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关于服务器连接不上的问题? #21

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xuandu opened this issue Sep 15, 2020 · 4 comments
Closed

关于服务器连接不上的问题? #21

xuandu opened this issue Sep 15, 2020 · 4 comments

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@xuandu
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xuandu commented Sep 15, 2020

您好:按照您书上说的,编译了Mac的demo,并下载配置docker。
Error Domain=NSURLErrorDomain Code=-1004 "Could not connect to the server." UserInfo={NSUnderlyingError=0x600000c52a00 {Error Domain=kCFErrorDomainCFNetwork Code=-1004 "(null)" UserInfo={_kCFStreamErrorCodeKey=61, _kCFStreamErrorDomainKey=1}}, NSErrorFailingURLStringKey=http://192.168.43.223:8080/join/11FE722DE93B4D398C66621463F73870?debug=loopback, NSErrorFailingURLKey=http://192.168.43.223:8080/join/11FE722DE93B4D398C66621463F73870?debug=loopback, _kCFStreamErrorDomainKey=1, _kCFStreamErrorCodeKey=61, NSLocalizedDescription=Could not connect to the server.}
本机做服务器,本机运行Macdemo循环测试;报错;使用网络为手机热点

控制台:
docker -v
Docker version 19.03.12, build 48a66213fe
docker run --rm -p 8080:8080 -p 8089:8089 -p 3033:3033 -p 3478:3478 -p 3478:3478/udp -p59000-65000:59000-65000/udp -e PUBLIC_IP=192.168.43.223 -it piasy/apprtc-server
docker: Error response from daemon: Ports are not available: listen udp 0.0.0.0:59962: bind: address already in use.
ERRO[0001] error waiting for container: context canceled

@Piasy
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Piasy commented Sep 15, 2020

使用网络为手机热点

通过手机热点,能连到 192.168.43.223 这个服务器么?

@xuandu
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xuandu commented Sep 15, 2020

我们公司网络限制,只能是手机热点共享给电脑,电脑的实时IP就是192.168.43.223;因为我的iOSdemo编译不成功,所以编译了Mac demo。也就是Mac回环,信令服务器也用的同一台电脑,都是192.168.43.223;至于您问的能连到吗?我不是很理解,因为我也只能是通过点击demo的start去尝试连接,我连接本地demo选择回环,点击start 然后就弹出了上边报错的窗口,并且小窗口显示client connecting;然后点击那个报错弹窗的OK后,小窗口变成client connected
这个是整个过程的控制台窗口的东西:

[123:743] [775] (audio_processing_impl.cc:438): Capture analyzer activated: 0
Capture post processor activated: 0
Render pre processor activated: 0
[123:743] [775] (webrtc_video_engine.cc:473): WebRtcVideoEngine::WebRtcVideoEngine()
[123:743] [775] (webrtc_voice_engine.cc:196): WebRtcVoiceEngine::WebRtcVoiceEngine
[123:744] [130355] (webrtc_voice_engine.cc:219): WebRtcVoiceEngine::Init
[123:744] [130355] (webrtc_voice_engine.cc:227): Supported send codecs in order of preference:
[123:744] [130355] (webrtc_voice_engine.cc:230): opus/48000/2 { minptime=10 useinbandfec=1 } (111)
[123:744] [130355] (webrtc_voice_engine.cc:230): ISAC/16000/1 (103)
[123:744] [130355] (webrtc_voice_engine.cc:230): ISAC/32000/1 (104)
[123:744] [130355] (webrtc_voice_engine.cc:230): G722/8000/1 (9)
[123:744] [130355] (webrtc_voice_engine.cc:230): ILBC/8000/1 (102)
[123:744] [130355] (webrtc_voice_engine.cc:230): PCMU/8000/1 (0)
[123:744] [130355] (webrtc_voice_engine.cc:230): PCMA/8000/1 (8)
[123:744] [130355] (webrtc_voice_engine.cc:230): CN/32000/1 (106)
[123:744] [130355] (webrtc_voice_engine.cc:230): CN/16000/1 (105)
[123:744] [130355] (webrtc_voice_engine.cc:230): CN/8000/1 (13)
[123:744] [130355] (webrtc_voice_engine.cc:230): telephone-event/48000/1 (110)
[123:744] [130355] (webrtc_voice_engine.cc:230): telephone-event/32000/1 (112)
[123:744] [130355] (webrtc_voice_engine.cc:230): telephone-event/16000/1 (113)
[123:758] [130355] (webrtc_voice_engine.cc:230): telephone-event/8000/1 (126)
[123:758] [130355] (webrtc_voice_engine.cc:233): Supported recv codecs in order of preference:
[123:758] [130355] (webrtc_voice_engine.cc:236): opus/48000/2 { minptime=10 useinbandfec=1 } (111)
[123:758] [130355] (webrtc_voice_engine.cc:236): ISAC/16000/1 (103)
[123:758] [130355] (webrtc_voice_engine.cc:236): ISAC/32000/1 (104)
[123:758] [130355] (webrtc_voice_engine.cc:236): G722/8000/1 (9)
[123:758] [130355] (webrtc_voice_engine.cc:236): ILBC/8000/1 (102)
[123:758] [130355] (webrtc_voice_engine.cc:236): PCMU/8000/1 (0)
[123:758] [130355] (webrtc_voice_engine.cc:236): PCMA/8000/1 (8)
[123:758] [130355] (webrtc_voice_engine.cc:236): CN/32000/1 (106)
[123:758] [130355] (webrtc_voice_engine.cc:236): CN/16000/1 (105)
[123:758] [130355] (webrtc_voice_engine.cc:236): CN/8000/1 (13)
[123:758] [130355] (webrtc_voice_engine.cc:236): telephone-event/48000/1 (110)
[123:758] [130355] (webrtc_voice_engine.cc:236): telephone-event/32000/1 (112)
[123:758] [130355] (webrtc_voice_engine.cc:236): telephone-event/16000/1 (113)
[123:758] [130355] (webrtc_voice_engine.cc:236): telephone-event/8000/1 (126)
[123:758] [130355] (audio_device_impl.cc:84): Create
[123:758] [130355] (audio_device_impl.cc:92): CreateForTest
[123:758] [130355] (audio_device_buffer.cc:63): AudioDeviceBuffer::ctor
[123:758] [130355] (audio_device_impl.cc:130): AudioDeviceModuleImpl
[123:758] [130355] (audio_device_impl.cc:134): CheckPlatform
[123:758] [130355] (audio_device_impl.cc:151): current platform is Mac
[123:758] [130355] (audio_device_impl.cc:164): CreatePlatformSpecificObjects
[123:758] [130355] (audio_device_impl.cc:946): PlatformAudioLayer
[123:758] [130355] (audio_mixer_manager_mac.cc:48): AudioMixerManagerMac created
[123:758] [130355] (audio_device_mac.cc:151): AudioDeviceMac created
[123:758] [130355] (audio_device_impl.cc:299): Mac OS X Audio APIs will be utilized.
[123:758] [130355] (audio_device_impl.cc:319): AttachAudioBuffer
[123:758] [130355] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000)
[123:758] [130355] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000)
[123:758] [130355] (audio_device_buffer.cc:200): SetRecordingChannels(1)
[123:758] [130355] (audio_device_buffer.cc:206): SetPlayoutChannels(2)
[123:758] [130355] (audio_device_impl.cc:339): Init
[123:758] [130355] (audio_device_impl.cc:683): SetPlayoutDevice(0)
[123:758] [130355] (audio_device_impl.cc:372): InitSpeaker
[123:759] [130355] (audio_device_mac.cc:1785): Output device: Apple Inc. Built-in Output
[123:759] [130355] (audio_device_impl.cc:587): StereoPlayoutIsAvailable
[123:759] [130355] (audio_mixer_manager_mac.cc:192): SpeakerIsInitialized
[123:759] [130355] (audio_device_impl.cc:594): output: 1
[123:759] [130355] (audio_device_impl.cc:599): SetStereoPlayout(1)
[123:759] [130355] (audio_device_buffer.cc:206): SetPlayoutChannels(2)
[123:759] [130355] (audio_device_impl.cc:745): SetRecordingDevice(0)
[123:759] [130355] (audio_device_impl.cc:378): InitMicrophone
[123:760] [130355] (audio_device_mac.cc:1783): Input device: Apple Inc. Built-in Microphone
[123:760] [130355] (audio_device_impl.cc:541): StereoRecordingIsAvailable
[123:760] [130355] (audio_mixer_manager_mac.cc:198): MicrophoneIsInitialized
[123:760] [130355] (audio_device_impl.cc:548): output: 1
[123:760] [130355] (audio_device_impl.cc:553): SetStereoRecording(1)
[123:760] [130355] (audio_device_buffer.cc:200): SetRecordingChannels(2)
[123:760] [130355] (apm_helpers.cc:32): Setting AGC mode to 0
[123:760] [130355] (audio_processing_impl.cc:727): Highpass filter activated: 0
[123:760] [130355] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0
[123:760] [130355] (audio_processing_impl.cc:747): Pre-amplifier activated: 0
[123:760] [130355] (audio_device_impl.cc:858): RegisterAudioCallback
[123:760] [130355] (audio_device_buffer.cc:81): RegisterAudioCallback
[123:760] [130355] (webrtc_voice_engine.cc:309): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 1, agc: 1, ns: 1, hf: 1, swap: 0, audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, typing: 1, experimental_agc: 0, extended_filter_aec: 0, delay_agnostic_aec: 0, experimental_ns: 0, residual_echo_detector: 1, }
[123:760] [130355] (audio_device_impl.cc:874): BuiltInAECIsAvailable
[123:760] [130355] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform
[123:760] [130355] (audio_device_impl.cc:877): output: 0
[123:761] [130355] (render_delay_buffer.cc:341): Applying total delay of 5 blocks.
[123:761] [130355] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms.
[123:761] [130355] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms.
[123:761] [130355] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms.
[123:761] [130355] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms.
[123:761] [130355] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms.
[123:761] [130355] (audio_processing_impl.cc:727): Highpass filter activated: 0
[123:761] [130355] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0
[123:761] [130355] (audio_processing_impl.cc:747): Pre-amplifier activated: 0
[123:761] [130355] (apm_helpers.cc:48): Echo control set to 1 with mode 0
[123:761] [130355] (audio_device_impl.cc:890): BuiltInAGCIsAvailable
[123:761] [130355] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform
[123:761] [130355] (audio_device_impl.cc:893): output: 0
[123:761] [130355] (audio_device_impl.cc:906): BuiltInNSIsAvailable
[123:761] [130355] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform
[123:761] [130355] (audio_device_impl.cc:909): output: 0
[123:761] [130355] (audio_processing_impl.cc:727): Highpass filter activated: 0
[123:762] [130355] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0
[123:762] [130355] (audio_processing_impl.cc:747): Pre-amplifier activated: 0
[123:762] [130355] (audio_processing_impl.cc:727): Highpass filter activated: 0
[123:762] [130355] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0
[123:762] [130355] (audio_processing_impl.cc:747): Pre-amplifier activated: 0
[123:762] [130355] (apm_helpers.cc:62): NS set to 1
[123:762] [130355] (webrtc_voice_engine.cc:447): Stereo swapping enabled? 0
[123:762] [130355] (webrtc_voice_engine.cc:452): NetEq capacity is 200
[123:762] [130355] (webrtc_voice_engine.cc:458): NetEq fast mode? 0
[123:762] [130355] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0
[123:762] [130355] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0
[123:762] [130355] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0
[123:762] [130355] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0
[123:762] [130355] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0
[123:762] [130355] (webrtc_voice_engine.cc:511): Setting AGC to 1
[123:762] [130355] (webrtc_voice_engine.cc:533): Typing detection is enabled? 1
[123:762] [130355] (audio_processing_impl.cc:727): Highpass filter activated: 1
[123:762] [130355] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0
[123:762] [130355] (audio_processing_impl.cc:747): Pre-amplifier activated: 0
[123:762] [775] (RTCLogging.mm:33): (ARDAppEngineClient.m:55 -[ARDAppEngineClient joinRoomWithRoomId:isLoopback:completionHandler:]): Joining room:5FE3DD410B2140FCAC5C00E35430CDF9 on room server.
2020-09-15 20:11:06.040557+0800 AppRTCMobile[45969:3134226] [] nw_socket_handle_socket_event [C3:3] Socket SO_ERROR [61: Connection refused]
2020-09-15 20:11:06.040704+0800 AppRTCMobile[45969:3134226] [BoringSSL] nw_protocol_boringssl_handshake_negotiate_proceed(726) [0x104748240] handshake failed at state 0
2020-09-15 20:11:06.040867+0800 AppRTCMobile[45969:3134226] [] nw_socket_handle_socket_event [C4:2] Socket SO_ERROR [61: Connection refused]
2020-09-15 20:11:06.041196+0800 AppRTCMobile[45969:3134226] Connection 3: received failure notification
2020-09-15 20:11:06.041229+0800 AppRTCMobile[45969:3134226] Connection 3: failed to connect 1:61, reason -1
2020-09-15 20:11:06.041259+0800 AppRTCMobile[45969:3134226] Connection 3: encountered error(1:61)
2020-09-15 20:11:06.041997+0800 AppRTCMobile[45969:3134226] Task <6D05E00D-CF2E-4FDA-AB80-5A6E30756762>.<3> HTTP load failed, 0/0 bytes (error code: -1004 [1:61])
2020-09-15 20:11:06.042120+0800 AppRTCMobile[45969:3134226] Connection 4: received failure notification
2020-09-15 20:11:06.042158+0800 AppRTCMobile[45969:3134226] Connection 4: failed to connect 1:61, reason -1
2020-09-15 20:11:06.042179+0800 AppRTCMobile[45969:3134226] Connection 4: encountered error(1:61)
2020-09-15 20:11:06.042279+0800 AppRTCMobile[45969:3134230] Task <6D05E00D-CF2E-4FDA-AB80-5A6E30756762>.<3> finished with error [-1004] Error Domain=NSURLErrorDomain Code=-1004 "Could not connect to the server." UserInfo={NSUnderlyingError=0x600000c3cfc0 {Error Domain=kCFErrorDomainCFNetwork Code=-1004 "(null)" UserInfo={_kCFStreamErrorCodeKey=61, _kCFStreamErrorDomainKey=1}}, NSErrorFailingURLStringKey=https://192.168.43.223:3033/iceconfig, NSErrorFailingURLKey=https://192.168.43.223:3033/iceconfig, _kCFStreamErrorDomainKey=1, _kCFStreamErrorCodeKey=61, NSLocalizedDescription=Could not connect to the server.}
[123:767] [16155] (RTCLogging.mm:33): (ARDAppClient.m:245 -[ARDAppClient connectToRoomWithId:settings:isLoopback:]_block_invoke): Error retrieving TURN servers: Could not connect to the server.
2020-09-15 20:11:06.043019+0800 AppRTCMobile[45969:3134226] Task .<4> HTTP load failed, 0/0 bytes (error code: -1004 [1:61])
2020-09-15 20:11:06.043228+0800 AppRTCMobile[45969:3134226] Task .<4> finished with error [-1004] Error Domain=NSURLErrorDomain Code=-1004 "Could not connect to the server." UserInfo={NSUnderlyingError=0x600000c3c540 {Error Domain=kCFErrorDomainCFNetwork Code=-1004 "(null)" UserInfo={_kCFStreamErrorCodeKey=61, _kCFStreamErrorDomainKey=1}}, NSErrorFailingURLStringKey=http://192.168.43.223:8080/join/5FE3DD410B2140FCAC5C00E35430CDF9?debug=loopback, NSErrorFailingURLKey=http://192.168.43.223:8080/join/5FE3DD410B2140FCAC5C00E35430CDF9?debug=loopback, _kCFStreamErrorDomainKey=1, _kCFStreamErrorCodeKey=61, NSLocalizedDescription=Could not connect to the server.}
[123:768] [130355] (webrtc_video_engine.cc:477): WebRtcVideoEngine::~WebRtcVideoEngine
[123:768] [130355] (webrtc_voice_engine.cc:205): WebRtcVoiceEngine::~WebRtcVoiceEngine
[123:768] [130355] (audio_device_impl.cc:809): StopPlayout
[123:768] [130355] (audio_device_mac.cc:1449): StopPlayout
[123:768] [130355] (audio_device_impl.cc:813): output: 0
[123:768] [130355] (audio_device_impl.cc:840): StopRecording
[123:768] [130355] (audio_device_mac.cc:1318): StopRecording
[123:768] [130355] (audio_device_impl.cc:844): output: 0
[123:768] [130355] (audio_device_impl.cc:858): RegisterAudioCallback
[123:768] [130355] (audio_device_buffer.cc:81): RegisterAudioCallback
[123:768] [130355] (audio_device_impl.cc:356): Terminate
[123:770] [130355] (audio_device_impl.cc:325): ~AudioDeviceModuleImpl
[123:770] [130355] (audio_device_mac.cc:161): ~AudioDeviceMac destroyed
[123:770] [130355] (audio_mixer_manager_mac.cc:52): ~AudioMixerManagerMac destroyed
[123:770] [130355] (audio_device_buffer.cc:75): AudioDeviceBuffer::~dtor

@xuandu
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xuandu commented Sep 23, 2020

你好,我现在使用Mac运行你书上的docker;并关闭防火墙;安卓demo去连接我的机器IP,但是我这个docker启动的时候run --rm -p 8080:8080 -p 8089:8089 -p 3033:3033 -p 3478:3478 -p 3478:3478/udp -p 59000-65000:59000-65000/udp -e PUBLIC_IP=192.168.43.223 -it piasy/apprtc-server
出现了下边的问题
docker: Error response from daemon: dial unix docker.raw.sock: connect: connection refused.
See 'docker run --help'.

@Piasy
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Piasy commented Sep 23, 2020

重启下 docker?

@Piasy Piasy closed this as completed Jan 22, 2021
Piasy pushed a commit that referenced this issue Jun 20, 2021
This reverts commit 76d3e7a.

Reason for revert: Causes multiple Chromium WPT tests to crash, preventing rolls.

Sample failed run:
https://ci.chromium.org/p/chromium/builders/try/win10_chromium_x64_rel_ng/685757?

Sample stack trace:
#0 0x7ff8623fbde9 base::debug::CollectStackTrace()
STDERR: #1 0x7ff862311ca3 [2665012:17:1009/162250.249660:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652370324, interval (us) = 834
STDERR: base::debug::StackTrace::StackTrace()
STDERR: #2 0x7ff8623fb93b base::debug::(anonymous namespace)::StackDumpSignalHandler()
STDERR: #3 0x7ff857a70140 [2665012:17:1009/162250.249947:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652370634, interval (us) = 742
STDERR: (/lib/x86_64-linux-gnu/libpthread-2.31.so+0x1413f)
STDERR: #4 0x7ff85778edb1 gsignal
STDERR: #5 0x7ff857778537 abort
STDERR: #6 0x7ff855d5eee2 [2665012:17:1009/162250.250342:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652371030, interval (us) = 706
STDERR: [2665012:17:1009/162250.250514:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652371204, interval (us) = 963
STDERR: rtc::webrtc_checks_impl::FatalLog()
STDERR: #7 0x7ff855f14e62 webrtc::LibvpxVp8Encoder::PrepareRawImagesForEncoding()
STDERR: #8 0x7ff855f14412 webrtc::LibvpxVp8Encoder::Encode()
STDERR: #9 0x7ff855bae765 webrtc::SimulcastEncoderAdapter::Encode()
STDERR: #10 0x7ff85607d598 webrtc::VideoStreamEncoder::EncodeVideoFrame()
STDERR: #11 0x7ff85607c60d webrtc::VideoStreamEncoder::MaybeEncodeVideoFrame()
STDERR: #12 0x7ff8560816f5 webrtc::webrtc_new_closure_impl::ClosureTask<>::Run()
STDERR: #13 0x7ff855b352b5 (anonymous namespace)::WebrtcTaskQueue::RunTask()
STDERR: #14 0x7ff855b3531e base::internal::Invoker<>::RunOnce()
STDERR: #15 0x7ff86239785b base::TaskAnnotator::RunTask()
STDERR: #16 0x7ff8623c8557 base::internal::TaskTracker::RunSkipOnShutdown()
STDERR: #17 0x7ff8623c7d92 base::internal::TaskTracker::RunTask()
STDERR: #18 0x7ff862415a06 base::internal::TaskTrackerPosix::RunTask()
STDERR: #19 0x7ff8623c75e6 base::internal::TaskTracker::RunAndPopNextTask()
STDERR: #20 0x7ff8623d3a8d base::internal::WorkerThread::RunWorker()
STDERR: #21 0x7ff8623d368d base::internal::WorkerThread::RunPooledWorker()
STDERR: #22 0x7ff862416509 base::(anonymous namespace)::ThreadFunc()
STDERR: #23 0x7ff857a64ea7 start_thread 

Original change's description:
> NV12 support for VP8 simulcast
>
> Tested using video_loopback with generated NV12 frames.
>
> Bug: webrtc:11635, webrtc:11975
> Change-Id: I14b2d663c55a83d80e48e226fcf706cb18903193
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186722
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32325}

TBR=ilnik@webrtc.org,eshr@google.com

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11635
Bug: webrtc:11975
Change-Id: I61c8aed1270bc9c2f7f0577fa5ca717a325f548a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187484
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32369}
Piasy pushed a commit that referenced this issue Jun 20, 2021
This reverts commit c5f7108.

Reason for revert: Causes Chromium WPT Tests to fail, preventing rolls.

Sample failed run:
https://ci.chromium.org/p/chromium/builders/try/linux-rel/511995?

Sample logs:

STDERR: # Fatal error in: ../../third_party/webrtc/pc/peer_connection.cc, line 575
STDERR: # last system error: 0
STDERR: # Check failed: (signaling_thread())->IsCurrent()
STDERR: # Received signal 6
STDERR: #0 0x7f81d39e3de9 base::debug::CollectStackTrace()
STDERR: #1 0x7f81d38f9ca3 base::debug::StackTrace::StackTrace()
STDERR: #2 0x7f81d39e393b base::debug::(anonymous namespace)::StackDumpSignalHandler()
STDERR: #3 0x7f81c9054140 (/lib/x86_64-linux-gnu/libpthread-2.31.so+0x1413f)
STDERR: #4 0x7f81c8d72db1 gsignal
STDERR: #5 0x7f81c8d5c537 abort
STDERR: #6 0x7f81c7344032 rtc::webrtc_checks_impl::FatalLog()
STDERR: #7 0x7f81c722e5c0 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>()
STDERR: #8 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach()
STDERR: #9 0x7f81c72d6e8e webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>()
STDERR: #10 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach()
STDERR: #11 0x7f81c71c6df3 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>()
STDERR: #12 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach()
STDERR: #13 0x7f81c73135bc rtc::OpenSSLStreamAdapter::ContinueSSL()
STDERR: #14 0x7f81c7312fd4 rtc::OpenSSLStreamAdapter::OnEvent()
STDERR: #15 0x7f81c71c30d9 cricket::StreamInterfaceChannel::OnPacketReceived()
STDERR: #16 0x7f81c71c640a cricket::DtlsTransport::OnReadPacket()
STDERR: #17 0x7f81c71cad61 cricket::P2PTransportChannel::OnReadPacket()
STDERR: #18 0x7f81c71bc90f cricket::Connection::OnReadPacket()
STDERR: #19 0x7f81c71e6255 cricket::UDPPort::HandleIncomingPacket()
STDERR: #20 0x7f81cd1f1bff blink::(anonymous namespace)::IpcPacketSocket::OnDataReceived()
STDERR: #21 0x7f81cd1f645d blink::P2PSocketClientImpl::DataReceived()
STDERR: #22 0x7f81cd50a16b network::mojom::blink::P2PSocketClientStubDispatch::Accept()
STDERR: #23 0x7f81d2b4f642 mojo::InterfaceEndpointClient::HandleValidatedMessage()
STDERR: #24 0x7f81d2b5304b mojo::MessageDispatcher::Accept()
STDERR: #25 0x7f81d2b50bb1 mojo::InterfaceEndpointClient::HandleIncomingMessage()
STDERR: #26 0x7f81d2b58a3a mojo::internal::MultiplexRouter::ProcessIncomingMessage()
STDERR: #27 0x7f81d2b57f7f mojo::internal::MultiplexRouter::Accept()
STDERR: #28 0x7f81d2b5304b mojo::MessageDispatcher::Accept()
STDERR: #29 0x7f81d2b48851 mojo::Connector::DispatchMessage()
STDERR: #30 0x7f81d2b492e7 mojo::Connector::ReadAllAvailableMessages()
STDERR: #31 0x7f81d2b490a3 mojo::Connector::OnHandleReadyInternal()
STDERR: #32 0x7f81d2b498f0 mojo::SimpleWatcher::DiscardReadyState()
STDERR: #33 0x7f81d2d0e67a mojo::SimpleWatcher::OnHandleReady()
STDERR: #34 0x7f81d2d0eb2b base::internal::Invoker<>::RunOnce()
STDERR: #35 0x7f81d397f85b base::TaskAnnotator::RunTask()
STDERR: #36 0x7f81d399a04c base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWorkImpl()
STDERR: #37 0x7f81d3999c78 base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWork()
STDERR: #38 0x7f81d391fe64 base::MessagePumpDefault::Run()
STDERR: #39 0x7f81d399a8dc base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::Run()
STDERR: #40 0x7f81d395ae55 base::RunLoop::Run()
STDERR: #41 0x7f81d39c87f2 base::Thread::Run()




Original change's description:
> Reland "Replace sigslot usages with robocaller library."
>
> This is a reland of 40261c3
>
> Note: Instead of changing the type of JsepTransportController->SignalSSLHandshakeError
> added a new member with a different name and used it in webrtc code.
> After this change do two more follow up CLs to completely remove the old code
> from google3.
>
> Original change's description:
> > Replace sigslot usages with robocaller library.
> >
> > - Replace all the top level signals from jsep_transport_controller.
> > - There are still sigslot usages in this file so keep the inheritance
> >   and that is the reason for not having a binary size gain in this CL.
> >
> > Bug: webrtc:11943
> > Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
> > Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32321}
>
> Bug: webrtc:11943
> Change-Id: Ia07394ee395f94836f6b576c3a97d119a7678e1a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186946
> Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32359}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,glahiru@webrtc.org

Change-Id: I6bce1775d10758ac4a9d05b855f473dd70bd9815
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187487
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32372}
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