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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2012 Google Inc. | 3 * Copyright 2012 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF | 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
26 */ | 26 */ |
27 | 27 |
28 #ifndef TALK_APP_WEBRTC_WEBRTCSESSION_H_ | 28 #ifndef TALK_APP_WEBRTC_WEBRTCSESSION_H_ |
29 #define TALK_APP_WEBRTC_WEBRTCSESSION_H_ | 29 #define TALK_APP_WEBRTC_WEBRTCSESSION_H_ |
30 | 30 |
31 #include <string> | 31 #include <string> |
32 | 32 |
33 #include "talk/app/webrtc/datachannel.h" | 33 #include "talk/app/webrtc/datachannel.h" |
34 #include "talk/app/webrtc/dtlscertificate.h" | |
34 #include "talk/app/webrtc/dtmfsender.h" | 35 #include "talk/app/webrtc/dtmfsender.h" |
35 #include "talk/app/webrtc/mediastreamprovider.h" | 36 #include "talk/app/webrtc/mediastreamprovider.h" |
36 #include "talk/app/webrtc/peerconnectioninterface.h" | 37 #include "talk/app/webrtc/peerconnectioninterface.h" |
37 #include "talk/app/webrtc/statstypes.h" | 38 #include "talk/app/webrtc/statstypes.h" |
38 #include "talk/media/base/mediachannel.h" | 39 #include "talk/media/base/mediachannel.h" |
39 #include "webrtc/p2p/base/session.h" | 40 #include "webrtc/p2p/base/session.h" |
40 #include "talk/session/media/mediasession.h" | 41 #include "talk/session/media/mediasession.h" |
41 #include "webrtc/base/sigslot.h" | 42 #include "webrtc/base/sigslot.h" |
42 #include "webrtc/base/sslidentity.h" | 43 #include "webrtc/base/sslidentity.h" |
43 #include "webrtc/base/thread.h" | 44 #include "webrtc/base/thread.h" |
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121 rtc::Thread* worker_thread, | 122 rtc::Thread* worker_thread, |
122 cricket::PortAllocator* port_allocator, | 123 cricket::PortAllocator* port_allocator, |
123 MediaStreamSignaling* mediastream_signaling); | 124 MediaStreamSignaling* mediastream_signaling); |
124 virtual ~WebRtcSession(); | 125 virtual ~WebRtcSession(); |
125 | 126 |
126 bool Initialize( | 127 bool Initialize( |
127 const PeerConnectionFactoryInterface::Options& options, | 128 const PeerConnectionFactoryInterface::Options& options, |
128 const MediaConstraintsInterface* constraints, | 129 const MediaConstraintsInterface* constraints, |
129 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, | 130 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, |
130 const PeerConnectionInterface::RTCConfiguration& rtc_configuration); | 131 const PeerConnectionInterface::RTCConfiguration& rtc_configuration); |
132 bool Initialize( | |
133 const PeerConnectionFactoryInterface::Options& options, | |
134 const MediaConstraintsInterface* constraints, | |
135 const rtc::scoped_refptr<webrtc::DtlsCertificate>& certificate, | |
136 const PeerConnectionInterface::RTCConfiguration& rtc_configuration); | |
131 // Deletes the voice, video and data channel and changes the session state | 137 // Deletes the voice, video and data channel and changes the session state |
132 // to STATE_RECEIVEDTERMINATE. | 138 // to STATE_RECEIVEDTERMINATE. |
133 void Terminate(); | 139 void Terminate(); |
134 | 140 |
135 void RegisterIceObserver(IceObserver* observer) { | 141 void RegisterIceObserver(IceObserver* observer) { |
136 ice_observer_ = observer; | 142 ice_observer_ = observer; |
137 } | 143 } |
138 | 144 |
139 virtual cricket::VoiceChannel* voice_channel() { | 145 virtual cricket::VoiceChannel* voice_channel() { |
140 return voice_channel_.get(); | 146 return voice_channel_.get(); |
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239 rtc::scoped_refptr<DataChannel> CreateDataChannel( | 245 rtc::scoped_refptr<DataChannel> CreateDataChannel( |
240 const std::string& label, | 246 const std::string& label, |
241 const InternalDataChannelInit* config) override; | 247 const InternalDataChannelInit* config) override; |
242 | 248 |
243 cricket::DataChannelType data_channel_type() const; | 249 cricket::DataChannelType data_channel_type() const; |
244 | 250 |
245 bool IceRestartPending() const; | 251 bool IceRestartPending() const; |
246 | 252 |
247 void ResetIceRestartLatch(); | 253 void ResetIceRestartLatch(); |
248 | 254 |
249 // Called when an SSLIdentity is generated or retrieved by | 255 // Called when a DtlsCertificate is generated or retrieved by |
250 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription. | 256 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription. |
251 void OnIdentityReady(rtc::SSLIdentity* identity); | 257 void OnCertificateReady( |
258 const rtc::scoped_refptr<DtlsCertificate>& certificate); | |
252 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp); | 259 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp); |
253 | 260 |
254 // For unit test. | 261 // For unit test. |
255 bool waiting_for_identity() const; | 262 bool IsWaitingForCertificate() const; |
263 rtc::scoped_refptr<DtlsCertificate> certificate() const { | |
tommi (sloooow) - chröme
2015/08/18 14:49:35
const scoped_refptr<>&
Can this be called safely
| |
264 return certificate_; | |
265 } | |
256 | 266 |
257 void set_metrics_observer( | 267 void set_metrics_observer( |
258 webrtc::MetricsObserverInterface* metrics_observer) { | 268 webrtc::MetricsObserverInterface* metrics_observer) { |
259 metrics_observer_ = metrics_observer; | 269 metrics_observer_ = metrics_observer; |
260 } | 270 } |
261 | 271 |
262 private: | 272 private: |
263 // Indicates the type of SessionDescription in a call to SetLocalDescription | 273 // Indicates the type of SessionDescription in a call to SetLocalDescription |
264 // and SetRemoteDescription. | 274 // and SetRemoteDescription. |
265 enum Action { | 275 enum Action { |
266 kOffer, | 276 kOffer, |
267 kPrAnswer, | 277 kPrAnswer, |
268 kAnswer, | 278 kAnswer, |
269 }; | 279 }; |
270 | 280 |
281 bool InitializeInternal( | |
282 const PeerConnectionFactoryInterface::Options& options, | |
283 const MediaConstraintsInterface* constraints, | |
284 const PeerConnectionInterface::RTCConfiguration& rtc_configuration); | |
285 void InitializeFactoryAfterConstruction( | |
286 const PeerConnectionFactoryInterface::Options& options); | |
287 | |
271 // Invokes ConnectChannels() on transport proxies, which initiates ice | 288 // Invokes ConnectChannels() on transport proxies, which initiates ice |
272 // candidates allocation. | 289 // candidates allocation. |
273 bool StartCandidatesAllocation(); | 290 bool StartCandidatesAllocation(); |
274 bool UpdateSessionState(Action action, cricket::ContentSource source, | 291 bool UpdateSessionState(Action action, cricket::ContentSource source, |
275 std::string* err_desc); | 292 std::string* err_desc); |
276 static Action GetAction(const std::string& type); | 293 static Action GetAction(const std::string& type); |
277 // Push the media parts of the local or remote session description | 294 // Push the media parts of the local or remote session description |
278 // down to all of the channels. | 295 // down to all of the channels. |
279 bool PushdownMediaDescription(cricket::ContentAction action, | 296 bool PushdownMediaDescription(cricket::ContentAction action, |
280 cricket::ContentSource source, | 297 cricket::ContentSource source, |
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386 bool dtls_enabled_; | 403 bool dtls_enabled_; |
387 // Specifies which kind of data channel is allowed. This is controlled | 404 // Specifies which kind of data channel is allowed. This is controlled |
388 // by the chrome command-line flag and constraints: | 405 // by the chrome command-line flag and constraints: |
389 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled, | 406 // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled, |
390 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is | 407 // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is |
391 // not set or false, SCTP is allowed (DCT_SCTP); | 408 // not set or false, SCTP is allowed (DCT_SCTP); |
392 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP); | 409 // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP); |
393 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE). | 410 // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE). |
394 cricket::DataChannelType data_channel_type_; | 411 cricket::DataChannelType data_channel_type_; |
395 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_; | 412 rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_; |
413 rtc::scoped_refptr<DtlsCertificate> certificate_; | |
396 | 414 |
397 rtc::scoped_ptr<WebRtcSessionDescriptionFactory> | 415 rtc::scoped_ptr<WebRtcSessionDescriptionFactory> |
398 webrtc_session_desc_factory_; | 416 webrtc_session_desc_factory_; |
399 | 417 |
400 sigslot::signal0<> SignalVoiceChannelDestroyed; | 418 sigslot::signal0<> SignalVoiceChannelDestroyed; |
401 sigslot::signal0<> SignalVideoChannelDestroyed; | 419 sigslot::signal0<> SignalVideoChannelDestroyed; |
402 sigslot::signal0<> SignalDataChannelDestroyed; | 420 sigslot::signal0<> SignalDataChannelDestroyed; |
403 | 421 |
404 // Member variables for caching global options. | 422 // Member variables for caching global options. |
405 cricket::AudioOptions audio_options_; | 423 cricket::AudioOptions audio_options_; |
406 cricket::VideoOptions video_options_; | 424 cricket::VideoOptions video_options_; |
407 MetricsObserverInterface* metrics_observer_; | 425 MetricsObserverInterface* metrics_observer_; |
408 | 426 |
409 // Declares the bundle policy for the WebRTCSession. | 427 // Declares the bundle policy for the WebRTCSession. |
410 PeerConnectionInterface::BundlePolicy bundle_policy_; | 428 PeerConnectionInterface::BundlePolicy bundle_policy_; |
411 | 429 |
412 // Declares the RTCP mux policy for the WebRTCSession. | 430 // Declares the RTCP mux policy for the WebRTCSession. |
413 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_; | 431 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_; |
414 | 432 |
415 DISALLOW_COPY_AND_ASSIGN(WebRtcSession); | 433 DISALLOW_COPY_AND_ASSIGN(WebRtcSession); |
416 }; | 434 }; |
417 } // namespace webrtc | 435 } // namespace webrtc |
418 | 436 |
419 #endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_ | 437 #endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_ |
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