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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
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 17:36:52 +000038#include "webrtc/base/sigslot.h"
39#include "webrtc/base/thread.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000040#include "talk/media/base/mediachannel.h"
41#include "talk/p2p/base/session.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000042#include "talk/session/media/mediasession.h"
43
44namespace cricket {
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000045
wu@webrtc.org364f2042013-11-20 21:49:41 +000046class BaseChannel;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000047class ChannelManager;
48class DataChannel;
49class StatsReport;
50class Transport;
51class VideoCapturer;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000052class VideoChannel;
53class VoiceChannel;
henrike@webrtc.orgb0ecc1c2014-03-26 22:44:28 +000054
henrike@webrtc.org28e20752013-07-10 00:45:36 +000055} // namespace cricket
56
57namespace webrtc {
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000058
henrike@webrtc.org28e20752013-07-10 00:45:36 +000059class IceRestartAnswerLatch;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000060class JsepIceCandidate;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000061class MediaStreamSignaling;
wu@webrtc.org91053e72013-08-10 07:18:04 +000062class WebRtcSessionDescriptionFactory;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000063
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000064extern const char kBundleWithoutRtcpMux[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000065extern const char kCreateChannelFailed[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000066extern const char kInvalidCandidates[];
67extern const char kInvalidSdp[];
68extern const char kMlineMismatch[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000069extern const char kPushDownTDFailed[];
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +000070extern const char kSdpWithoutDtlsFingerprint[];
71extern const char kSdpWithoutSdesCrypto[];
mallinath@webrtc.org19f27e62013-10-13 17:18:27 +000072extern const char kSdpWithoutIceUfragPwd[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000073extern const char kSdpWithoutSdesAndDtlsDisabled[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000074extern const char kSessionError[];
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000075extern const char kSessionErrorDesc[];
henrike@webrtc.org28e20752013-07-10 00:45:36 +000076
77// ICE state callback interface.
78class IceObserver {
79 public:
wu@webrtc.org364f2042013-11-20 21:49:41 +000080 IceObserver() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +000081 // Called any time the IceConnectionState changes
82 virtual void OnIceConnectionChange(
83 PeerConnectionInterface::IceConnectionState new_state) {}
84 // Called any time the IceGatheringState changes
85 virtual void OnIceGatheringChange(
86 PeerConnectionInterface::IceGatheringState new_state) {}
87 // New Ice candidate have been found.
88 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
89 // All Ice candidates have been found.
90 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
91 // (via PeerConnectionObserver)
92 virtual void OnIceComplete() {}
93
94 protected:
95 ~IceObserver() {}
wu@webrtc.org364f2042013-11-20 21:49:41 +000096
97 private:
98 DISALLOW_COPY_AND_ASSIGN(IceObserver);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000099};
100
101class WebRtcSession : public cricket::BaseSession,
102 public AudioProviderInterface,
103 public DataChannelFactory,
104 public VideoProviderInterface,
wu@webrtc.org78187522013-10-07 23:32:02 +0000105 public DtmfProviderInterface,
106 public DataChannelProviderInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000107 public:
108 WebRtcSession(cricket::ChannelManager* channel_manager,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000109 rtc::Thread* signaling_thread,
110 rtc::Thread* worker_thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000111 cricket::PortAllocator* port_allocator,
112 MediaStreamSignaling* mediastream_signaling);
113 virtual ~WebRtcSession();
114
wu@webrtc.org97077a32013-10-25 21:18:33 +0000115 bool Initialize(const PeerConnectionFactoryInterface::Options& options,
116 const MediaConstraintsInterface* constraints,
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000117 DTLSIdentityServiceInterface* dtls_identity_service,
118 PeerConnectionInterface::IceTransportsType ice_transport);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000119 // Deletes the voice, video and data channel and changes the session state
120 // to STATE_RECEIVEDTERMINATE.
121 void Terminate();
122
123 void RegisterIceObserver(IceObserver* observer) {
124 ice_observer_ = observer;
125 }
126
127 virtual cricket::VoiceChannel* voice_channel() {
128 return voice_channel_.get();
129 }
130 virtual cricket::VideoChannel* video_channel() {
131 return video_channel_.get();
132 }
133 virtual cricket::DataChannel* data_channel() {
134 return data_channel_.get();
135 }
136
henrike@webrtc.orgb90991d2014-03-04 19:54:57 +0000137 void SetSdesPolicy(cricket::SecurePolicy secure_policy);
138 cricket::SecurePolicy SdesPolicy() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000139
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000140 // Get current ssl role from transport.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000141 bool GetSslRole(rtc::SSLRole* role);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000142
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000143 // Generic error message callback from WebRtcSession.
144 // TODO - It may be necessary to supply error code as well.
145 sigslot::signal0<> SignalError;
146
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000147 void CreateOffer(
148 CreateSessionDescriptionObserver* observer,
149 const PeerConnectionInterface::RTCOfferAnswerOptions& options);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000150 void CreateAnswer(CreateSessionDescriptionObserver* observer,
151 const MediaConstraintsInterface* constraints);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000152 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000153 bool SetLocalDescription(SessionDescriptionInterface* desc,
154 std::string* err_desc);
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000155 // The ownership of |desc| will be transferred after this call.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000156 bool SetRemoteDescription(SessionDescriptionInterface* desc,
157 std::string* err_desc);
158 bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000159
160 bool UpdateIce(PeerConnectionInterface::IceTransportsType type);
161
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000162 const SessionDescriptionInterface* local_description() const {
163 return local_desc_.get();
164 }
165 const SessionDescriptionInterface* remote_description() const {
166 return remote_desc_.get();
167 }
168
169 // Get the id used as a media stream track's "id" field from ssrc.
xians@webrtc.org4cb01282014-06-12 14:57:05 +0000170 virtual bool GetLocalTrackIdBySsrc(uint32 ssrc, std::string* track_id);
171 virtual bool GetRemoteTrackIdBySsrc(uint32 ssrc, std::string* track_id);
172
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000173
174 // AudioMediaProviderInterface implementation.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000175 virtual void SetAudioPlayout(uint32 ssrc, bool enable,
176 cricket::AudioRenderer* renderer) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000177 virtual void SetAudioSend(uint32 ssrc, bool enable,
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000178 const cricket::AudioOptions& options,
179 cricket::AudioRenderer* renderer) OVERRIDE;
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000180 virtual void SetAudioPlayoutVolume(uint32 ssrc, double volume) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000181
182 // Implements VideoMediaProviderInterface.
183 virtual bool SetCaptureDevice(uint32 ssrc,
184 cricket::VideoCapturer* camera) OVERRIDE;
185 virtual void SetVideoPlayout(uint32 ssrc,
186 bool enable,
187 cricket::VideoRenderer* renderer) OVERRIDE;
188 virtual void SetVideoSend(uint32 ssrc, bool enable,
189 const cricket::VideoOptions* options) OVERRIDE;
190
191 // Implements DtmfProviderInterface.
192 virtual bool CanInsertDtmf(const std::string& track_id);
193 virtual bool InsertDtmf(const std::string& track_id,
194 int code, int duration);
195 virtual sigslot::signal0<>* GetOnDestroyedSignal();
196
wu@webrtc.org78187522013-10-07 23:32:02 +0000197 // Implements DataChannelProviderInterface.
198 virtual bool SendData(const cricket::SendDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000199 const rtc::Buffer& payload,
wu@webrtc.org78187522013-10-07 23:32:02 +0000200 cricket::SendDataResult* result) OVERRIDE;
201 virtual bool ConnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE;
202 virtual void DisconnectDataChannel(DataChannel* webrtc_data_channel) OVERRIDE;
wu@webrtc.orgcecfd182013-10-30 05:18:12 +0000203 virtual void AddSctpDataStream(uint32 sid) OVERRIDE;
wu@webrtc.orgcecfd182013-10-30 05:18:12 +0000204 virtual void RemoveSctpDataStream(uint32 sid) OVERRIDE;
wu@webrtc.org07a6fbe2013-11-04 18:41:34 +0000205 virtual bool ReadyToSendData() const OVERRIDE;
wu@webrtc.org78187522013-10-07 23:32:02 +0000206
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000207 // Implements DataChannelFactory.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000208 rtc::scoped_refptr<DataChannel> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000209 const std::string& label,
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000210 const InternalDataChannelInit* config) OVERRIDE;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000211
212 cricket::DataChannelType data_channel_type() const;
213
wu@webrtc.org91053e72013-08-10 07:18:04 +0000214 bool IceRestartPending() const;
215
216 void ResetIceRestartLatch();
217
218 // Called when an SSLIdentity is generated or retrieved by
219 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000220 void OnIdentityReady(rtc::SSLIdentity* identity);
wu@webrtc.org91053e72013-08-10 07:18:04 +0000221
222 // For unit test.
223 bool waiting_for_identity() const;
224
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000225 private:
226 // Indicates the type of SessionDescription in a call to SetLocalDescription
227 // and SetRemoteDescription.
228 enum Action {
229 kOffer,
230 kPrAnswer,
231 kAnswer,
232 };
wu@webrtc.org91053e72013-08-10 07:18:04 +0000233
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000234 // Invokes ConnectChannels() on transport proxies, which initiates ice
235 // candidates allocation.
236 bool StartCandidatesAllocation();
237 bool UpdateSessionState(Action action, cricket::ContentSource source,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000238 std::string* err_desc);
239 static Action GetAction(const std::string& type);
240
241 // Transport related callbacks, override from cricket::BaseSession.
242 virtual void OnTransportRequestSignaling(cricket::Transport* transport);
243 virtual void OnTransportConnecting(cricket::Transport* transport);
244 virtual void OnTransportWritable(cricket::Transport* transport);
mallinath@webrtc.org385857d2014-02-14 00:56:12 +0000245 virtual void OnTransportCompleted(cricket::Transport* transport);
246 virtual void OnTransportFailed(cricket::Transport* transport);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000247 virtual void OnTransportProxyCandidatesReady(
248 cricket::TransportProxy* proxy,
249 const cricket::Candidates& candidates);
250 virtual void OnCandidatesAllocationDone();
251
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000252 // Creates local session description with audio and video contents.
253 bool CreateDefaultLocalDescription();
254 // Enables media channels to allow sending of media.
255 void EnableChannels();
256 // Creates a JsepIceCandidate and adds it to the local session description
257 // and notify observers. Called when a new local candidate have been found.
258 void ProcessNewLocalCandidate(const std::string& content_name,
259 const cricket::Candidates& candidates);
260 // Returns the media index for a local ice candidate given the content name.
261 // Returns false if the local session description does not have a media
262 // content called |content_name|.
263 bool GetLocalCandidateMediaIndex(const std::string& content_name,
264 int* sdp_mline_index);
265 // Uses all remote candidates in |remote_desc| in this session.
266 bool UseCandidatesInSessionDescription(
267 const SessionDescriptionInterface* remote_desc);
268 // Uses |candidate| in this session.
269 bool UseCandidate(const IceCandidateInterface* candidate);
270 // Deletes the corresponding channel of contents that don't exist in |desc|.
271 // |desc| can be null. This means that all channels are deleted.
272 void RemoveUnusedChannelsAndTransports(
273 const cricket::SessionDescription* desc);
274
275 // Allocates media channels based on the |desc|. If |desc| doesn't have
276 // the BUNDLE option, this method will disable BUNDLE in PortAllocator.
277 // This method will also delete any existing media channels before creating.
278 bool CreateChannels(const cricket::SessionDescription* desc);
279
280 // Helper methods to create media channels.
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000281 bool CreateVoiceChannel(const cricket::ContentInfo* content);
282 bool CreateVideoChannel(const cricket::ContentInfo* content);
283 bool CreateDataChannel(const cricket::ContentInfo* content);
284
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000285 // Copy the candidates from |saved_candidates_| to |dest_desc|.
286 // The |saved_candidates_| will be cleared after this function call.
287 void CopySavedCandidates(SessionDescriptionInterface* dest_desc);
288
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000289 // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
290 // messages.
291 void OnDataChannelMessageReceived(cricket::DataChannel* channel,
292 const cricket::ReceiveDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000293 const rtc::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000294
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000295 std::string BadStateErrMsg(State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000296 void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
297
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000298 bool ValidateBundleSettings(const cricket::SessionDescription* desc);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000299 bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000300 // Below methods are helper methods which verifies SDP.
301 bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
302 cricket::ContentSource source,
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000303 std::string* err_desc);
sergeyu@chromium.org0be6aa02013-08-23 23:21:25 +0000304
305 // Check if a call to SetLocalDescription is acceptable with |action|.
306 bool ExpectSetLocalDescription(Action action);
307 // Check if a call to SetRemoteDescription is acceptable with |action|.
308 bool ExpectSetRemoteDescription(Action action);
309 // Verifies a=setup attribute as per RFC 5763.
310 bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
311 Action action);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000312
jiayl@webrtc.orge10d28c2014-07-17 17:07:49 +0000313 // Returns true if we are ready to push down the remote candidate.
314 // |remote_desc| is the new remote description, or NULL if the current remote
315 // description should be used. Output |valid| is true if the candidate media
316 // index is valid.
317 bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
318 const SessionDescriptionInterface* remote_desc,
319 bool* valid);
320
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000321 std::string GetSessionErrorMsg();
322
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000323 rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
324 rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
325 rtc::scoped_ptr<cricket::DataChannel> data_channel_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000326 cricket::ChannelManager* channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000327 MediaStreamSignaling* mediastream_signaling_;
328 IceObserver* ice_observer_;
329 PeerConnectionInterface::IceConnectionState ice_connection_state_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000330 rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
331 rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000332 // Candidates that arrived before the remote description was set.
333 std::vector<IceCandidateInterface*> saved_candidates_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000334 // If the remote peer is using a older version of implementation.
335 bool older_version_remote_peer_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000336 bool dtls_enabled_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000337 // Specifies which kind of data channel is allowed. This is controlled
338 // by the chrome command-line flag and constraints:
339 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
340 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
341 // not set or false, SCTP is allowed (DCT_SCTP);
342 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
343 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
344 cricket::DataChannelType data_channel_type_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000345 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000346
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000347 rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
wu@webrtc.org91053e72013-08-10 07:18:04 +0000348 webrtc_session_desc_factory_;
349
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000350 sigslot::signal0<> SignalVoiceChannelDestroyed;
351 sigslot::signal0<> SignalVideoChannelDestroyed;
352 sigslot::signal0<> SignalDataChannelDestroyed;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000353
henrike@webrtc.org6e3dbc22014-03-25 17:09:47 +0000354 // Member variables for caching global options.
355 cricket::AudioOptions audio_options_;
356 cricket::VideoOptions video_options_;
357
wu@webrtc.org364f2042013-11-20 21:49:41 +0000358 DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
359};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000360} // namespace webrtc
361
362#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_