricket::VideoOptions video_options_;
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000630 MetricsObserverInterface* metrics_observer_;
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000631
pthatcher@webrtc.org877ac762015-02-04 22:03:09 +0000632 // Declares the bundle policy for the WebRTCSession.
633 PeerConnectionInterface::BundlePolicy bundle_policy_;
634
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700635 // Declares the RTCP mux policy for the WebRTCSession.
636 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
637
zhihuang184a3fd2016-06-14 11:47:14 -0700638 bool received_first_video_packet_ = false;
639 bool received_first_audio_packet_ = false;
640
zhihuang9763d562016-08-05 11:14:50 -0700641#ifdef HAVE_QUIC
642 std::unique_ptr<QuicDataTransport> quic_data_transport_;
643#endif // HAVE_QUIC
644
henrikg3c089d72015-09-16 05:37:44 -0700645 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
wu@webrtc.org364f2042013-11-20 21:49:41 +0000646};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000647} // namespace webrtc
648
Henrik Kjellander15583c12016-02-10 10:53:12 +0100649#endif // WEBRTC_API_WEBRTCSESSION_H_