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  1. ean definitions from ServletContext resource [/WEB-INF/red5-web.xml]
  2. [INFO] [Loader:/oflaDemo] org.springframework.beans.factory.config.PropertyPlaceholderConfigurer - Loading properties file from ServletContext resource [/WEB-INF/red5-web.properties]
  3. [INFO] [Loader:/oflaDemo] org.red5.server.Context - Setting parent bean factory as core
  4. [INFO] [Loader:/oflaDemo] org.red5.server.scope.WebScope - Set server [Server@34a6b51amap['/chat' -> 'default', '/live' -> 'default', '192.168.1.2/WebRTCApp' -> 'default', '/vod' -> 'default', '/' -> 'default', 'localhost:5080/installer' -> 'default', 'localhost/installer' -> 'default', '127.0.0.1/WebRTCApp' -> 'default', 'localhost/WebRTCApp' -> 'default']]
  5. [INFO] [Loader:/oflaDemo] org.red5.server.Server - Add mapping global: default host: context: oflaDemo
  6. oflaDemo appStart
  7. [WARN] [NioProcessor-18] org.red5.net.websocket.codec.WebSocketDecoder - Handshake failed
  8. org.red5.net.websocket.WebSocketException: Handshake failed, path not enabled
  9. at org.red5.net.websocket.codec.WebSocketDecoder.parseClientRequest(WebSocketDecoder.java:279)
  10. at org.red5.net.websocket.codec.WebSocketDecoder.doHandShake(WebSocketDecoder.java:155)
  11. at org.red5.net.websocket.codec.WebSocketDecoder.doDecode(WebSocketDecoder.java:99)
  12. at org.apache.mina.filter.codec.CumulativeProtocolDecoder.decode(CumulativeProtocolDecoder.java:181)
  13. at org.apache.mina.filter.codec.ProtocolCodecFilter.messageReceived(ProtocolCodecFilter.java:231)
  14. at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:542)
  15. at org.apache.mina.core.filterchain.DefaultIoFilterChain.access$1300(DefaultIoFilterChain.java:48)
  16. at org.apache.mina.core.filterchain.DefaultIoFilterChain$EntryImpl$1.messageReceived(DefaultIoFilterChain.java:947)
  17. at org.apache.mina.core.filterchain.IoFilterAdapter.messageReceived(IoFilterAdapter.java:109)
  18. at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:542)
  19. at org.apache.mina.core.filterchain.DefaultIoFilterChain.fireMessageReceived(DefaultIoFilterChain.java:535)
  20. at org.apache.mina.core.polling.AbstractPollingIoProcessor.read(AbstractPollingIoProcessor.java:703)
  21. at org.apache.mina.core.polling.AbstractPollingIoProcessor.process(AbstractPollingIoProcessor.java:659)
  22. at org.apache.mina.core.polling.AbstractPollingIoProcessor.process(AbstractPollingIoProcessor.java:648)
  23. at org.apache.mina.core.polling.AbstractPollingIoProcessor.access$600(AbstractPollingIoProcessor.java:68)
  24. at org.apache.mina.core.polling.AbstractPollingIoProcessor$Processor.run(AbstractPollingIoProcessor.java:1120)
  25. at org.apache.mina.util.NamePreservingRunnable.run(NamePreservingRunnable.java:64)
  26. at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1149)
  27. at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:624)
  28. at java.lang.Thread.run(Thread.java:748)
  29. [WARN] [NioProcessor-19] org.red5.net.websocket.codec.WebSocketDecoder - Handshake failed
  30. org.red5.net.websocket.WebSocketException: Handshake failed, path not enabled
  31. at org.red5.net.websocket.codec.WebSocketDecoder.parseClientRequest(WebSocketDecoder.java:279)
  32. at org.red5.net.websocket.codec.WebSocketDecoder.doHandShake(WebSocketDecoder.java:155)
  33. at org.red5.net.websocket.codec.WebSocketDecoder.doDecode(WebSocketDecoder.java:99)
  34. at org.apache.mina.filter.codec.CumulativeProtocolDecoder.decode(CumulativeProtocolDecoder.java:181)
  35. at org.apache.mina.filter.codec.ProtocolCodecFilter.messageReceived(ProtocolCodecFilter.java:231)
  36. at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:542)
  37. at org.apache.mina.core.filterchain.DefaultIoFilterChain.access$1300(DefaultIoFilterChain.java:48)
  38. at org.apache.mina.core.filterchain.DefaultIoFilterChain$EntryImpl$1.messageReceived(DefaultIoFilterChain.java:947)
  39. at org.apache.mina.core.filterchain.IoFilterAdapter.messageReceived(IoFilterAdapter.java:109)
  40. at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:542)
  41. at org.apache.mina.core.filterchain.DefaultIoFilterChain.fireMessageReceived(DefaultIoFilterChain.java:535)
  42. at org.apache.mina.core.polling.AbstractPollingIoProcessor.read(AbstractPollingIoProcessor.java:703)
  43. at org.apache.mina.core.polling.AbstractPollingIoProcessor.process(AbstractPollingIoProcessor.java:659)
  44. at org.apache.mina.core.polling.AbstractPollingIoProcessor.process(AbstractPollingIoProcessor.java:648)
  45. at org.apache.mina.core.polling.AbstractPollingIoProcessor.access$600(AbstractPollingIoProcessor.java:68)
  46. at org.apache.mina.core.polling.AbstractPollingIoProcessor$Processor.run(AbstractPollingIoProcessor.java:1120)
  47. at org.apache.mina.util.NamePreservingRunnable.run(NamePreservingRunnable.java:64)
  48. at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1149)
  49. at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:624)
  50. at java.lang.Thread.run(Thread.java:748)
  51. [WARN] [NioProcessor-20] org.red5.net.websocket.codec.WebSocketDecoder - Handshake failed
  52. org.red5.net.websocket.WebSocketException: Handshake failed, path not enabled
  53. at org.red5.net.websocket.codec.WebSocketDecoder.parseClientRequest(WebSocketDecoder.java:279)
  54. at org.red5.net.websocket.codec.WebSocketDecoder.doHandShake(WebSocketDecoder.java:155)
  55. at org.red5.net.websocket.codec.WebSocketDecoder.doDecode(WebSocketDecoder.java:99)
  56. at org.apache.mina.filter.codec.CumulativeProtocolDecoder.decode(CumulativeProtocolDecoder.java:181)
  57. at org.apache.mina.filter.codec.ProtocolCodecFilter.messageReceived(ProtocolCodecFilter.java:231)
  58. at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:542)
  59. at org.apache.mina.core.filterchain.DefaultIoFilterChain.access$1300(DefaultIoFilterChain.java:48)
  60. at org.apache.mina.core.filterchain.DefaultIoFilterChain$EntryImpl$1.messageReceived(DefaultIoFilterChain.java:947)
  61. at org.apache.mina.core.filterchain.IoFilterAdapter.messageReceived(IoFilterAdapter.java:109)
  62. at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:542)
  63. at org.apache.mina.core.filterchain.DefaultIoFilterChain.fireMessageReceived(DefaultIoFilterChain.java:535)
  64. at org.apache.mina.core.polling.AbstractPollingIoProcessor.read(AbstractPollingIoProcessor.java:703)
  65. at org.apache.mina.core.polling.AbstractPollingIoProcessor.process(AbstractPollingIoProcessor.java:659)
  66. at org.apache.mina.core.polling.AbstractPollingIoProcessor.process(AbstractPollingIoProcessor.java:648)
  67. at org.apache.mina.core.polling.AbstractPollingIoProcessor.access$600(AbstractPollingIoProcessor.java:68)
  68. at org.apache.mina.core.polling.AbstractPollingIoProcessor$Processor.run(AbstractPollingIoProcessor.java:1120)
  69. at org.apache.mina.util.NamePreservingRunnable.run(NamePreservingRunnable.java:64)
  70. at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1149)
  71. at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:624)
  72. at java.lang.Thread.run(Thread.java:748)
  73. [INFO] [NioProcessor-21] io.antmedia.webrtc.WebSocketListener - onWSMessage: {"command":"publish","streamName":"stream1"}
  74.  
  75. Output #0, flv, to 'rtmp://127.0.0.1/WebRTCApp/stream1':
  76. Stream #0:0: Unknown: none
  77. Stream #0:1: Unknown: none
  78. [libx264 @ 0x7f70e0b68e40] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2
  79. [libx264 @ 0x7f70e0b68e40] profile High, level 3.0
  80. [libx264 @ 0x7f70e0b68e40] 264 - core 148 - H.264/MPEG-4 AVC codec - Copyleft 2003-2016 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=1 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=400 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
  81. [libx264 @ 0x7f70e0b68e40] final ratefactor: 33.12
  82. [aac @ 0x7f70e0b6b160] Qavg: -nan
  83. io.antmedia.webrtc.receiver.FrameRecorder$Exception: avio_open2 error() error -5: Could not open 'null'
  84. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.startUnsafe(FFmpegFrameRecorder.java:815)
  85. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.start(FFmpegFrameRecorder.java:364)
  86. at io.antmedia.webrtc.WebSocketListener.getNewRecorder(WebSocketListener.java:282)
  87. at io.antmedia.webrtc.WebSocketListener.takeAction(WebSocketListener.java:96)
  88. at io.antmedia.webrtc.WebSocketListener.onWSMessage(WebSocketListener.java:245)
  89. at org.red5.net.websocket.WebSocketScope.onMessage(WebSocketScope.java:234)
  90. at org.red5.net.websocket.WebSocketConnection.receive(WebSocketConnection.java:110)
  91. at org.red5.net.websocket.WebSocketHandler.messageReceived(WebSocketHandler.java:51)
  92. at org.apache.mina.core.filterchain.DefaultIoFilterChain$TailFilter.messageReceived(DefaultIoFilterChain.java:858)
  93. at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:542)
  94. at org.apache.mina.core.filterchain.DefaultIoFilterChain.access$1300(DefaultIoFilterChain.java:48)
  95. at org.apache.mina.core.filterchain.DefaultIoFilterChain$EntryImpl$1.messageReceived(DefaultIoFilterChain.java:947)
  96. at org.apache.mina.filter.codec.ProtocolCodecFilter$ProtocolDecoderOutputImpl.flush(ProtocolCodecFilter.java:398)
  97. at org.apache.mina.filter.codec.ProtocolCodecFilter.messageReceived(ProtocolCodecFilter.java:234)
  98. at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:542)
  99. at org.apache.mina.core.filterchain.DefaultIoFilterChain.access$1300(DefaultIoFilterChain.java:48)
  100. at org.apache.mina.core.filterchain.DefaultIoFilterChain$EntryImpl$1.messageReceived(DefaultIoFilterChain.java:947)
  101. at org.apache.mina.core.filterchain.IoFilterAdapter.messageReceived(IoFilterAdapter.java:109)
  102. at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:542)
  103. at org.apache.mina.core.filterchain.DefaultIoFilterChain.fireMessageReceived(DefaultIoFilterChain.java:535)
  104. at org.apache.mina.core.polling.AbstractPollingIoProcessor.read(AbstractPollingIoProcessor.java:703)
  105. at org.apache.mina.core.polling.AbstractPollingIoProcessor.process(AbstractPollingIoProcessor.java:659)
  106. at org.apache.mina.core.polling.AbstractPollingIoProcessor.process(AbstractPollingIoProcessor.java:648)
  107. at org.apache.mina.core.polling.AbstractPollingIoProcessor.access$600(AbstractPollingIoProcessor.java:68)
  108. at org.apache.mina.core.polling.AbstractPollingIoProcessor$Processor.run(AbstractPollingIoProcessor.java:1120)
  109. at org.apache.mina.util.NamePreservingRunnable.run(NamePreservingRunnable.java:64)
  110. at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1149)
  111. at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:624)
  112. at java.lang.Thread.run(Thread.java:748)
  113. (custom_audio_device_module.cc:14): Createworker_thread
  114. (audio_device_buffer.cc:62): AudioDeviceBuffer::ctor
  115. (audio_device_impl.cc:128): AudioDeviceModuleImpl
  116. (custom_audio_device_module.cc:44): CustomAudioDeviceModule
  117. (audio_device_impl.cc:136): CheckPlatform
  118. (audio_device_impl.cc:150): current platform is Linux
  119. (audio_device_impl.cc:344): AttachAudioBuffer
  120. (file_audio_device.cc:536): AttachAudioBuffer
  121. (audio_device_buffer.cc:178): SetRecordingSampleRate(0)
  122. (audio_device_buffer.cc:185): SetPlayoutSampleRate(0)
  123. (audio_device_buffer.cc:202): SetRecordingChannels(0)
  124. (audio_device_buffer.cc:209): SetPlayoutChannels(0)
  125. (webrtcvoiceengine.cc:236): WebRtcVoiceEngine::WebRtcVoiceEngine
  126. (webrtcvoiceengine.cc:243): Supported send codecs in order of preference:
  127. (webrtcvoiceengine.cc:246): opus/48000/2 { minptime=10 useinbandfec=1 } (111)
  128. (webrtcvoiceengine.cc:246): ISAC/16000/1 (103)
  129. (webrtcvoiceengine.cc:246): ISAC/32000/1 (104)
  130. (webrtcvoiceengine.cc:246): G722/8000/1 (9)
  131. (webrtcvoiceengine.cc:246): ILBC/8000/1 (102)
  132. (webrtcvoiceengine.cc:246): PCMU/8000/1 (0)
  133. (webrtcvoiceengine.cc:246): PCMA/8000/1 (8)
  134. (webrtcvoiceengine.cc:246): CN/32000/1 (106)
  135. (webrtcvoiceengine.cc:246): CN/16000/1 (105)
  136. (webrtcvoiceengine.cc:246): CN/8000/1 (13)
  137. (webrtcvoiceengine.cc:246): telephone-event/48000/1 (110)
  138. (webrtcvoiceengine.cc:246): telephone-event/32000/1 (112)
  139. (webrtcvoiceengine.cc:246): telephone-event/16000/1 (113)
  140. (webrtcvoiceengine.cc:246): telephone-event/8000/1 (126)
  141. (webrtcvoiceengine.cc:249): Supported recv codecs in order of preference:
  142. (webrtcvoiceengine.cc:252): opus/48000/2 { minptime=10 useinbandfec=1 } (111)
  143. (webrtcvoiceengine.cc:252): ISAC/16000/1 (103)
  144. (webrtcvoiceengine.cc:252): ISAC/32000/1 (104)
  145. (webrtcvoiceengine.cc:252): G722/8000/1 (9)
  146. (webrtcvoiceengine.cc:252): ILBC/8000/1 (102)
  147. (webrtcvoiceengine.cc:252): PCMU/8000/1 (0)
  148. (webrtcvoiceengine.cc:252): PCMA/8000/1 (8)
  149. (webrtcvoiceengine.cc:252): CN/32000/1 (106)
  150. (webrtcvoiceengine.cc:252): CN/16000/1 (105)
  151. (webrtcvoiceengine.cc:252): CN/8000/1 (13)
  152. (webrtcvoiceengine.cc:252): telephone-event/48000/1 (110)
  153. (webrtcvoiceengine.cc:252): telephone-event/32000/1 (112)
  154. (webrtcvoiceengine.cc:252): telephone-event/16000/1 (113)
  155. (webrtcvoiceengine.cc:252): telephone-event/8000/1 (126)
  156. (webrtcvoiceengine.cc:260): VoiceEngine 4.1.0
  157. (voe_base_impl.cc:258): virtual int webrtc::VoEBaseImpl::Init(webrtc::AudioDeviceModule *, webrtc::AudioProcessing *, const rtc::scoped_refptr<AudioDecoderFactory> &): An external ADM implementation will be used in VoiceEngine
  158. (audio_device_impl.cc:1452): RegisterEventObserver
  159. (audio_device_impl.cc:1465): RegisterAudioCallback
  160. (audio_device_buffer.cc:77): RegisterAudioCallback
  161. (audio_device_impl.cc:467): Init
  162. (file_audio_device.cc:57): Init
  163. (audio_device_impl.cc:1196): SetPlayoutDevice(0)
  164. (file_audio_device.cc:114): SetPlayoutDevice
  165. (audio_device_impl.cc:516): InitSpeaker
  166. (file_audio_device.cc:335): InitSpeaker
  167. (audio_device_impl.cc:1291): SetRecordingDevice(0)
  168. (file_audio_device.cc:129): SetRecordingDevice index: 0
  169. (audio_device_impl.cc:526): InitMicrophone
  170. (file_audio_device.cc:342): InitMicrophone
  171. (audio_device_impl.cc:1005): StereoPlayoutIsAvailable
  172. (file_audio_device.cc:442): StereoPlayoutIsAvailable
  173. (audio_device_impl.cc:1015): output: 1
  174. (audio_device_impl.cc:1024): SetStereoPlayout(1)
  175. (file_audio_device.cc:164): PlayoutIsInitialized
  176. (file_audio_device.cc:447): SetStereoPlayout
  177. (audio_device_buffer.cc:209): SetPlayoutChannels(2)
  178. (audio_device_impl.cc:891): StereoRecordingIsAvailable
  179. (file_audio_device.cc:458): StereoRecordingIsAvailable
  180. (audio_device_impl.cc:901): output: 1
  181. (audio_device_impl.cc:910): SetStereoRecording(1)
  182. (file_audio_device.cc:196): RecordingIsInitialized
  183. (file_audio_device.cc:464): SetStereoRecording
  184. (audio_device_buffer.cc:202): SetRecordingChannels(2)
  185. (webrtcvoiceengine.cc:640): webrtc: TransmitMixer::SetAudioProcessingModule(audioProcessingModule=0xe80fa520)
  186. (webrtcvoiceengine.cc:640): webrtc: OutputMixer::SetAudioProcessingModule(audioProcessingModule=0xe80fa520)
  187. (audio_processing_impl.cc:641): Level controller activated: 0
  188. (audio_processing_impl.cc:648): Highpass filter activated: 1
  189. (webrtcvoiceengine.cc:336): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: true, agc: true, ns: true, hf: true, swap: false, audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, typing: true, agc_delta: 0, experimental_agc: false, extended_filter_aec: false, delay_agnostic_aec: false, experimental_ns: false, intelligibility_enhancer: false, level_control: false, residual_echo_detector: true, }
  190. (audio_device_impl.cc:1760): BuiltInAECIsAvailable
  191. (audio_device_generic.cc:51): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform
  192. (audio_device_impl.cc:1763): output: 0
  193. (apm_helpers.cc:106): Echo control set to 1 with mode 0
  194. (apm_helpers.cc:116): EC metrics set to 1
  195. (audio_device_impl.cc:1776): BuiltInAGCIsAvailable
  196. (audio_device_generic.cc:61): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform
  197. (audio_device_impl.cc:1779): output: 0
  198. (audio_device_impl.cc:1071): SetAGC(1)
  199. (file_audio_device.cc:315): SetAGC
  200. (apm_helpers.cc:67): Failed to set AGC mode in ADM: 1
  201. (webrtcvoiceengine.cc:450): Adjusting AGC level from default -2dB to -2dB
  202. (audio_device_impl.cc:1792): BuiltInNSIsAvailable
  203. (audio_device_generic.cc:71): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform
  204. (audio_device_impl.cc:1795): output: 0
  205. (apm_helpers.cc:141): NS set to 1
  206. (webrtcvoiceengine.cc:481): Stereo swapping enabled? 0
  207. (webrtcvoiceengine.cc:486): NetEq capacity is 50
  208. (webrtcvoiceengine.cc:492): NetEq fast mode? 0
  209. (webrtcvoiceengine.cc:499): Typing detection is enabled? 1
  210. (apm_helpers.cc:166): VAD set to 1 for typing detection.
  211. (webrtcvoiceengine.cc:510): Delay agnostic aec is enabled? 0
  212. (webrtcvoiceengine.cc:519): Extended filter aec is enabled? 0
  213. (webrtcvoiceengine.cc:528): Experimental ns is enabled? 0
  214. (webrtcvoiceengine.cc:534): Intelligibility Enhancer is enabled? 0
  215. (webrtcvoiceengine.cc:544): Level control: 0
  216. (audio_processing_impl.cc:641): Level controller activated: 0
  217. (audio_processing_impl.cc:648): Highpass filter activated: 1
  218. (audio_device_impl.cc:1441): Recording
  219. (file_audio_device.cc:310): Recording
  220. (audio_device_impl.cc:957): SetRecordingChannel(both)
  221. (file_audio_device.cc:469): StereoRecording
  222. (audio_device_buffer.cc:216): SetRecordingChannel(2)
  223. (audio_device_buffer.cc:217): Not implemented
  224. (adm_helpers.cc:47): Unable to set recording channel to kChannelBoth.
  225. (audio_device_impl.cc:1291): SetRecordingDevice(0)
  226. (file_audio_device.cc:129): SetRecordingDevice index: 0
  227. (audio_device_impl.cc:526): InitMicrophone
  228. (file_audio_device.cc:342): InitMicrophone
  229. (audio_device_impl.cc:891): StereoRecordingIsAvailable
  230. (file_audio_device.cc:458): StereoRecordingIsAvailable
  231. (audio_device_impl.cc:901): output: 1
  232. (audio_device_impl.cc:910): SetStereoRecording(1)
  233. (file_audio_device.cc:196): RecordingIsInitialized
  234. (file_audio_device.cc:464): SetStereoRecording
  235. (audio_device_buffer.cc:202): SetRecordingChannels(2)
  236. (adm_helpers.cc:80): Set recording device.
  237. (audio_device_impl.cc:1399): Playing
  238. (file_audio_device.cc:256): Playing
  239. (audio_device_impl.cc:1196): SetPlayoutDevice(0)
  240. (file_audio_device.cc:114): SetPlayoutDevice
  241. (audio_device_impl.cc:516): InitSpeaker
  242. (file_audio_device.cc:335): InitSpeaker
  243. (audio_device_impl.cc:1005): StereoPlayoutIsAvailable
  244. (file_audio_device.cc:442): StereoPlayoutIsAvailable
  245. (audio_device_impl.cc:1015): output: 1
  246. (audio_device_impl.cc:1024): SetStereoPlayout(1)
  247. (file_audio_device.cc:164): PlayoutIsInitialized
  248. (file_audio_device.cc:447): SetStereoPlayout
  249. (audio_device_buffer.cc:209): SetPlayoutChannels(2)
  250. (adm_helpers.cc:124): Set playout device.
  251. (audio_processing_impl.cc:641): Level controller activated: 0
  252. (audio_processing_impl.cc:648): Highpass filter activated: 0
  253. (audio_device_impl.cc:1465): RegisterAudioCallback
  254. (audio_device_buffer.cc:77): RegisterAudioCallback
  255. (audio_device_impl.cc:1465): RegisterAudioCallback
  256. (audio_device_buffer.cc:77): RegisterAudioCallback
  257. (webrtcvideoengine2.cc:467): WebRtcVideoEngine2::WebRtcVideoEngine2()
  258. (webrtcvideoengine2.cc:475): WebRtcVideoEngine2::Init
  259. (peerconnection_jni.cc:1349): InvokeJavaCallbacksOnFactoryThreads.
  260. (peerconnection_jni.cc:1330): Network thread JavaCallback
  261. (peerconnection_jni.cc:1334): Worker thread JavaCallback
  262. (peerconnection_jni.cc:1338): Signaling thread JavaCallback
  263. (peerconnection_jni.cc:1822): jrtc 1
  264. (bitrate_prober.cc:63): Bandwidth probing enabled, set to inactive
  265. (delay_based_bwe.cc:173): Using Trendline filter for delay change estimation.
  266. (cpu_info.cc:50): Available number of cores: 1
  267. (remote_bitrate_estimator_single_stream.cc:58): RemoteBitrateEstimatorSingleStream: Instantiating.
  268. (send_side_congestion_controller.cc:185): SignalNetworkState Down
  269. (paced_sender.cc:279): PacedSender paused.
  270. (delay_based_bwe.cc:361): BWE Setting start bitrate to: 300000
  271. (paced_sender.cc:485): ProcessThreadAttached 0x0x7f70e81ea0e0
  272. (webrtcvideoengine2.cc:612): Internally supported codecs: {VideoCodec[0:VP8], VideoCodec[0:VP9], VideoCodec[0:red], VideoCodec[0:ulpfec]}
  273. (opensslidentity.cc:41): Making key pair
  274. (opensslidentity.cc:89): Returning key pair
  275. (opensslidentity.cc:96): Making certificate for WebRTC
  276. (opensslidentity.cc:143): Returning certificate
  277. [INFO] [NioProcessor-21] org.red5.net.websocket.codec.WebSocketDecoder - Not enough data available to decode, socket may be closed/closing
  278. [INFO] [NioProcessor-21] org.red5.net.websocket.codec.WebSocketDecoder - Not enough data available to decode, socket may be closed/closing
  279. [INFO] [NioProcessor-21] io.antmedia.webrtc.WebSocketListener - onWSMessage: {"command":"takeConfiguration","type":"offer","sdp":"v=0\r\no=- 7856647205929912001 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video\r\na=msid-semantic: WMS fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:86UI\r\na=ice-pwd:5QuM7kkxweh5M7vCcMUk3rMk\r\na=ice-options:trickle\r\na=fingerprint:sha-256 5E:CB:A4:FF:48:91:F8:B0:D0:47:AE:63:0C:F4:E1:D1:27:93:C1:A4:35:51:43:F4:4E:19:F9:90:F1:B8:70:7A\r\na=setup:actpass\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=sendonly\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000\r\na=rtpmap:113 telephone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\na=ssrc:2798648372 cname:ZsaV4bHJ05lWG/sk\r\na=ssrc:2798648372 msid:fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o d1f9a50f-b30c-4a2f-af65-7726684d0338\r\na=ssrc:2798648372 mslabel:fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o\r\na=ssrc:2798648372 label:d1f9a50f-b30c-4a2f-af65-7726684d0338\r\nm=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 124 127 125 123\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:86UI\r\na=ice-pwd:5QuM7kkxweh5M7vCcMUk3rMk\r\na=ice-options:trickle\r\na=fingerprint:sha-256 5E:CB:A4:FF:48:91:F8:B0:D0:47:AE:63:0C:F4:E1:D1:27:93:C1:A4:35:51:43:F4:4E:19:F9:90:F1:B8:70:7A\r\na=setup:actpass\r\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:4 urn:3gpp:video-orientation\r\na=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=sendonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:96 VP8/90000\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:98 VP9/90000\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtpmap:99 rtx/90000\r\na=fmtp:99 apt=98\r\na=rtpmap:100 H264/90000\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:101 rtx/90000\r\na=fmtp:101 apt=100\r\na=rtpmap:102 red/90000\r\na=rtpmap:124 rtx/90000\r\na=fmtp:124 apt=102\r\na=rtpmap:127 ulpfec/90000\r\na=rtpmap:125 H264/90000\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=420032\r\na=rtpmap:123 rtx/90000\r\na=fmtp:123 apt=125\r\na=ssrc-group:FID 3010411960 2208960908\r\na=ssrc:3010411960 cname:ZsaV4bHJ05lWG/sk\r\na=ssrc:3010411960 msid:fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o 0a1611a0-2627-40cd-a5f7-3ade60f18100\r\na=ssrc:3010411960 mslabel:fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o\r\na=ssrc:3010411960 label:0a1611a0-2627-40cd-a5f7-3ade60f18100\r\na=ssrc:2208960908 cname:ZsaV4bHJ05lWG/sk\r\na=ssrc:2208960908 msid:fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o 0a1611a0-2627-40cd-a5f7-3ade60f18100\r\na=ssrc:2208960908 mslabel:fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o\r\na=ssrc:2208960908 label:0a1611a0-2627-40cd-a5f7-3ade60f18100\r\n"}
  280.  
  281. received sdp type is offer
  282. (p2ptransportchannel.cc:400): Set ping most likely connection to 0
  283. (p2ptransportchannel.cc:420): Set presume writable when fully relayed to 0
  284. (webrtcvoiceengine.cc:1494): Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
  285. (webrtcvoiceengine.cc:336): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
  286. (webrtcvoiceengine.cc:486): NetEq capacity is 50
  287. (webrtcvoiceengine.cc:492): NetEq fast mode? 0
  288. (webrtcvoiceengine.cc:510): Delay agnostic aec is enabled? 0
  289. (webrtcvoiceengine.cc:519): Extended filter aec is enabled? 0
  290. (webrtcvoiceengine.cc:528): Experimental ns is enabled? 0
  291. (webrtcvoiceengine.cc:534): Intelligibility Enhancer is enabled? 0
  292. (webrtcvoiceengine.cc:544): Level control: 0
  293. (audio_processing_impl.cc:641): Level controller activated: 0
  294. (audio_processing_impl.cc:648): Highpass filter activated: 1
  295. (webrtcvoiceengine.cc:1513): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
  296. (channel.cc:182): Created channel for audio
  297. (channel.cc:334): Setting RTP Transport for audio on audio transport 0x243f520
  298. (p2ptransportchannel.cc:400): Set ping most likely connection to 0
  299. (p2ptransportchannel.cc:420): Set presume writable when fully relayed to 0
  300. (webrtcvideoengine2.cc:484): CreateChannel. Options: VideoOptions {}
  301. (webrtcvideoengine2.cc:612): Internally supported codecs: {VideoCodec[0:VP8], VideoCodec[0:VP9], VideoCodec[0:red], VideoCodec[0:ulpfec]}
  302. (channel.cc:182): Created channel for video
  303. (channel.cc:334): Setting RTP Transport for video on video transport 0x2440b60
  304. (transportcontroller.cc:638): Set remote transport description on audio
  305. (p2ptransportchannel.cc:334): Remote supports ICE renomination ? 0
  306. (transportcontroller.cc:638): Set remote transport description on video
  307. (p2ptransportchannel.cc:334): Remote supports ICE renomination ? 0
  308. (webrtcsession.cc:848): Session:4616553514388041885 Old state:STATE_INIT New state:STATE_RECEIVEDOFFER
  309. (channel.cc:1792): Setting remote voice description
  310. (webrtcvoiceengine.cc:1322): WebRtcVoiceMediaChannel::SetSendParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:0:1], AudioCodec[104:ISAC:32000:0:1], AudioCodec[9:G722:8000:0:1], AudioCodec[0:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:1], AudioCodec[113:telephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], max_bandwidth_bps: -1, options: AudioOptions {}}
  311. (webrtcvoiceengine.cc:1711): Recreate all the receive streams because the send codec has changed.
  312. (webrtcvoiceengine.cc:2163): WebRtcVoiceMediaChannel::SetMaxSendBitrate.
  313. (webrtcvoiceengine.cc:1494): Setting voice channel options: AudioOptions {}
  314. (webrtcvoiceengine.cc:336): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
  315. (webrtcvoiceengine.cc:486): NetEq capacity is 50
  316. (webrtcvoiceengine.cc:492): NetEq fast mode? 0
  317. (webrtcvoiceengine.cc:510): Delay agnostic aec is enabled? 0
  318. (webrtcvoiceengine.cc:519): Extended filter aec is enabled? 0
  319. (webrtcvoiceengine.cc:528): Experimental ns is enabled? 0
  320. (webrtcvoiceengine.cc:534): Intelligibility Enhancer is enabled? 0
  321. (webrtcvoiceengine.cc:544): Level control: 0
  322. (audio_processing_impl.cc:641): Level controller activated: 0
  323. (audio_processing_impl.cc:648): Highpass filter activated: 1
  324. (webrtcvoiceengine.cc:1513): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
  325. (webrtcvoiceengine.cc:1892): AddRecvStream: {id:d1f9a50f-b30c-4a2f-af65-7726684d0338;ssrcs:[2798648372];ssrc_groups:;cname:ZsaV4bHJ05lWG/sk;sync_label:fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o}
  326. (neteq_impl.cc:109): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=50, background_noise_mode=2, playout_mode=0, enable_fast_accelerate=false, enable_muted_state= true
  327. (audio_receive_stream.cc:71): AudioReceiveStream: {rtp: {remote_ssrc: 2798648372, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), voe_channel_id: 0, sync_group: fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o}
  328. (call.cc:980): UpdateAggregateNetworkState: aggregate_state=down
  329. (send_side_congestion_controller.cc:185): SignalNetworkState Down
  330. (paced_sender.cc:279): PacedSender paused.
  331. (webrtcvoiceengine.cc:1254): Stopping playout for channel #0
  332. (webrtcvoiceengine.cc:1254): Stopping playout for channel #0
  333. (channel.cc:1396): Add remote ssrc: 2798648372
  334. (channel.cc:1734): Changing voice state, recv=0 send=0
  335. (channel.cc:2065): Setting remote video description
  336. (webrtcvideoengine2.cc:759): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:H264], VideoCodec[101:rtx], VideoCodec[102:red], VideoCodec[124:rtx], VideoCodec[127:ulpfec], VideoCodec[125:H264], VideoCodec[123:rtx]], extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}], max_bandwidth_bps: -1, }
  337. (webrtcvideoengine2.cc:612): Internally supported codecs: {VideoCodec[0:VP8], VideoCodec[0:VP9], VideoCodec[0:red], VideoCodec[0:ulpfec]}
  338. (webrtcmediaengine.cc:198): Unsupported RTP extension: {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}
  339. (webrtcvideoengine2.cc:768): Using codec: VideoCodec[96:VP8]
  340. (webrtcvideoengine2.cc:816): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed.
  341. (webrtcvideoengine2.cc:1192): AddRecvStream: {id:0a1611a0-2627-40cd-a5f7-3ade60f18100;ssrcs:[3010411960,2208960908];ssrc_groups:{semantics:FID;ssrcs:[3010411960,2208960908]};cname:ZsaV4bHJ05lWG/sk;sync_label:fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o}
  342. (video_receive_stream.cc:198): VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 96, payload_name: VP8, codec_params: {}}, {decoder: (VideoDecoder), payload_type: 98, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 3010411960, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 102, red_payload_type: 100, red_rtx_payload_type: 101}, rtx_ssrc: 2208960908, rtx_payload_types: {96 (apt) -> 97 (pt), 98 (apt) -> 99 (pt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o, pre_decode_callback: nullptr, target_delay_ms: 0}
  343. (call.cc:980): UpdateAggregateNetworkState: aggregate_state=down
  344. (send_side_congestion_controller.cc:185): SignalNetworkState Down
  345. (paced_sender.cc:279): PacedSender paused.
  346. (channel.cc:1396): Add remote ssrc: 3010411960
  347. (channel.cc:1987): Changing video state, send=0
  348. (webrtcsession.cc:661): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport.
  349. (webrtcvoiceengine.cc:2020): SetOutputVolume() to 1 for recv stream with ssrc 2798648372
  350. (webrtcvideoengine2.cc:1307): SetSink: ssrc:3010411960 (ptr)
  351. [WARN] [<noname> - p???] io.antmedia.webrtc.WebSocketListener - onAddStream
  352. (peerconnection_jni.cc:2499): AudioTrack::nativeAddSink
  353. (peerconnection_jni.cc:2479): VideoTrack::nativeAddRenderer
  354. (webrtcsession.cc:677): Local and Remote descriptions must be applied to get the SSL Role of the session.
  355. (webrtcsession.cc:677): Local and Remote descriptions must be applied to get the SSL Role of the session.
  356. [WARN] [<noname> - p???] io.antmedia.webrtc.WebSocketListener - onCreate Success
  357. (webrtcsession.cc:1083): BUNDLE already enabled for audio on audio.
  358. (channel.cc:334): Setting RTP Transport for video on audio transport 0x243f520
  359. (call.cc:921): Transport video is disconnected
  360. (webrtcsession.cc:1099): Enabled BUNDLE for video on audio.
  361. (transportcontroller.cc:610): Set local transport description on audio
  362. (p2ptransportchannel.cc:323): Set ICE ufrag: BSyN pwd: RAWiPuyn/8wqcRGcwgvpCch0 on transport audio
  363. (dtlstransportchannel.cc:315): Jingle:DtlsTransport[audio|1|__]: DTLS setup complete.
  364. (channel.cc:904): Channel enabled
  365. (channel.cc:1734): Changing voice state, recv=0 send=0
  366. (channel.cc:904): Channel enabled
  367. (channel.cc:1987): Changing video state, send=0
  368. (webrtcsession.cc:848): Session:4616553514388041885 Old state:STATE_RECEIVEDOFFER New state:STATE_INPROGRESS
  369. (channel.cc:1747): Setting local voice description
  370. (channel.cc:1218): Enabling rtcp-mux for audio; no longer need RTCP transport for audio
  371. (webrtcvoiceengine.cc:1354): WebRtcVoiceMediaChannel::SetRecvParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[103:ISAC:16000:0:1], AudioCodec[104:ISAC:32000:0:1], AudioCodec[9:G722:8000:0:1], AudioCodec[0:PCMU:8000:0:1], AudioCodec[8:PCMA:8000:0:1], AudioCodec[106:CN:32000:0:1], AudioCodec[105:CN:16000:0:1], AudioCodec[13:CN:8000:0:1], AudioCodec[110:telephone-event:48000:0:1], AudioCodec[112:telephone-event:32000:0:1], AudioCodec[113:telephone-event:16000:0:1], AudioCodec[126:telephone-event:8000:0:1]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}]}
  372. (webrtcvoiceengine.cc:1523): Setting receive voice codecs.
  373. (call.cc:980): UpdateAggregateNetworkState: aggregate_state=down
  374. (send_side_congestion_controller.cc:185): SignalNetworkState Down
  375. (paced_sender.cc:279): PacedSender paused.
  376. (audio_receive_stream.cc:114): ~AudioReceiveStream: {rtp: {remote_ssrc: 2798648372, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), voe_channel_id: 0, sync_group: fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o}
  377. (audio_receive_stream.cc:71): AudioReceiveStream: {rtp: {remote_ssrc: 2798648372, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), voe_channel_id: 0, sync_group: fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o}
  378. (call.cc:980): UpdateAggregateNetworkState: aggregate_state=down
  379. (send_side_congestion_controller.cc:185): SignalNetworkState Down
  380. (paced_sender.cc:279): PacedSender paused.
  381. (webrtcvoiceengine.cc:1254): Stopping playout for channel #0
  382. (call.cc:980): UpdateAggregateNetworkState: aggregate_state=down
  383. (send_side_congestion_controller.cc:185): SignalNetworkState Down
  384. (paced_sender.cc:279): PacedSender paused.
  385. (audio_receive_stream.cc:114): ~AudioReceiveStream: {rtp: {remote_ssrc: 2798648372, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), voe_channel_id: 0, sync_group: fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o}
  386. (audio_receive_stream.cc:71): AudioReceiveStream: {rtp: {remote_ssrc: 2798648372, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), voe_channel_id: 0, sync_group: fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o}
  387. (call.cc:980): UpdateAggregateNetworkState: aggregate_state=down
  388. (send_side_congestion_controller.cc:185): SignalNetworkState Down
  389. (paced_sender.cc:279): PacedSender paused.
  390. (webrtcvoiceengine.cc:1254): Stopping playout for channel #0
  391. (webrtcvoiceengine.cc:1251): Starting playout for channel #0
  392. (audio_device_impl.cc:1399): Playing
  393. (file_audio_device.cc:256): Playing
  394. (audio_device_impl.cc:1312): InitPlayout
  395. (audio_device_impl.cc:1346): PlayoutIsInitialized
  396. (file_audio_device.cc:164): PlayoutIsInitialized
  397. (file_audio_device.cc:154): InitPlayout
  398. (audio_device_buffer.cc:185): SetPlayoutSampleRate(48000)
  399. (audio_device_buffer.cc:209): SetPlayoutChannels(2)
  400. (audio_device_impl.cc:1318): output: 0
  401. (audio_device_impl.cc:1366): StartPlayout
  402. (audio_device_impl.cc:1399): Playing
  403. (file_audio_device.cc:256): Playing
  404. (audio_device_buffer.cc:94): StartPlayout
  405. (file_audio_device.cc:201): StartPlayout
  406. (file_audio_device.cc:225): Started playout capture:
  407. (audio_device_impl.cc:1373): output: 0
  408. (audio_device_buffer.cc:354): Size of playout buffer: 960
  409. (channel.cc:1734): Changing voice state, recv=1 send=0
  410. (channel.cc:2020): Setting local video description
  411. (channel.cc:1218): Enabling rtcp-mux for video; no longer need RTCP transport for audio
  412. (webrtcvideoengine2.cc:985): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[102:red], VideoCodec[124:rtx], VideoCodec[127:ulpfec]], extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}]}
  413. (webrtcvideoengine2.cc:612): Internally supported codecs: {VideoCodec[0:VP8], VideoCodec[0:VP9], VideoCodec[0:red], VideoCodec[0:ulpfec]}
  414. (webrtcvideoengine2.cc:994): Changing recv codecs from {VideoCodec[96:VP8], VideoCodec[98:VP9]} to {VideoCodec[96:VP8], VideoCodec[98:VP9]}
  415. (webrtcvideoengine2.cc:2345): RecreateWebRtcStream (recv) because of SetRecvParameters
  416. (call.cc:980): UpdateAggregateNetworkState: aggregate_state=down
  417. (send_side_congestion_controller.cc:185): SignalNetworkState Down
  418. (paced_sender.cc:279): PacedSender paused.
  419. (video_receive_stream.cc:228): ~VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 96, payload_name: VP8, codec_params: {}}, {decoder: (VideoDecoder), payload_type: 98, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 3010411960, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 102, red_payload_type: 100, red_rtx_payload_type: 101}, rtx_ssrc: 2208960908, rtx_payload_types: {96 (apt) -> 97 (pt), 98 (apt) -> 99 (pt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o, pre_decode_callback: nullptr, target_delay_ms: 0}
  420. (video_receive_stream.cc:198): VideoReceiveStream: {decoders: [{decoder: (VideoDecoder), payload_type: 96, payload_name: VP8, codec_params: {}}, {decoder: (VideoDecoder), payload_type: 98, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 3010411960, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 127, red_payload_type: 102, red_rtx_payload_type: 124}, rtx_ssrc: 2208960908, rtx_payload_types: {96 (apt) -> 97 (pt), 98 (apt) -> 99 (pt), }, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}]}, renderer: (renderer), render_delay_ms: 10, sync_group: fU2roDuumsfyyVMpjze7sk4FxVxhdCXsU16o, pre_decode_callback: nullptr, target_delay_ms: 0}
  421. (call.cc:980): UpdateAggregateNetworkState: aggregate_state=down
  422. (send_side_congestion_controller.cc:185): SignalNetworkState Down
  423. (paced_sender.cc:279): PacedSender paused.
  424. (channel.cc:1987): Changing video state, send=0
  425. (webrtcsession.cc:666): Non-rejected SCTP m= section is needed to get the SSL Role of the SCTP transport.
  426. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  427. (basicportallocator.cc:274): Start getting ports with prune_turn_ports disabled
  428. (basicportallocator.cc:676): Network manager has started
  429. (basicportallocator.cc:594): Allocate ports on 4 networks
  430. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  431. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  432. (basicportallocator.cc:1143): Jingle:Net[venet0:0:51.254.59.155/32:Unknown]: Allocation Phase=Udp
  433. (port.cc:214): Jingle:Port[0x2484e50::1:0:local:Net[venet0:0:51.254.59.155/32:Unknown]]: Port created with network cost 50
  434. (basicportallocator.cc:1234): AllocationSequence: UDPPort will be handling the STUN candidate generation.
  435. (basicportallocator.cc:698): Adding allocated port for audio
  436. (basicportallocator.cc:718): Jingle:Port[0x2484e50:audio:1:0:local:Net[venet0:0:51.254.59.155/32:Unknown]]: Added port to allocator
  437. (basicportallocator.cc:735): Jingle:Port[0x2484e50:audio:1:0:local:Net[venet0:0:51.254.59.155/32:Unknown]]: Gathered candidate: Cand[:3111367026:1:udp:2122260223:51.254.59.155:49235:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:4:50:0]
  438. (basicportallocator.cc:762): Jingle:Port[0x2484e50:audio:1:0:local:Net[venet0:0:51.254.59.155/32:Unknown]]: Port ready.
  439. (physicalsocketserver.cc:570): Socket::OPT_DSCP not supported.
  440. (p2ptransportchannel.cc:528): Jingle:Port[0x2484e50:audio:1:0:local:Net[venet0:0:51.254.59.155/32:Unknown]]: SetOption(5, 0) failed: 0
  441. (basicportallocator.cc:778): Not yet signaling candidate because protocol is not yet enabled.
  442. (stunport.cc:372): Jingle:Port[0x2484e50:audio:1:0:local:Net[venet0:0:51.254.59.155/32:Unknown]]: Starting STUN host lookup for stun.l.google.com:19302
  443. (basicportallocator.cc:894): Signaling candidate because protocol was enabled: Cand[:3111367026:1:udp:2122260223:51.254.59.155:49235:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:4:50:0]
  444. (basicportallocator.cc:1143): Jingle:Net[venet0:127.0.0.2/32:Unknown]: Allocation Phase=Udp
  445. (port.cc:214): Jingle:Port[0x2486a90::1:0:local:Net[venet0:127.0.0.2/32:Unknown]]: Port created with network cost 50
  446. (basicportallocator.cc:1234): AllocationSequence: UDPPort will be handling the STUN candidate generation.
  447. (basicportallocator.cc:698): Adding allocated port for audio
  448. (basicportallocator.cc:718): Jingle:Port[0x2486a90:audio:1:0:local:Net[venet0:127.0.0.2/32:Unknown]]: Added port to allocator
  449. (basicportallocator.cc:735): Jingle:Port[0x2486a90:audio:1:0:local:Net[venet0:127.0.0.2/32:Unknown]]: Gathered candidate: Cand[:1220512899:1:udp:2122194687:127.0.0.2:34973:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:3:50:0]
  450. (basicportallocator.cc:762): Jingle:Port[0x2486a90:audio:1:0:local:Net[venet0:127.0.0.2/32:Unknown]]: Port ready.
  451. (physicalsocketserver.cc:570): Socket::OPT_DSCP not supported.
  452. (p2ptransportchannel.cc:528): Jingle:Port[0x2486a90:audio:1:0:local:Net[venet0:127.0.0.2/32:Unknown]]: SetOption(5, 0) failed: 0
  453. (basicportallocator.cc:778): Not yet signaling candidate because protocol is not yet enabled.
  454. (stunport.cc:372): Jingle:Port[0x2486a90:audio:1:0:local:Net[venet0:127.0.0.2/32:Unknown]]: Starting STUN host lookup for stun.l.google.com:19302
  455. [WARN] [<noname> - p???] io.antmedia.webrtc.WebSocketListener - onIceCandidate
  456. (basicportallocator.cc:894): Signaling candidate because protocol was enabled: Cand[:1220512899:1:udp:2122194687:127.0.0.2:34973:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:3:50:0]
  457. (basicportallocator.cc:1143): Jingle:Net[lo:::1/128:Loopback]: Allocation Phase=Udp
  458. (port.cc:214): Jingle:Port[0x2488550::1:0:local:Net[lo:::1/128:Loopback]]: Port created with network cost 0
  459. (basicportallocator.cc:1234): AllocationSequence: UDPPort will be handling the STUN candidate generation.
  460. (basicportallocator.cc:698): Adding allocated port for audio
  461. (basicportallocator.cc:718): Jingle:Port[0x2488550:audio:1:0:local:Net[lo:::1/128:Loopback]]: Added port to allocator
  462. (basicportallocator.cc:735): Jingle:Port[0x2488550:audio:1:0:local:Net[lo:::1/128:Loopback]]: Gathered candidate: Cand[:559267639:1:udp:2122136831:[::1]:40457:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:2:0:0]
  463. (basicportallocator.cc:762): Jingle:Port[0x2488550:audio:1:0:local:Net[lo:::1/128:Loopback]]: Port ready.
  464. (physicalsocketserver.cc:570): Socket::OPT_DSCP not supported.
  465. (p2ptransportchannel.cc:528): Jingle:Port[0x2488550:audio:1:0:local:Net[lo:::1/128:Loopback]]: SetOption(5, 0) failed: 0
  466. (basicportallocator.cc:778): Not yet signaling candidate because protocol is not yet enabled.
  467. (stunport.cc:372): Jingle:Port[0x2488550:audio:1:0:local:Net[lo:::1/128:Loopback]]: Starting STUN host lookup for stun.l.google.com:19302
  468. [WARN] [<noname> - p???] io.antmedia.webrtc.WebSocketListener - onIceCandidate
  469. (basicportallocator.cc:894): Signaling candidate because protocol was enabled: Cand[:559267639:1:udp:2122136831:[::1]:40457:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:2:0:0]
  470. (basicportallocator.cc:1143): Jingle:Net[lo:127.0.0.0/8:Loopback]: Allocation Phase=Udp
  471. (port.cc:214): Jingle:Port[0x2489ed0::1:0:local:Net[lo:127.0.0.0/8:Loopback]]: Port created with network cost 0
  472. (basicportallocator.cc:1234): AllocationSequence: UDPPort will be handling the STUN candidate generation.
  473. (basicportallocator.cc:698): Adding allocated port for audio
  474. (basicportallocator.cc:718): Jingle:Port[0x2489ed0:audio:1:0:local:Net[lo:127.0.0.0/8:Loopback]]: Added port to allocator
  475. (basicportallocator.cc:735): Jingle:Port[0x2489ed0:audio:1:0:local:Net[lo:127.0.0.0/8:Loopback]]: Gathered candidate: Cand[:1510613869:1:udp:2122063615:127.0.0.1:40383:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:1:0:0]
  476. (basicportallocator.cc:762): Jingle:Port[0x2489ed0:audio:1:0:local:Net[lo:127.0.0.0/8:Loopback]]: Port ready.
  477. (physicalsocketserver.cc:570): Socket::OPT_DSCP not supported.
  478. (p2ptransportchannel.cc:528): Jingle:Port[0x2489ed0:audio:1:0:local:Net[lo:127.0.0.0/8:Loopback]]: SetOption(5, 0) failed: 0
  479. (basicportallocator.cc:778): Not yet signaling candidate because protocol is not yet enabled.
  480. (stunport.cc:372): Jingle:Port[0x2489ed0:audio:1:0:local:Net[lo:127.0.0.0/8:Loopback]]: Starting STUN host lookup for stun.l.google.com:19302
  481. [WARN] [<noname> - p???] io.antmedia.webrtc.WebSocketListener - onIceCandidate
  482. (basicportallocator.cc:894): Signaling candidate because protocol was enabled: Cand[:1510613869:1:udp:2122063615:127.0.0.1:40383:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:1:0:0]
  483. [WARN] [<noname> - p???] io.antmedia.webrtc.WebSocketListener - onIceCandidate
  484. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  485. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  486. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  487. (stunport.cc:506): sendto : [0x00000016] Invalid argument
  488. (thread.cc:258): Waiting for the thread to join, but blocking calls have been disallowed
  489. (thread.cc:258): Waiting for the thread to join, but blocking calls have been disallowed
  490. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  491. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  492. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  493. (stunport.cc:384): Jingle:Port[0x2488550:audio:1:0:local:Net[lo:::1/128:Loopback]]: StunPort: stun host lookup received error 0
  494. (basicportallocator.cc:844): Jingle:Port[0x2488550:audio:1:0:local:Net[lo:::1/128:Loopback]]: Port completed gathering candidates.
  495. (thread.cc:258): Waiting for the thread to join, but blocking calls have been disallowed
  496. (stunport.cc:506): sendto : [0x00000016] Invalid argument
  497. (thread.cc:258): Waiting for the thread to join, but blocking calls have been disallowed
  498. (basicportallocator.cc:844): Jingle:Port[0x2484e50:audio:1:0:local:Net[venet0:0:51.254.59.155/32:Unknown]]: Port completed gathering candidates.
  499. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  500. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  501. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  502. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  503. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  504. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  505. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  506. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  507. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  508. (basicportallocator.cc:1143): Jingle:Net[venet0:0:51.254.59.155/32:Unknown]: Allocation Phase=Relay
  509. (basicportallocator.cc:1143): Jingle:Net[venet0:127.0.0.2/32:Unknown]: Allocation Phase=Relay
  510. (basicportallocator.cc:1143): Jingle:Net[lo:::1/128:Loopback]: Allocation Phase=Relay
  511. (basicportallocator.cc:1143): Jingle:Net[lo:127.0.0.0/8:Loopback]: Allocation Phase=Relay
  512. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  513. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  514. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  515. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  516. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  517. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  518. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  519. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  520. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  521. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  522. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  523. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  524. (basicportallocator.cc:1143): Jingle:Net[venet0:0:51.254.59.155/32:Unknown]: Allocation Phase=Tcp
  525. (port.cc:214): Jingle:Port[0x248c230::1:0:local:Net[venet0:0:51.254.59.155/32:Unknown]]: Port created with network cost 50
  526. (basicportallocator.cc:698): Adding allocated port for audio
  527. (basicportallocator.cc:718): Jingle:Port[0x248c230:audio:1:0:local:Net[venet0:0:51.254.59.155/32:Unknown]]: Added port to allocator
  528. (basicportallocator.cc:735): Jingle:Port[0x248c230:audio:1:0:local:Net[venet0:0:51.254.59.155/32:Unknown]]: Gathered candidate: Cand[:4159720834:1:tcp:1518280447:51.254.59.155:39415:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:4:50:0]
  529. (basicportallocator.cc:762): Jingle:Port[0x248c230:audio:1:0:local:Net[venet0:0:51.254.59.155/32:Unknown]]: Port ready.
  530. (physicalsocketserver.cc:570): Socket::OPT_DSCP not supported.
  531. (p2ptransportchannel.cc:528): Jingle:Port[0x248c230:audio:1:0:local:Net[venet0:0:51.254.59.155/32:Unknown]]: SetOption(5, 0) failed: 0
  532. (basicportallocator.cc:778): Not yet signaling candidate because protocol is not yet enabled.
  533. (basicportallocator.cc:844): Jingle:Port[0x248c230:audio:1:0:local:Net[venet0:0:51.254.59.155/32:Unknown]]: Port completed gathering candidates.
  534. (basicportallocator.cc:894): Signaling candidate because protocol was enabled: Cand[:4159720834:1:tcp:1518280447:51.254.59.155:39415:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:4:50:0]
  535. [WARN] [<noname> - p???] io.antmedia.webrtc.WebSocketListener - onIceCandidate
  536. (basicportallocator.cc:1143): Jingle:Net[venet0:127.0.0.2/32:Unknown]: Allocation Phase=Tcp
  537. (port.cc:214): Jingle:Port[0x248d5a0::1:0:local:Net[venet0:127.0.0.2/32:Unknown]]: Port created with network cost 50
  538. (basicportallocator.cc:698): Adding allocated port for audio
  539. (basicportallocator.cc:718): Jingle:Port[0x248d5a0:audio:1:0:local:Net[venet0:127.0.0.2/32:Unknown]]: Added port to allocator
  540. (basicportallocator.cc:735): Jingle:Port[0x248d5a0:audio:1:0:local:Net[venet0:127.0.0.2/32:Unknown]]: Gathered candidate: Cand[:104624243:1:tcp:1518214911:127.0.0.2:33148:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:3:50:0]
  541. (basicportallocator.cc:762): Jingle:Port[0x248d5a0:audio:1:0:local:Net[venet0:127.0.0.2/32:Unknown]]: Port ready.
  542. (physicalsocketserver.cc:570): Socket::OPT_DSCP not supported.
  543. (p2ptransportchannel.cc:528): Jingle:Port[0x248d5a0:audio:1:0:local:Net[venet0:127.0.0.2/32:Unknown]]: SetOption(5, 0) failed: 0
  544. (basicportallocator.cc:778): Not yet signaling candidate because protocol is not yet enabled.
  545. (basicportallocator.cc:844): Jingle:Port[0x248d5a0:audio:1:0:local:Net[venet0:127.0.0.2/32:Unknown]]: Port completed gathering candidates.
  546. (basicportallocator.cc:894): Signaling candidate because protocol was enabled: Cand[:104624243:1:tcp:1518214911:127.0.0.2:33148:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:3:50:0]
  547. (basicportallocator.cc:1143): Jingle:Net[lo:::1/128:Loopback]: Allocation Phase=Tcp
  548. (port.cc:214): Jingle:Port[0x248ef30::1:0:local:Net[lo:::1/128:Loopback]]: Port created with network cost 0
  549. [WARN] [<noname> - p???] io.antmedia.webrtc.WebSocketListener - onIceCandidate
  550. (basicportallocator.cc:698): Adding allocated port for audio
  551. (basicportallocator.cc:718): Jingle:Port[0x248ef30:audio:1:0:local:Net[lo:::1/128:Loopback]]: Added port to allocator
  552. (basicportallocator.cc:735): Jingle:Port[0x248ef30:audio:1:0:local:Net[lo:::1/128:Loopback]]: Gathered candidate: Cand[:1876313031:1:tcp:1518157055:[::1]:46688:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:2:0:0]
  553. (basicportallocator.cc:762): Jingle:Port[0x248ef30:audio:1:0:local:Net[lo:::1/128:Loopback]]: Port ready.
  554. (physicalsocketserver.cc:570): Socket::OPT_DSCP not supported.
  555. (p2ptransportchannel.cc:528): Jingle:Port[0x248ef30:audio:1:0:local:Net[lo:::1/128:Loopback]]: SetOption(5, 0) failed: 0
  556. (basicportallocator.cc:778): Not yet signaling candidate because protocol is not yet enabled.
  557. (basicportallocator.cc:844): Jingle:Port[0x248ef30:audio:1:0:local:Net[lo:::1/128:Loopback]]: Port completed gathering candidates.
  558. (basicportallocator.cc:894): Signaling candidate because protocol was enabled: Cand[:1876313031:1:tcp:1518157055:[::1]:46688:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:2:0:0]
  559. [WARN] [<noname> - p???] io.antmedia.webrtc.WebSocketListener - onIceCandidate
  560. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  561. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  562. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  563. (basicportallocator.cc:1143): Jingle:Net[lo:127.0.0.0/8:Loopback]: Allocation Phase=Tcp
  564. (port.cc:214): Jingle:Port[0x2490300::1:0:local:Net[lo:127.0.0.0/8:Loopback]]: Port created with network cost 0
  565. (basicportallocator.cc:698): Adding allocated port for audio
  566. (basicportallocator.cc:718): Jingle:Port[0x2490300:audio:1:0:local:Net[lo:127.0.0.0/8:Loopback]]: Added port to allocator
  567. (basicportallocator.cc:735): Jingle:Port[0x2490300:audio:1:0:local:Net[lo:127.0.0.0/8:Loopback]]: Gathered candidate: Cand[:344579997:1:tcp:1518083839:127.0.0.1:39733:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:1:0:0]
  568. (basicportallocator.cc:762): Jingle:Port[0x2490300:audio:1:0:local:Net[lo:127.0.0.0/8:Loopback]]: Port ready.
  569. (physicalsocketserver.cc:570): Socket::OPT_DSCP not supported.
  570. (p2ptransportchannel.cc:528): Jingle:Port[0x2490300:audio:1:0:local:Net[lo:127.0.0.0/8:Loopback]]: SetOption(5, 0) failed: 0
  571. (basicportallocator.cc:778): Not yet signaling candidate because protocol is not yet enabled.
  572. (basicportallocator.cc:844): Jingle:Port[0x2490300:audio:1:0:local:Net[lo:127.0.0.0/8:Loopback]]: Port completed gathering candidates.
  573. (basicportallocator.cc:894): Signaling candidate because protocol was enabled: Cand[:344579997:1:tcp:1518083839:127.0.0.1:39733:local::0:BSyN:RAWiPuyn/8wqcRGcwgvpCch0:1:0:0]
  574. (messagequeue.cc:527): Message took 83ms to dispatch. Posted from: OnMessage@../../webrtc/p2p/client/basicportallocator.cc:1173
  575. (basicportallocator.cc:1143): Jingle:Net[venet0:0:51.254.59.155/32:Unknown]: Allocation Phase=SslTcp
  576. (basicportallocator.cc:1143): Jingle:Net[venet0:127.0.0.2/32:Unknown]: Allocation Phase=SslTcp
  577. (basicportallocator.cc:1143): Jingle:Net[lo:::1/128:Loopback]: Allocation Phase=SslTcp
  578. [WARN] [<noname> - p???] io.antmedia.webrtc.WebSocketListener - onIceCandidate
  579. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  580. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  581. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  582. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  583. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  584. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  585. (file_audio_device.cc:504): PlayoutWarning
  586. (file_audio_device.cc:508): PlayoutError
  587. (file_audio_device.cc:512): RecordingWarning
  588. (file_audio_device.cc:516): RecordingError
  589. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  590. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  591. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  592. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  593. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  594. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  595. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  596. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  597. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  598. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  599. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  600. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  601. (basicportallocator.cc:1143): Jingle:Net[lo:127.0.0.0/8:Loopback]: Allocation Phase=SslTcp
  602. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  603. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  604. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  605. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  606. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  607. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  608. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  609. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  610. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  611. (stunport.cc:506): sendto : [0x00000016] Invalid argument
  612. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  613. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  614. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  615. (stunport.cc:506): sendto : [0x00000016] Invalid argument
  616. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  617. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  618. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  619. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  620. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  621. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  622. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  623. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  624. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  625. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  626. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  627. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  628. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  629. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  630. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  631. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  632. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  633. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  634. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  635. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  636. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  637. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  638. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  639. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  640. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  641. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  642. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  643. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  644. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  645. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  646. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  647. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  648. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  649. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  650. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  651. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  652. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  653. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  654. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  655. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  656. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  657. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  658. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  659. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  660. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  661. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  662. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  663. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  664. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  665. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  666. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  667. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  668. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  669. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  670. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  671. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  672. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  673. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  674. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  675. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  676. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  677. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  678. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  679. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  680. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  681. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  682. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  683. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  684. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  685. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  686. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  687. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  688. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  689. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  690. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  691. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  692. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  693. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  694. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  695. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  696. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  697. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  698. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  699. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  700. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  701. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  702. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  703. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  704. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  705. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  706. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  707. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  708. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  709. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  710. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  711. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  712. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
  713. at io.antmedia.webrtc.receiver.FFmpegFrameRecorder.recordSamples(FFmpegFrameRecorder.java:1055)
  714. at io.antmedia.webrtc.receiver.ReceiverConnectionContext$1.onData(ReceiverConnectionContext.java:91)
  715. io.antmedia.webrtc.receiver.FrameRecorder$Exception: No audio output stream (Is audioChannels > 0 and has start() been called?)
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