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Issue 2324853002: Adding logs to track potential cause of not starting port allocation.
Patch Set: Created 4 years, 3 months ago
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1 /* 1 /*
2 * Copyright 2015 The WebRTC Project Authors. All rights reserved. 2 * Copyright 2015 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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448 ContentAction action, 448 ContentAction action,
449 std::string* err) { 449 std::string* err) {
450 RTC_DCHECK(network_thread_->IsCurrent()); 450 RTC_DCHECK(network_thread_->IsCurrent());
451 451
452 Transport* transport = GetTransport_n(transport_name); 452 Transport* transport = GetTransport_n(transport_name);
453 if (!transport) { 453 if (!transport) {
454 // If we didn't find a transport, that's not an error; 454 // If we didn't find a transport, that's not an error;
455 // it could have been deleted as a result of bundling. 455 // it could have been deleted as a result of bundling.
456 // TODO(deadbeef): Make callers smarter so they won't attempt to set a 456 // TODO(deadbeef): Make callers smarter so they won't attempt to set a
457 // description on a deleted transport. 457 // description on a deleted transport.
458 // This assumes we always bundle all media on the audio transport, so I
459 // only log a message here.
460 if (transport_name == CN_AUDIO) {
pthatcher1 2016/09/08 17:55:27 We don't always bundle on audio, and this would lo
honghaiz3 2016/09/08 21:30:11 If we are using data-channel only, then the transp
461 LOG(LS_ERROR) << "No transport is found for audio when setting "
462 << "local transport description";
463 }
458 return true; 464 return true;
459 } 465 }
460 466
461 // Older versions of Chrome expect the ICE role to be re-determined when an 467 // Older versions of Chrome expect the ICE role to be re-determined when an
462 // ICE restart occurs, and also don't perform conflict resolution correctly, 468 // ICE restart occurs, and also don't perform conflict resolution correctly,
463 // so for now we can't safely stop doing this, unless the application opts in 469 // so for now we can't safely stop doing this, unless the application opts in
464 // by setting |redetermine_role_on_ice_restart_| to false. 470 // by setting |redetermine_role_on_ice_restart_| to false.
465 // See: https://bugs.chromium.org/p/chromium/issues/detail?id=628676 471 // See: https://bugs.chromium.org/p/chromium/issues/detail?id=628676
466 // TODO(deadbeef): Remove this when these old versions of Chrome reach a low 472 // TODO(deadbeef): Remove this when these old versions of Chrome reach a low
467 // enough population. 473 // enough population.
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696 RTC_FROM_HERE, this, MSG_ICEGATHERINGSTATE, 702 RTC_FROM_HERE, this, MSG_ICEGATHERINGSTATE,
697 new rtc::TypedMessageData<IceGatheringState>(new_gathering_state)); 703 new rtc::TypedMessageData<IceGatheringState>(new_gathering_state));
698 } 704 }
699 } 705 }
700 706
701 void TransportController::OnDtlsHandshakeError(rtc::SSLHandshakeError error) { 707 void TransportController::OnDtlsHandshakeError(rtc::SSLHandshakeError error) {
702 SignalDtlsHandshakeError(error); 708 SignalDtlsHandshakeError(error);
703 } 709 }
704 710
705 } // namespace cricket 711 } // namespace cricket
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