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henrike@webrtc.org28e20752013-07-10 00:45:361/*
2 * libjingle
3 * Copyright 2012, Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#ifndef TALK_APP_WEBRTC_WEBRTCSESSION_H_
29#define TALK_APP_WEBRTC_WEBRTCSESSION_H_
30
31#include <string>
32
33#include "talk/app/webrtc/peerconnectioninterface.h"
34#include "talk/app/webrtc/dtmfsender.h"
35#include "talk/app/webrtc/mediastreamprovider.h"
36#include "talk/app/webrtc/datachannel.h"
37#include "talk/app/webrtc/statstypes.h"
buildbot@webrtc.orgd4e598d2014-07-29 17:36:5238#include "webrtc/base/sigslot.h"
39#include "webrtc/base/thread.h"
henrike@webrtc.org28e20752013-07-10 00:45:3640#include "talk/media/base/mediachannel.h"
41#include "talk/p2p/base/session.h"
henrike@webrtc.org28e20752013-07-10 00:45:3642#include "talk/session/media/mediasession.h"
43
44namespace cricket {
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:2845
wu@webrtc.org364f2042013-11-20 21:49:4146class BaseChannel;
henrike@webrtc.org28e20752013-07-10 00:45:3647class ChannelManager;
48class DataChannel;
49class StatsReport;
50class Transport;
51class VideoCapturer;
henrike@webrtc.org28e20752013-07-10 00:45:3652class VideoChannel;
53class VoiceChannel;
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:2854
henrike@webrtc.org28e20752013-07-10 00:45:3655} // namespace cricket
56
57namespace webrtc {
buildbot@webrtc.org41451d42014-05-03 05:39:4558
henrike@webrtc.org28e20752013-07-10 00:45:3659class IceRestartAnswerLatch;
buildbot@webrtc.org41451d42014-05-03 05:39:4560class JsepIceCandidate;
henrike@webrtc.org28e20752013-07-10 00:45:3661class MediaStreamSignaling;
wu@webrtc.org91053e72013-08-10 07:18:0462class WebRtcSessionDescriptionFactory;
henrike@webrtc.org28e20752013-07-10 00:45:3663
henrike@webrtc.org1e09a712013-07-26 19:17:5964extern const char kBundleWithoutRtcpMux[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:5465extern const char kCreateChannelFailed[];
henrike@webrtc.org28e20752013-07-10 00:45:3666extern const char kInvalidCandidates[];
67extern const char kInvalidSdp[];
68extern const char kMlineMismatch[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:5469extern const char kPushDownTDFailed[];
henrike@webrtc.orgb90991d2014-03-04 19:54:5770extern const char kSdpWithoutDtlsFingerprint[];
71extern const char kSdpWithoutSdesCrypto[];
mallinath@webrtc.org19f27e62013-10-13 17:18:2772extern const char kSdpWithoutIceUfragPwd[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:5473extern const char kSdpWithoutSdesAndDtlsDisabled[];
henrike@webrtc.org28e20752013-07-10 00:45:3674extern const char kSessionError[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:5475extern const char kSessionErrorDesc[];
buildbot@webrtc.org53df88c2014-08-07 22:46:0176// Maximum number of received video streams that will be processed by webrtc
77// even if they are not signalled beforehand.
78extern const int kMaxUnsignalledRecvStreams;
henrike@webrtc.org28e20752013-07-10 00:45:3679
80// ICE state callback interface.
81class IceObserver {
82 public:
wu@webrtc.org364f2042013-11-20 21:49:4183 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:3684 // Called any time the IceConnectionState changes
85 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;
92 // All Ice candidates have been found.
93 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
94 // (via PeerConnectionObserver)
95 virtual void OnIceComplete() {}
96
97 protected:
98 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:4199
100 private:
101 DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36102};
103
104class WebRtcSession : public cricket::BaseSession,
105 public AudioProviderInterface,
106 public DataChannelFactory,
107 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02108 public DtmfProviderInterface,
109 public DataChannelProviderInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36110 public:
111 WebRtcSession(cricket::ChannelManager* channel_manager,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52112 rtc::Thread* signaling_thread,
113 rtc::Thread* worker_thread,
henrike@webrtc.org28e20752013-07-10 00:45:36114 cricket::PortAllocator* port_allocator,
115 MediaStreamSignaling* mediastream_signaling);
116 virtual ~WebRtcSession();
117
wu@webrtc.org97077a32013-10-25 21:18:33118 bool Initialize(const PeerConnectionFactoryInterface::Options& options,
119 const MediaConstraintsInterface* constraints,
buildbot@webrtc.org41451d42014-05-03 05:39:45120 DTLSIdentityServiceInterface* dtls_identity_service,
121 PeerConnectionInterface::IceTransportsType ice_transport);
henrike@webrtc.org28e20752013-07-10 00:45:36122 // Deletes the voice, video and data channel and changes the session state
123 // to STATE_RECEIVEDTERMINATE.
124 void Terminate();
125
126 void RegisterIceObserver(IceObserver* observer) {
127 ice_observer_ = observer;
128 }
129
130 virtual cricket::VoiceChannel* voice_channel() {
131 return voice_channel_.get();
132 }
133 virtual cricket::VideoChannel* video_channel() {
134 return video_channel_.get();
135 }
136 virtual cricket::DataChannel* data_channel() {
137 return data_channel_.get();
138 }
139
henrike@webrtc.orgb90991d2014-03-04 19:54:57140 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
141 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36142
sergeyu@chromium.org0be6aa02013-08-23 23:21:25143 // Get current ssl role from transport.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52144 bool GetSslRole(rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25145
henrike@webrtc.org28e20752013-07-10 00:45:36146 // Generic error message callback from WebRtcSession.
147 // TODO - It may be necessary to supply error code as well.
148 sigslot::signal0<> SignalError;
149
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16150 void CreateOffer(
151 CreateSessionDescriptionObserver* observer,
152 const PeerConnectionInterface::RTCOfferAnswerOptions& options);
wu@webrtc.org91053e72013-08-10 07:18:04153 void CreateAnswer(CreateSessionDescriptionObserver* observer,
154 const MediaConstraintsInterface* constraints);
henrike@webrtc.org28654cb2013-07-22 21:07:49155 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36156 bool SetLocalDescription(SessionDescriptionInterface* desc,
157 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49158 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36159 bool SetRemoteDescription(SessionDescriptionInterface* desc,
160 std::string* err_desc);
161 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45162
163 bool UpdateIce(PeerConnectionInterface::IceTransportsType type);
164
henrike@webrtc.org28e20752013-07-10 00:45:36165 const SessionDescriptionInterface* local_description() const {
166 return local_desc_.get();
167 }
168 const SessionDescriptionInterface* remote_description() const {
169 return remote_desc_.get();
170 }
171
172 // Get the id used as a media stream track's "id" field from ssrc.
xians@webrtc.org4cb01282014-06-12 14:57:05173 virtual bool GetLocalTrackIdBySsrc(uint32 ssrc, std::string* track_id);
174 virtual bool GetRemoteTrackIdBySsrc(uint32 ssrc, std::string* track_id);
175
henrike@webrtc.org28e20752013-07-10 00:45:36176
177 // AudioMediaProviderInterface implementation.
henrike@webrtc.org1e09a712013-07-26 19:17:59178 virtual void SetAudioPlayout(uint32 ssrc, bool enable,
179 cricket::AudioRenderer* renderer) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36180 virtual void SetAudioSend(uint32 ssrc, bool enable,
henrike@webrtc.org1e09a712013-07-26 19:17:59181 const cricket::AudioOptions& options,
182 cricket::AudioRenderer* renderer) OVERRIDE;
wu@webrtc.orgb9a088b2014-02-13 23:18:49183 virtual void SetAudioPlayoutVolume(uint32 ssrc, double volume) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36184
185 // Implements VideoMediaProviderInterface.
186 virtual bool SetCaptureDevice(uint32 ssrc,
187 cricket::VideoCapturer* camera) OVERRIDE;
188 virtual void SetVideoPlayout(uint32 ssrc,
189 bool enable,
190 cricket::VideoRenderer* renderer) OVERRIDE;
191 virtual void SetVideoSend(uint32 ssrc, bool enable,
192 const cricket::VideoOptions* options) OVERRIDE;
193
194 // Implements DtmfProviderInterface.
195 virtual bool CanInsertDtmf(const std::string& track_id);
196 virtual bool InsertDtmf(const std::string& track_id,
197 int code, int duration);
198 virtual sigslot::signal0<>* GetOnDestroyedSignal();
199
wu@webrtc.org78187522013-10-07 23:32:02200 // Implements DataChannelProviderInterface.
201 virtual bool SendData(const cricket::SendDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52202 const rtc::Buffer& payload,
wu@webrtc.org78187522013-10-07 23:32:02203 cricket::SendDataResult* result) OVERRIDE;
204 virtual bool ConnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE;
205 virtual void DisconnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE;
wu@webrtc.orgcecfd182013-10-30 05:18:12206 virtual void AddSctpDataStream(uint32 sid) OVERRIDE;
wu@webrtc.orgcecfd182013-10-30 05:18:12207 virtual void RemoveSctpDataStream(uint32 sid) OVERRIDE;
wu@webrtc.org07a6fbe2013-11-04 18:41:34208 virtual bool ReadyToSendData() const OVERRIDE;
wu@webrtc.org78187522013-10-07 23:32:02209
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58210 // Implements DataChannelFactory.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52211 rtc::scoped_refptr<DataChannel> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36212 const std::string& label,
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58213 const InternalDataChannelInit* config) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36214
215 cricket::DataChannelType data_channel_type() const;
216
wu@webrtc.org91053e72013-08-10 07:18:04217 bool IceRestartPending() const;
218
219 void ResetIceRestartLatch();
220
221 // Called when an SSLIdentity is generated or retrieved by
222 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52223 void OnIdentityReady(rtc::SSLIdentity* identity);
wu@webrtc.org91053e72013-08-10 07:18:04224
225 // For unit test.
226 bool waiting_for_identity() const;
227
henrike@webrtc.org28e20752013-07-10 00:45:36228 private:
229 // Indicates the type of SessionDescription in a call to SetLocalDescription
230 // and SetRemoteDescription.
231 enum Action {
232 kOffer,
233 kPrAnswer,
234 kAnswer,
235 };
wu@webrtc.org91053e72013-08-10 07:18:04236
henrike@webrtc.org28e20752013-07-10 00:45:36237 // Invokes ConnectChannels() on transport proxies, which initiates ice
238 // candidates allocation.
239 bool StartCandidatesAllocation();
240 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36241 std::string* err_desc);
242 static Action GetAction(const std::string& type);
243
244 // Transport related callbacks, override from cricket::BaseSession.
245 virtual void OnTransportRequestSignaling(cricket::Transport* transport);
246 virtual void OnTransportConnecting(cricket::Transport* transport);
247 virtual void OnTransportWritable(cricket::Transport* transport);
mallinath@webrtc.org385857d2014-02-14 00:56:12248 virtual void OnTransportCompleted(cricket::Transport* transport);
249 virtual void OnTransportFailed(cricket::Transport* transport);
henrike@webrtc.org28e20752013-07-10 00:45:36250 virtual void OnTransportProxyCandidatesReady(
251 cricket::TransportProxy* proxy,
252 const cricket::Candidates& candidates);
253 virtual void OnCandidatesAllocationDone();
254
henrike@webrtc.org28e20752013-07-10 00:45:36255 // Creates local session description with audio and video contents.
256 bool CreateDefaultLocalDescription();
257 // Enables media channels to allow sending of media.
258 void EnableChannels();
259 // Creates a JsepIceCandidate and adds it to the local session description
260 // and notify observers. Called when a new local candidate have been found.
261 void ProcessNewLocalCandidate(const std::string& content_name,
262 const cricket::Candidates& candidates);
263 // Returns the media index for a local ice candidate given the content name.
264 // Returns false if the local session description does not have a media
265 // content called |content_name|.
266 bool GetLocalCandidateMediaIndex(const std::string& content_name,
267 int* sdp_mline_index);
268 // Uses all remote candidates in |remote_desc| in this session.
269 bool UseCandidatesInSessionDescription(
270 const SessionDescriptionInterface* remote_desc);
271 // Uses |candidate| in this session.
272 bool UseCandidate(const IceCandidateInterface* candidate);
273 // Deletes the corresponding channel of contents that don't exist in |desc|.
274 // |desc| can be null. This means that all channels are deleted.
275 void RemoveUnusedChannelsAndTransports(
276 const cricket::SessionDescription* desc);
277
278 // Allocates media channels based on the |desc|. If |desc| doesn't have
279 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
280 // This method will also delete any existing media channels before creating.
281 bool CreateChannels(const cricket::SessionDescription* desc);
282
283 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59284 bool CreateVoiceChannel(const cricket::ContentInfo* content);
285 bool CreateVideoChannel(const cricket::ContentInfo* content);
286 bool CreateDataChannel(const cricket::ContentInfo* content);
287
henrike@webrtc.org28e20752013-07-10 00:45:36288 // Copy the candidates from |saved_candidates_| to |dest_desc|.
289 // The |saved_candidates_| will be cleared after this function call.
290 void CopySavedCandidates(SessionDescriptionInterface* dest_desc);
291
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58292 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
293 // messages.
294 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
295 const cricket::ReceiveDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52296 const rtc::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36297
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54298 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36299 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
300
sergeyu@chromium.org0be6aa02013-08-23 23:21:25301 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59302 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25303 // Below methods are helper methods which verifies SDP.
304 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
305 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54306 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25307
308 // Check if a call to SetLocalDescription is acceptable with |action|.
309 bool ExpectSetLocalDescription(Action action);
310 // Check if a call to SetRemoteDescription is acceptable with |action|.
311 bool ExpectSetRemoteDescription(Action action);
312 // Verifies a=setup attribute as per RFC 5763.
313 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
314 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59315
jiayl@webrtc.orge10d28c2014-07-17 17:07:49316 // Returns true if we are ready to push down the remote candidate.
317 // |remote_desc| is the new remote description, or NULL if the current remote
318 // description should be used. Output |valid| is true if the candidate media
319 // index is valid.
320 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
321 const SessionDescriptionInterface* remote_desc,
322 bool* valid);
323
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54324 std::string GetSessionErrorMsg();
325
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52326 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
327 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
328 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36329 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36330 MediaStreamSignaling* mediastream_signaling_;
331 IceObserver* ice_observer_;
332 PeerConnectionInterface::IceConnectionState ice_connection_state_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52333 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
334 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36335 // Candidates that arrived before the remote description was set.
336 std::vector<IceCandidateInterface*> saved_candidates_;
henrike@webrtc.org28e20752013-07-10 00:45:36337 // If the remote peer is using a older version of implementation.
338 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10339 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36340 // Specifies which kind of data channel is allowed. This is controlled
341 // by the chrome command-line flag and constraints:
342 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
343 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
344 // not set or false, SCTP is allowed (DCT_SCTP);
345 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
346 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
347 cricket::DataChannelType data_channel_type_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52348 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04349
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52350 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04351 webrtc_session_desc_factory_;
352
henrike@webrtc.org28e20752013-07-10 00:45:36353 sigslot::signal0<> SignalVoiceChannelDestroyed;
354 sigslot::signal0<> SignalVideoChannelDestroyed;
355 sigslot::signal0<> SignalDataChannelDestroyed;
henrike@webrtc.org28e20752013-07-10 00:45:36356
henrike@webrtc.org6e3dbc22014-03-25 17:09:47357 // Member variables for caching global options.
358 cricket::AudioOptions audio_options_;
359 cricket::VideoOptions video_options_;
360
wu@webrtc.org364f2042013-11-20 21:49:41361 DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
362};
henrike@webrtc.org28e20752013-07-10 00:45:36363} // namespace webrtc
364
365#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_