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1 /* | 1 /* |
2 * Copyright 2004 The WebRTC Project Authors. All rights reserved. | 2 * Copyright 2004 The WebRTC Project Authors. All rights reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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312 | 312 |
313 // Destroys the channel with the given names. | 313 // Destroys the channel with the given names. |
314 // This will usually be called from the worker thread, but that | 314 // This will usually be called from the worker thread, but that |
315 // shouldn't be an issue since the main thread will be blocked in | 315 // shouldn't be an issue since the main thread will be blocked in |
316 // Send when doing so. | 316 // Send when doing so. |
317 virtual void DestroyChannel(const std::string& content_name, | 317 virtual void DestroyChannel(const std::string& content_name, |
318 int component); | 318 int component); |
319 | 319 |
320 rtc::SSLIdentity* identity() { return identity_; } | 320 rtc::SSLIdentity* identity() { return identity_; } |
321 | 321 |
| 322 // Set the ice connection receiving timeout. |
| 323 void SetIceConnectionReceivingTimeout(int timeout_ms); |
| 324 |
322 protected: | 325 protected: |
323 // Specifies the identity to use in this session. | 326 // Specifies the identity to use in this session. |
324 bool SetIdentity(rtc::SSLIdentity* identity); | 327 bool SetIdentity(rtc::SSLIdentity* identity); |
325 | 328 |
326 bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version); | 329 bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version); |
327 | 330 |
328 bool PushdownTransportDescription(ContentSource source, | 331 bool PushdownTransportDescription(ContentSource source, |
329 ContentAction action, | 332 ContentAction action, |
330 std::string* error_desc); | 333 std::string* error_desc); |
331 void set_initiator(bool initiator) { initiator_ = initiator; } | 334 void set_initiator(bool initiator) { initiator_ = initiator; } |
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451 bool initiator_; | 454 bool initiator_; |
452 rtc::SSLIdentity* identity_; | 455 rtc::SSLIdentity* identity_; |
453 rtc::SSLProtocolVersion ssl_max_version_; | 456 rtc::SSLProtocolVersion ssl_max_version_; |
454 rtc::scoped_ptr<const SessionDescription> local_description_; | 457 rtc::scoped_ptr<const SessionDescription> local_description_; |
455 rtc::scoped_ptr<SessionDescription> remote_description_; | 458 rtc::scoped_ptr<SessionDescription> remote_description_; |
456 uint64 ice_tiebreaker_; | 459 uint64 ice_tiebreaker_; |
457 // This flag will be set to true after the first role switch. This flag | 460 // This flag will be set to true after the first role switch. This flag |
458 // will enable us to stop any role switch during the call. | 461 // will enable us to stop any role switch during the call. |
459 bool role_switch_; | 462 bool role_switch_; |
460 TransportMap transports_; | 463 TransportMap transports_; |
| 464 |
| 465 // Timeout value in milliseconds for which no ICE connection receives |
| 466 // any packets. |
| 467 int ice_receiving_timeout_; |
461 }; | 468 }; |
462 | 469 |
463 } // namespace cricket | 470 } // namespace cricket |
464 | 471 |
465 #endif // WEBRTC_P2P_BASE_SESSION_H_ | 472 #endif // WEBRTC_P2P_BASE_SESSION_H_ |
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