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{ "shards": [ { "bot_dimensions": [ { "key": "caches", "value": [ "swarming_module_cache_vpython", "xcode_ios_10b61", "xcode_ios_10l232m", "xcode_ios_10o45e" ] }, { "key": "cores", "value": [ "8" ] }, { "key": "cpu", "value": [ "x86", "x86-64", "x86-64-E5-2697_v2" ] }, { "key": "gce", "value": [ "0" ] }, { "key": "gpu", "value": [ "none" ] }, { "key": "id", "value": [ "vm857-m4" ] }, { "key": "mac_model", "value": [ "VMware7,1" ] }, { "key": "machine_type", "value": [ "n1-highcpu-8" ] }, { "key": "os", "value": [ "Mac", "Mac-10.13", "Mac-10.13.6" ] }, { "key": "pool", "value": [ "Chrome" ] }, { "key": "python", "value": [ "2.7.10" ] }, { "key": "server_version", "value": [ "4260-fb32526" ] }, { "key": "ssd", "value": [ "1" ] }, { "key": "zone", "value": [ "us", "us-golo", "us-golo-4" ] } ], "bot_id": "vm857-m4", "bot_version": "84fce6b5d9235ac606544652bb2ef7411c74b0807d26a0712f3ff2c315b69522", "cipd_pins": { "client_package": { "package_name": "infra/tools/cipd/mac-amd64", "version": "1vQ99wQk82cRyVYu18rqMpXtb7XZ3WHvL3HFXTcLHN4C" }, "packages": [ { "package_name": "infra/tools/mac_toolchain/mac-amd64", "path": ".", "version": "1065f97353165dfafa89b64bfdaf9b424d705bc7" }, { "package_name": "infra/python/cpython/mac-amd64", "path": ".swarming_module", "version": "6dd10e31dc5d4cbb3c8f42a6fbd9485aeeb9ef0c" }, { "package_name": "infra/tools/luci/logdog/butler/mac-amd64", "path": ".swarming_module", "version": "3138aa2054ab30eac6f7e1e589e590571ae44692" }, { "package_name": "infra/tools/luci/vpython-native/mac-amd64", "path": ".swarming_module", "version": "QDLPZ3UAgZNoR5d69kckoahXHUl1oKiq5UVzhbr3X4EC" }, { "package_name": "infra/tools/luci/vpython/mac-amd64", "path": ".swarming_module", "version": "up_SZsefNVjCYV99Mh2FddLgkRCR_FxXo_Qn2bhROIsC" } ] }, "completed_ts": "2019-04-17T15:00:44.784200", "costs_usd": [ 0.023940506315561265 ], "created_ts": "2019-04-17T14:58:23.344190", "duration": 46.80366110801697, "modified_ts": "2019-04-17T15:00:44.784200", "name": "apprtcmobile_tests (iPhone 6s iOS 11.4)/Mac-10.13/59830afca4/iOS64 Sim Debug (iOS 11)/829", "output": "[I 2019-04-17 07:59:07] About to install Xcode 10l232m in Xcode.app for ios\n[07:59:12:INFO] Using Xcode version 10.0 build 10L232m at /b/s/w/ir/Xcode.app/Contents/Developer\nDevice wiped.\n[07:59:17:DEBUG] Setting sigterm handler.\n[07:59:19:INFO] User defaults from command line:\n[07:59:19:INFO] IDETestRunSpecificationPath = /var/folders/2j/22s2gz0s7hn48k32d47clxf80000gm/T/71309B97-5B1A-46EC-B9C8-E7EEAA9EA94A\n[07:59:19:INFO] \n[07:59:19:INFO] Testing started on 'iPhone 6s'\n[07:59:19:INFO] 2019-04-17 07:59:19.510 xcodebuild[890:11879] IDETestOperationsObserverDebug: Writing diagnostic log for test session to:\n[07:59:19:INFO] /Users/chrome-bot/Library/Developer/Xcode/DerivedData/temporary-artzivirmkrfysewqiyzwhklfbfy/Logs/Test/Test-Transient Testing-2019.04.17_07-59-19--0700.xcresult/1_Test/Diagnostics/apprtcmobile_tests_module-F6CF2FAC-455F-4953-9093-6EE69111A069/apprtcmobile_tests_module-6EF39412-3D86-420D-BA43-4ABDC73FE9A6/Session-apprtcmobile_tests_module-2019-04-17_075919-HsXOkr.log\n[07:59:19:INFO] 2019-04-17 07:59:19.510 xcodebuild[890:11868] [MT] IDETestOperationsObserverDebug: (9F84C5FC-B6B3-43F5-85BA-3F089C671CB7) Beginning test session apprtcmobile_tests_module-9F84C5FC-B6B3-43F5-85BA-3F089C671CB7 at 2019-04-17 07:59:19.510 with Xcode 10L232m on target <DVTiPhoneSimulator: 0x7f9973148730> {\n[07:59:19:INFO] \t\tSimDevice: iPhone 6s (107B8296-D0C3-4CD8-868A-616F96DF22CC, iOS 11.4, Shutdown)\n[07:59:19:INFO] } (11.4 (15F79))\n[07:59:36:INFO] objc[1010]: Class RTCDispatcher is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114415a70) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/apprtcmobile_tests (0x10e855330). One of the two will be used. Which one is undefined.\n[07:59:36:INFO] objc[1010]: Class RTCCameraPreviewView is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114415a98) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/apprtcmobile_tests (0x10e855358). One of the two will be used. Which one is undefined.\n[07:59:36:INFO] objc[1010]: Class RTCEncodedImage is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114414940) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/apprtcmobile_tests (0x10e8553d0). One of the two will be used. Which one is undefined.\n[07:59:36:INFO] objc[1010]: Class RTCRtpFragmentationHeader is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114414990) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/apprtcmobile_tests (0x10e855420). One of the two will be used. Which one is undefined.\n[07:59:36:INFO] objc[1010]: Class RTCVideoCapturer is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x1144149b8) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/apprtcmobile_tests (0x10e855448). One of the two will be used. Which one is undefined.\n[07:59:36:INFO] objc[1010]: Class RTCVideoCodecInfo is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114414a08) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/apprtcmobile_tests (0x10e855498). One of the two will be used. Which one is undefined.\n[07:59:36:INFO] objc[1010]: Class RTCVideoEncoderQpThresholds is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114414a58) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/apprtcmobile_tests (0x10e8554e8). One of the two will be used. Which one is undefined.\n[07:59:36:INFO] objc[1010]: Class RTCVideoEncoderSettings is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114414ad0) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/apprtcmobile_tests (0x10e855560). One of the two will be used. Which one is undefined.\n[07:59:36:INFO] objc[1010]: Class RTCVideoFrame is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114414af8) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/107B8296-D0C3-4CD8-868A-616F96DF22CC/data/Containers/Bundle/Application/43866A63-7EE4-4959-9ED9-D5BDF224D884/apprtcmobile_tests.app/apprtcmobile_tests (0x10e855588). One of the two will be used. Which one is undefined.\n[07:59:36:INFO] 2019-04-17 07:59:36.362 apprtcmobile_tests[1010:13312] +[CATransaction synchronize] called within transaction\n[07:59:36:INFO] Test Suite 'All tests' started at 2019-04-17 07:59:36.548\n[07:59:36:INFO] Test Suite 'apprtcmobile_tests.app' started at 2019-04-17 07:59:36.549\n[07:59:36:INFO] Test Suite 'ARDAppClientTest' started at 2019-04-17 07:59:36.549\n[07:59:36:INFO] Test Case '-[ARDAppClientTest testSession]' started.\n[07:59:36:INFO] [000:000] [771] (audio_device_module.mm:21): CreateAudioDeviceModule\n[07:59:36:INFO] [000:001] [771] (audio_device_module_ios.mm:44): current platform is IOS\n[07:59:36:INFO] [000:002] [771] (audio_device_module_ios.mm:45): iPhone Audio APIs will be utilized.\n[07:59:36:INFO] [000:020] [771] (audio_processing_impl.cc:438): Capture analyzer activated: 0\n[07:59:36:INFO] Capture post processor activated: 0\n[07:59:36:INFO] Render pre processor activated: 0\n[07:59:36:INFO] [000:023] [771] (webrtc_video_engine.cc:470): WebRtcVideoEngine::WebRtcVideoEngine()\n[07:59:36:INFO] [000:027] [771] (webrtc_voice_engine.cc:196): WebRtcVoiceEngine::WebRtcVoiceEngine\n[07:59:36:INFO] [000:036] [16147] (webrtc_voice_engine.cc:219): WebRtcVoiceEngine::Init\n[07:59:36:INFO] [000:036] [16147] (webrtc_voice_engine.cc:227): Supported send codecs in order of preference:\n[07:59:36:INFO] [000:045] [16147] (webrtc_voice_engine.cc:230): opus/48000/2 { minptime=10 useinbandfec=1 } (111)\n[07:59:36:INFO] [000:045] [16147] (webrtc_voice_engine.cc:230): ISAC/16000/1 (103)\n[07:59:36:INFO] [000:045] [16147] (webrtc_voice_engine.cc:230): ISAC/32000/1 (104)\n[07:59:36:INFO] [000:045] [16147] (webrtc_voice_engine.cc:230): G722/8000/1 (9)\n[07:59:36:INFO] [000:045] [16147] (webrtc_voice_engine.cc:230): ILBC/8000/1 (102)\n[07:59:36:INFO] [000:045] [16147] (webrtc_voice_engine.cc:230): PCMU/8000/1 (0)\n[07:59:36:INFO] [000:045] [16147] (webrtc_voice_engine.cc:230): PCMA/8000/1 (8)\n[07:59:36:INFO] [000:045] [16147] (webrtc_voice_engine.cc:230): CN/32000/1 (106)\n[07:59:36:INFO] [000:045] [16147] (webrtc_voice_engine.cc:230): CN/16000/1 (105)\n[07:59:36:INFO] [000:045] [16147] (webrtc_voice_engine.cc:230): CN/8000/1 (13)\n[07:59:36:INFO] [000:045] [16147] (webrtc_voice_engine.cc:230): telephone-event/48000/1 (110)\n[07:59:36:INFO] [000:045] [16147] (webrtc_voice_engine.cc:230): telephone-event/32000/1 (112)\n[07:59:36:INFO] [000:046] [16147] (webrtc_voice_engine.cc:230): telephone-event/16000/1 (113)\n[07:59:36:INFO] [000:046] [16147] (webrtc_voice_engine.cc:230): telephone-event/8000/1 (126)\n[07:59:36:INFO] [000:046] [16147] (webrtc_voice_engine.cc:233): Supported recv codecs in order of preference:\n[07:59:36:INFO] [000:046] [16147] (webrtc_voice_engine.cc:236): opus/48000/2 { minptime=10 useinbandfec=1 } (111)\n[07:59:36:INFO] [000:046] [16147] (webrtc_voice_engine.cc:236): ISAC/16000/1 (103)\n[07:59:36:INFO] [000:046] [16147] (webrtc_voice_engine.cc:236): ISAC/32000/1 (104)\n[07:59:36:INFO] [000:046] [16147] (webrtc_voice_engine.cc:236): G722/8000/1 (9)\n[07:59:36:INFO] [000:046] [16147] (webrtc_voice_engine.cc:236): ILBC/8000/1 (102)\n[07:59:36:INFO] [000:046] [16147] (webrtc_voice_engine.cc:236): PCMU/8000/1 (0)\n[07:59:36:INFO] [000:047] [16147] (webrtc_voice_engine.cc:236): PCMA/8000/1 (8)\n[07:59:36:INFO] [000:047] [16147] (webrtc_voice_engine.cc:236): CN/32000/1 (106)\n[07:59:36:INFO] [000:047] [16147] (webrtc_voice_engine.cc:236): CN/16000/1 (105)\n[07:59:36:INFO] [000:047] [16147] (webrtc_voice_engine.cc:236): CN/8000/1 (13)\n[07:59:36:INFO] [000:047] [16147] (webrtc_voice_engine.cc:236): telephone-event/48000/1 (110)\n[07:59:36:INFO] [000:047] [16147] (webrtc_voice_engine.cc:236): telephone-event/32000/1 (112)\n[07:59:36:INFO] [000:047] [16147] (webrtc_voice_engine.cc:236): telephone-event/16000/1 (113)\n[07:59:36:INFO] [000:047] [16147] (webrtc_voice_engine.cc:236): telephone-event/8000/1 (126)\n[07:59:36:INFO] [000:047] [16147] (audio_device_module_ios.mm:69): Init\n[07:59:36:INFO] [000:048] [16147] (audio_device_buffer.cc:63): AudioDeviceBuffer::ctor\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:117): AudioDeviceIOS::ctor<NSThread: 0x610000068300>{number = 3, name = (null)}\n[07:59:36:INFO] [000:049] [16147] (audio_device_module_ios.mm:49): AttachAudioBuffer\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:133): AudioDeviceIOS::AttachAudioBuffer\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:140): AudioDeviceIOS::Init\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:85): LogDeviceInfo\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:87): system name: iOS\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:88): system version: 11.4\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:89): device type: iPhone\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:90): device name: x86_64\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:91): process name: apprtcmobile_tests\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:92): process ID: 1010\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:93): OS version: Version 11.4 (Build 15F79)\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:94): processing cores: 8\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:95): low power mode: 0\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:97): TARGET_IPHONE_SIMULATOR is defined\n[07:59:36:INFO] [000:049] [16147] (audio_device_ios.mm:99): DeviceIsSimulator: 1\n[07:59:36:INFO] [000:050] [16147] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer\n[07:59:36:INFO] [000:051] [16147] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000)\n[07:59:36:INFO] [000:051] [16147] (audio_device_buffer.cc:206): SetPlayoutChannels(1)\n[07:59:36:INFO] [000:051] [16147] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000)\n[07:59:36:INFO] [000:051] [16147] (audio_device_buffer.cc:200): SetRecordingChannels(1)\n[07:59:36:INFO] [000:051] [16147] (audio_device_module_ios.mm:406): SetPlayoutDevice(0)\n[07:59:36:INFO] [000:051] [16147] (audio_device_ios.mm:1006): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetPlayoutDevice(uint16_t): Not implemented\n[07:59:36:INFO] [000:051] [16147] (audio_device_module_ios.mm:108): InitSpeaker\n[07:59:36:INFO] [000:051] [16147] (audio_device_module_ios.mm:310): StereoPlayoutIsAvailable\n[07:59:36:INFO] [000:051] [16147] (audio_device_module_ios.mm:317): output: 0\n[07:59:36:INFO] [000:051] [16147] (audio_device_module_ios.mm:322): SetStereoPlayout(0)\n[07:59:36:INFO] [000:051] [16147] (audio_device_ios.mm:1059): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetStereoPlayout(bool): Not implemented\n[07:59:36:INFO] [000:051] [16147] (audio_device_module_ios.mm:330): stereo playout is not supported\n[07:59:36:INFO] [000:051] [16147] (adm_helpers.cc:57): Failed to set stereo playout mode.\n[07:59:36:INFO] [000:051] [16147] (audio_device_module_ios.mm:468): SetRecordingDevice(0)\n[07:59:36:INFO] [000:051] [16147] (audio_device_ios.mm:1108): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetRecordingDevice(uint16_t): Not implemented\n[07:59:36:INFO] [000:051] [16147] (audio_device_module_ios.mm:114): InitMicrophone\n[07:59:36:INFO] [000:052] [16147] (audio_device_module_ios.mm:277): StereoRecordingIsAvailable\n[07:59:36:INFO] [000:052] [16147] (audio_device_module_ios.mm:284): output: 0\n[07:59:36:INFO] [000:052] [16147] (audio_device_module_ios.mm:289): SetStereoRecording(0)\n[07:59:36:INFO] [000:052] [16147] (adm_helpers.cc:77): Failed to set stereo recording mode.\n[07:59:36:INFO] [000:052] [16147] (apm_helpers.cc:32): Setting AGC mode to 2\n[07:59:36:INFO] [000:052] [16147] (audio_processing_impl.cc:727): Highpass filter activated: 0\n[07:59:36:INFO] [000:052] [16147] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:36:INFO] [000:052] [16147] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:36:INFO] [000:054] [16147] (audio_device_module_ios.mm:581): RegisterAudioCallback\n[07:59:36:INFO] [000:054] [16147] (audio_device_buffer.cc:81): RegisterAudioCallback\n[07:59:36:INFO] [000:055] [16147] (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, }\n[07:59:36:INFO] [000:055] [16147] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[07:59:36:INFO] [000:055] [16147] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[07:59:36:INFO] [000:055] [16147] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[07:59:36:INFO] [000:055] [16147] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[07:59:36:INFO] [000:055] [16147] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[07:59:36:INFO] [000:055] [16147] (audio_device_module_ios.mm:600): output: 0\n[07:59:36:INFO] [000:055] [16147] (audio_processing_impl.cc:727): Highpass filter activated: 0\n[07:59:36:INFO] [000:055] [16147] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:36:INFO] [000:055] [16147] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:36:INFO] [000:055] [16147] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[07:59:36:INFO] [000:055] [16147] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[07:59:36:INFO] [000:055] [16147] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[07:59:36:INFO] [000:055] [16147] (audio_device_module_ios.mm:616): output: 0\n[07:59:36:INFO] [000:055] [16147] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[07:59:36:INFO] [000:055] [16147] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[07:59:36:INFO] [000:055] [16147] (audio_device_module_ios.mm:632): output: 0\n[07:59:36:INFO] [000:055] [16147] (audio_processing_impl.cc:727): Highpass filter activated: 0\n[07:59:36:INFO] [000:056] [16147] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:36:INFO] [000:056] [16147] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:36:INFO] [000:056] [16147] (apm_helpers.cc:62): NS set to 0\n[07:59:36:INFO] [000:056] [16147] (webrtc_voice_engine.cc:447): Stereo swapping enabled? 0\n[07:59:36:INFO] [000:056] [16147] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[07:59:36:INFO] [000:056] [16147] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[07:59:36:INFO] [000:056] [16147] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[07:59:36:INFO] [000:056] [16147] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[07:59:36:INFO] [000:056] [16147] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[07:59:36:INFO] [000:056] [16147] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[07:59:36:INFO] [000:056] [16147] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[07:59:36:INFO] [000:056] [16147] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[07:59:36:INFO] [000:056] [16147] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[07:59:36:INFO] [000:056] [16147] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:36:INFO] [000:056] [16147] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:36:INFO] [000:056] [16147] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:36:INFO] (RTCLogging.mm:33): (ARDAppClient.m:270 -[ARDAppClient connectToRoomWithId:settings:isLoopback:]_block_invoke): Joined room:testRoom on room server.\n[07:59:36:INFO] [000:064] [771] (openssl_identity.cc:44): Making key pair\n[07:59:36:INFO] [000:205] [771] (openssl_identity.cc:92): Returning key pair\n[07:59:36:INFO] [000:205] [771] (openssl_certificate.cc:58): Making certificate for WebRTC\n[07:59:36:INFO] [000:212] [771] (openssl_certificate.cc:108): Returning certificate\n[07:59:36:INFO] [000:215] [771] (RTCCertificate.mm:63): CERT PEM\n[07:59:36:INFO] [000:215] [771] (RTCCertificate.mm:64): -----BEGIN CERTIFICATE-----\n[07:59:36:INFO] MIIBnjCCAQegAwIBAgIJAIxPdfgrlJ3sMA0GCSqGSIb3DQEBCwUAMBExDzANBgNV\n[07:59:36:INFO] BAMMBldlYlJUQzAeFw0xOTA0MTYxNDU5MzZaFw0xOTA0MTcxNTAxMTZaMBExDzAN\n[07:59:36:INFO] BgNVBAMMBldlYlJUQzCBnzANBgkqhkiG9w0BAQEFAAOBjQAwgYkCgYEAtDBSjvKM\n[07:59:36:INFO] /J5Y7b+erGXUXz5pYLsCctYuIs8dUqtG2yck1qXdWrMqyTcm7tONeVfFHpixQx2E\n[07:59:36:INFO] Lo9AM7OUpImCfVDCyI0jo4+Tnwalh7+85vFNh7b756CsraSwmVvZHPBur6521wNN\n[07:59:36:INFO] XdTh6eCb3OnnUk22GiOEA22zynUo4TSwQRUCAwEAATANBgkqhkiG9w0BAQsFAAOB\n[07:59:36:INFO] gQAN4J7oDua+F4MRmhG414XuyTszvSFITHiuZhcNoxK4Jb73gnarOhbA48FE1+AX\n[07:59:36:INFO] hx5A4ux9NLyVQqiwP/n7RuqVZbrEUA2ml10USz7w8YgfhTAP4b5FtiEiT/OmWUYT\n[07:59:36:INFO] yYvS4gOjfXmmD1O/jtz/QUhu67sD9W0qOVxk7mnEWrkUYg==\n[07:59:36:INFO] -----END CERTIFICATE-----\n[07:59:36:INFO] \n[07:59:36:INFO] [000:215] [771] (RTCConfiguration.mm:213): Have configured cert - using it.\n[07:59:36:INFO] [000:217] [771] (RTCConfiguration.mm:218): Created cert from PEM strings.\n[07:59:36:INFO] [000:223] [16147] (rtc_event_log_impl.cc:71): Creating legacy encoder for RTC event log.\n[07:59:36:INFO] [000:226] [16147] (peer_connection_factory.cc:407): Using default network controller factory\n[07:59:36:INFO] [000:231] [16147] (bitrate_prober.cc:66): Bandwidth probing enabled, set to inactive\n[07:59:36:INFO] [000:234] [16147] (paced_sender.cc:418): ProcessThreadAttached 0xd9f01c60\n[07:59:36:INFO] [000:235] [16147] (cpu_info.cc:49): Available number of cores: 8\n[07:59:36:INFO] [000:239] [16147] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85\n[07:59:36:INFO] [000:239] [16147] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating.\n[07:59:36:INFO] [000:258] [23563] (webrtc_session_description_factory.cc:314): Using certificate supplied to the constructor.\n[07:59:36:INFO] [000:260] [23563] (peer_connection.cc:1405): Adding audio transceiver in response to a call to AddTrack.\n[07:59:36:INFO] (RTCLogging.mm:33): (ARDAppClient.m:396 -[ARDAppClient peerConnectionShouldNegotiate:]): WARNING: Renegotiation needed but unimplemented.\n[07:59:36:INFO] [000:265] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60000000a1a0): created DTMF sender: RTCDtmfSender {\n[07:59:36:INFO] remainingTones:\n[07:59:36:INFO] duration: 0.100000 sec\n[07:59:36:INFO] interToneGap: 0.050000 sec\n[07:59:36:INFO] }\n[07:59:36:INFO] [000:265] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600000034640): created sender: RTCRtpSender {\n[07:59:36:INFO] senderId: ARDAMSa0\n[07:59:36:INFO] }\n[07:59:36:INFO] [000:266] [771] (RTCLogging.mm:33): (RTCFileVideoCapturer.m:103 -[RTCFileVideoCapturer stopCapture]): File capturer stopped.\n[07:59:36:INFO] [000:267] [23563] (peer_connection.cc:1405): Adding video transceiver in response to a call to AddTrack.\n[07:59:36:INFO] [000:267] [23563] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[07:59:36:INFO] [000:267] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x608000035360): created sender: RTCRtpSender {\n[07:59:36:INFO] senderId: ARDAMSv0\n[07:59:36:INFO] }\n[07:59:36:INFO] [000:268] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60c00000a6d0): created DTMF sender: RTCDtmfSender {\n[07:59:36:INFO] remainingTones:\n[07:59:36:INFO] duration: 0.100000 sec\n[07:59:36:INFO] interToneGap: 0.050000 sec\n[07:59:36:INFO] }\n[07:59:36:INFO] [000:268] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60c000038d40): created sender: RTCRtpSender {\n[07:59:36:INFO] senderId: ARDAMSa0\n[07:59:36:INFO] }\n[07:59:36:INFO] [000:269] [771] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x600000050650): created receiver: RTCRtpReceiver {\n[07:59:36:INFO] receiverId: 02e154fc-5824-4435-8f57-eb1d2c6dbfc1\n[07:59:36:INFO] }\n[07:59:36:INFO] [000:269] [771] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x60c0000543d0): created transceiver: RTCRtpTransceiver {\n[07:59:36:INFO] sender: RTCRtpSender {\n[07:59:36:INFO] senderId: ARDAMSa0\n[07:59:36:INFO] }\n[07:59:36:INFO] receiver: RTCRtpReceiver {\n[07:59:36:INFO] receiverId: 02e154fc-5824-4435-8f57-eb1d2c6dbfc1\n[07:59:36:INFO] }\n[07:59:36:INFO] }\n[07:59:36:INFO] [000:269] [23563] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[07:59:36:INFO] [000:269] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x604000037f20): created sender: RTCRtpSender {\n[07:59:36:INFO] senderId: ARDAMSv0\n[07:59:36:INFO] }\n[07:59:36:INFO] [000:269] [771] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x600000050680): created receiver: RTCRtpReceiver {\n[07:59:36:INFO] receiverId: ef7ff4ed-10d0-4351-8398-2efa37dc555c\n[07:59:36:INFO] }\n[07:59:36:INFO] [000:270] [771] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x6040000543d0): created transceiver: RTCRtpTransceiver {\n[07:59:36:INFO] sender: RTCRtpSender {\n[07:59:36:INFO] senderId: ARDAMSv0\n[07:59:36:INFO] }\n[07:59:36:INFO] receiver: RTCRtpReceiver {\n[07:59:36:INFO] receiverId: ef7ff4ed-10d0-4351-8398-2efa37dc555c\n[07:59:36:INFO] }\n[07:59:36:INFO] }\n[07:59:36:INFO] [000:275] [23563] (media_session.cc:335): Duplicate id found. Reassigning from 102 to 127\n[07:59:36:INFO] [000:275] [23563] (media_session.cc:335): Duplicate id found. Reassigning from 104 to 125\n[07:59:36:INFO] [000:275] [23563] (media_session.cc:335): Duplicate id found. Reassigning from 106 to 124\n[07:59:36:INFO] [000:275] [23563] (media_session.cc:335): Duplicate id found. Reassigning from 103 to 123\n[07:59:36:INFO] [000:275] [23563] (media_session.cc:335): Duplicate id found. Reassigning from 105 to 122\n[07:59:36:INFO] [000:276] [23563] (media_session.cc:335): Duplicate id found. Reassigning from 1 to 14\n[07:59:36:INFO] [000:276] [23563] (media_session.cc:335): Duplicate id found. Reassigning from 2 to 13\n[07:59:36:INFO] [000:276] [23563] (media_session.cc:335): Duplicate id found. Reassigning from 3 to 12\n[07:59:36:INFO] [000:276] [23563] (media_session.cc:335): Duplicate id found. Reassigning from 5 to 11\n[07:59:36:INFO] [000:282] [16147] (rtc_event_log_impl.cc:186): Starting WebRTC event log. (Timestamp, UTC) = (273218610, 1555513176844841).\n[07:59:36:INFO] [000:284] [771] (audio_device_module.mm:21): CreateAudioDeviceModule\n[07:59:36:INFO] [000:284] [771] (audio_device_module_ios.mm:44): current platform is IOS\n[07:59:36:INFO] [000:284] [771] (audio_device_module_ios.mm:45): iPhone Audio APIs will be utilized.\n[07:59:36:INFO] [000:285] [771] (audio_processing_impl.cc:438): Capture analyzer activated: 0\n[07:59:36:INFO] Capture post processor activated: 0\n[07:59:36:INFO] Render pre processor activated: 0\n[07:59:36:INFO] [000:285] [771] (webrtc_video_engine.cc:470): WebRtcVideoEngine::WebRtcVideoEngine()\n[07:59:36:INFO] [000:285] [771] (webrtc_voice_engine.cc:196): WebRtcVoiceEngine::WebRtcVoiceEngine\n[07:59:36:INFO] [000:286] [40455] (webrtc_voice_engine.cc:219): WebRtcVoiceEngine::Init\n[07:59:36:INFO] [000:286] [40455] (webrtc_voice_engine.cc:227): Supported send codecs in order of preference:\n[07:59:36:INFO] [000:286] [40455] (webrtc_voice_engine.cc:230): opus/48000/2 { minptime=10 useinbandfec=1 } (111)\n[07:59:36:INFO] [000:286] [40455] (webrtc_voice_engine.cc:230): ISAC/16000/1 (103)\n[07:59:36:INFO] [000:286] [40455] (webrtc_voice_engine.cc:230): ISAC/32000/1 (104)\n[07:59:36:INFO] [000:286] [40455] (webrtc_voice_engine.cc:230): G722/8000/1 (9)\n[07:59:36:INFO] [000:286] [40455] (webrtc_voice_engine.cc:230): ILBC/8000/1 (102)\n[07:59:36:INFO] [000:286] [40455] (webrtc_voice_engine.cc:230): PCMU/8000/1 (0)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:230): PCMA/8000/1 (8)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:230): CN/32000/1 (106)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:230): CN/16000/1 (105)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:230): CN/8000/1 (13)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:230): telephone-event/48000/1 (110)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:230): telephone-event/32000/1 (112)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:230): telephone-event/16000/1 (113)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:230): telephone-event/8000/1 (126)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:233): Supported recv codecs in order of preference:\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:236): opus/48000/2 { minptime=10 useinbandfec=1 } (111)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:236): ISAC/16000/1 (103)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:236): ISAC/32000/1 (104)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:236): G722/8000/1 (9)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:236): ILBC/8000/1 (102)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:236): PCMU/8000/1 (0)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:236): PCMA/8000/1 (8)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:236): CN/32000/1 (106)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:236): CN/16000/1 (105)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:236): CN/8000/1 (13)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:236): telephone-event/48000/1 (110)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:236): telephone-event/32000/1 (112)\n[07:59:36:INFO] [000:287] [40455] (webrtc_voice_engine.cc:236): telephone-event/16000/1 (113)\n[07:59:36:INFO] [000:288] [40455] (webrtc_voice_engine.cc:236): telephone-event/8000/1 (126)\n[07:59:36:INFO] [000:288] [40455] (audio_device_module_ios.mm:69): Init\n[07:59:36:INFO] [000:288] [40455] (audio_device_buffer.cc:63): AudioDeviceBuffer::ctor\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:117): AudioDeviceIOS::ctor<NSThread: 0x600000069400>{number = 4, name = (null)}\n[07:59:36:INFO] [000:288] [40455] (audio_device_module_ios.mm:49): AttachAudioBuffer\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:133): AudioDeviceIOS::AttachAudioBuffer\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:140): AudioDeviceIOS::Init\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:85): LogDeviceInfo\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:87): system name: iOS\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:88): system version: 11.4\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:89): device type: iPhone\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:90): device name: x86_64\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:91): process name: apprtcmobile_tests\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:92): process ID: 1010\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:93): OS version: Version 11.4 (Build 15F79)\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:94): processing cores: 8\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:95): low power mode: 0\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:97): TARGET_IPHONE_SIMULATOR is defined\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:99): DeviceIsSimulator: 1\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer\n[07:59:36:INFO] [000:288] [40455] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000)\n[07:59:36:INFO] [000:288] [40455] (audio_device_buffer.cc:206): SetPlayoutChannels(1)\n[07:59:36:INFO] [000:288] [40455] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000)\n[07:59:36:INFO] [000:288] [40455] (audio_device_buffer.cc:200): SetRecordingChannels(1)\n[07:59:36:INFO] [000:288] [40455] (audio_device_module_ios.mm:406): SetPlayoutDevice(0)\n[07:59:36:INFO] [000:288] [40455] (audio_device_ios.mm:1006): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetPlayoutDevice(uint16_t): Not implemented\n[07:59:36:INFO] [000:288] [40455] (audio_device_module_ios.mm:108): InitSpeaker\n[07:59:36:INFO] [000:288] [40455] (audio_device_module_ios.mm:310): StereoPlayoutIsAvailable\n[07:59:36:INFO] [000:288] [40455] (audio_device_module_ios.mm:317): output: 0\n[07:59:36:INFO] [000:289] [40455] (audio_device_module_ios.mm:322): SetStereoPlayout(0)\n[07:59:36:INFO] [000:289] [40455] (audio_device_ios.mm:1059): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetStereoPlayout(bool): Not implemented\n[07:59:36:INFO] [000:289] [40455] (audio_device_module_ios.mm:330): stereo playout is not supported\n[07:59:36:INFO] [000:289] [40455] (adm_helpers.cc:57): Failed to set stereo playout mode.\n[07:59:36:INFO] [000:289] [40455] (audio_device_module_ios.mm:468): SetRecordingDevice(0)\n[07:59:36:INFO] [000:289] [40455] (audio_device_ios.mm:1108): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetRecordingDevice(uint16_t): Not implemented\n[07:59:36:INFO] [000:289] [40455] (audio_device_module_ios.mm:114): InitMicrophone\n[07:59:36:INFO] [000:289] [40455] (audio_device_module_ios.mm:277): StereoRecordingIsAvailable\n[07:59:36:INFO] [000:289] [40455] (audio_device_module_ios.mm:284): output: 0\n[07:59:36:INFO] [000:289] [40455] (audio_device_module_ios.mm:289): SetStereoRecording(0)\n[07:59:36:INFO] [000:289] [40455] (adm_helpers.cc:77): Failed to set stereo recording mode.\n[07:59:36:INFO] [000:289] [40455] (apm_helpers.cc:32): Setting AGC mode to 2\n[07:59:36:INFO] [000:289] [40455] (audio_processing_impl.cc:727): Highpass filter activated: 0\n[07:59:36:INFO] [000:289] [40455] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:36:INFO] [000:289] [40455] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:36:INFO] [000:290] [40455] (audio_device_module_ios.mm:581): RegisterAudioCallback\n[07:59:36:INFO] [000:290] [40455] (audio_device_buffer.cc:81): RegisterAudioCallback\n[07:59:36:INFO] [000:290] [40455] (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, }\n[07:59:36:INFO] [000:290] [40455] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[07:59:36:INFO] [000:290] [40455] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[07:59:36:INFO] [000:290] [40455] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[07:59:36:INFO] [000:290] [40455] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[07:59:36:INFO] [000:290] [40455] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[07:59:36:INFO] [000:290] [40455] (audio_device_module_ios.mm:600): output: 0\n[07:59:36:INFO] [000:290] [40455] (audio_processing_impl.cc:727): Highpass filter activated: 0\n[07:59:36:INFO] [000:290] [40455] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:36:INFO] [000:290] [40455] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:36:INFO] [000:290] [40455] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[07:59:36:INFO] [000:290] [40455] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[07:59:36:INFO] [000:290] [40455] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[07:59:36:INFO] [000:290] [40455] (audio_device_module_ios.mm:616): output: 0\n[07:59:36:INFO] [000:290] [40455] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[07:59:36:INFO] [000:290] [40455] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[07:59:36:INFO] [000:291] [40455] (audio_device_module_ios.mm:632): output: 0\n[07:59:36:INFO] [000:291] [40455] (audio_processing_impl.cc:727): Highpass filter activated: 0\n[07:59:36:INFO] [000:291] [40455] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:36:INFO] [000:291] [40455] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:36:INFO] [000:291] [40455] (apm_helpers.cc:62): NS set to 0\n[07:59:36:INFO] [000:291] [40455] (webrtc_voice_engine.cc:447): Stereo swapping enabled? 0\n[07:59:36:INFO] [000:291] [40455] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[07:59:36:INFO] [000:291] [40455] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[07:59:36:INFO] [000:291] [40455] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[07:59:36:INFO] [000:291] [40455] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[07:59:36:INFO] [000:291] [40455] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[07:59:36:INFO] [000:291] [40455] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[07:59:36:INFO] [000:291] [40455] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[07:59:36:INFO] [000:291] [40455] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[07:59:36:INFO] [000:291] [40455] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[07:59:36:INFO] [000:291] [40455] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:36:INFO] [000:291] [40455] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:36:INFO] [000:291] [40455] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:36:INFO] (RTCLogging.mm:33): (ARDAppClient.m:270 -[ARDAppClient connectToRoomWithId:settings:isLoopback:]_block_invoke): Joined room:testRoom on room server.\n[07:59:36:INFO] [000:292] [771] (openssl_identity.cc:44): Making key pair\n[07:59:37:INFO] [000:560] [771] (openssl_identity.cc:92): Returning key pair\n[07:59:37:INFO] [000:560] [771] (openssl_certificate.cc:58): Making certificate for WebRTC\n[07:59:37:INFO] [000:563] [771] (openssl_certificate.cc:108): Returning certificate\n[07:59:37:INFO] [000:563] [771] (RTCCertificate.mm:63): CERT PEM\n[07:59:37:INFO] [000:563] [771] (RTCCertificate.mm:64): -----BEGIN CERTIFICATE-----\n[07:59:37:INFO] MIIBnjCCAQegAwIBAgIJAK/EkwoBFUpoMA0GCSqGSIb3DQEBCwUAMBExDzANBgNV\n[07:59:37:INFO] BAMMBldlYlJUQzAeFw0xOTA0MTYxNDU5MzZaFw0xOTA0MTcxNTAxMTZaMBExDzAN\n[07:59:37:INFO] BgNVBAMMBldlYlJUQzCBnzANBgkqhkiG9w0BAQEFAAOBjQAwgYkCgYEAzmUwmf+j\n[07:59:37:INFO] ce2UYuRhuI43Z6MYoYZwELmS2GodN/2rXudDElczmmnVFJzwMkXWGY1PLgJBxBsc\n[07:59:37:INFO] HE3YRp6JbQqMra5WpamepwXzmBUiDdtW2keW5fYmlFOH/KtjhWgaDLBIgHbWc/3k\n[07:59:37:INFO] G4NQY7mMSq9Go9FCRSlka3IU8YQ067xC4mkCAwEAATANBgkqhkiG9w0BAQsFAAOB\n[07:59:37:INFO] gQCzhey9w7TCy5Euae7voemq1qBpZYgyLca15uKJSes6qRtZasR1bRTJHBLocldg\n[07:59:37:INFO] 63X6+7KRzlpc7nCI8t3R9W5xZVtHltNTEEuEb/TUDt/DF/03/aidpjtNb4PGZcpk\n[07:59:37:INFO] Dna6YQWDoYwTXEA9fhGtMoWAfS+PAhp8JjRtIk133Yk9cw==\n[07:59:37:INFO] -----END CERTIFICATE-----\n[07:59:37:INFO] \n[07:59:37:INFO] [000:564] [771] (RTCConfiguration.mm:213): Have configured cert - using it.\n[07:59:37:INFO] [000:565] [771] (RTCConfiguration.mm:218): Created cert from PEM strings.\n[07:59:37:INFO] [000:565] [40455] (rtc_event_log_impl.cc:71): Creating legacy encoder for RTC event log.\n[07:59:37:INFO] [000:566] [40455] (peer_connection_factory.cc:407): Using default network controller factory\n[07:59:37:INFO] [000:566] [40455] (bitrate_prober.cc:66): Bandwidth probing enabled, set to inactive\n[07:59:37:INFO] [000:566] [40455] (paced_sender.cc:418): ProcessThreadAttached 0xd9f04b90\n[07:59:37:INFO] [000:566] [40455] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85\n[07:59:37:INFO] [000:566] [40455] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating.\n[07:59:37:INFO] [000:567] [41991] (webrtc_session_description_factory.cc:314): Using certificate supplied to the constructor.\n[07:59:37:INFO] [000:567] [41991] (peer_connection.cc:1405): Adding audio transceiver in response to a call to AddTrack.\n[07:59:37:INFO] (RTCLogging.mm:33): (ARDAppClient.m:396 -[ARDAppClient peerConnectionShouldNegotiate:]): WARNING: Renegotiation needed but unimplemented.\n[07:59:37:INFO] [000:568] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60400000af40): created DTMF sender: RTCDtmfSender {\n[07:59:37:INFO] remainingTones:\n[07:59:37:INFO] duration: 0.100000 sec\n[07:59:37:INFO] interToneGap: 0.050000 sec\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:568] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x604000038280): created sender: RTCRtpSender {\n[07:59:37:INFO] senderId: ARDAMSa0\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:568] [771] (RTCLogging.mm:33): (RTCFileVideoCapturer.m:103 -[RTCFileVideoCapturer stopCapture]): File capturer stopped.\n[07:59:37:INFO] [000:568] [41991] (peer_connection.cc:1405): Adding video transceiver in response to a call to AddTrack.\n[07:59:37:INFO] [000:569] [41991] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[07:59:37:INFO] [000:569] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x6040000382a0): created sender: RTCRtpSender {\n[07:59:37:INFO] senderId: ARDAMSv0\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:569] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60c00000a720): created DTMF sender: RTCDtmfSender {\n[07:59:37:INFO] remainingTones:\n[07:59:37:INFO] duration: 0.100000 sec\n[07:59:37:INFO] interToneGap: 0.050000 sec\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:570] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x61400023c2c0): created sender: RTCRtpSender {\n[07:59:37:INFO] senderId: ARDAMSa0\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:570] [771] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x6080000512b0): created receiver: RTCRtpReceiver {\n[07:59:37:INFO] receiverId: aac8edf5-3432-4199-bf0a-f872af25c469\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:570] [771] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x614000059890): created transceiver: RTCRtpTransceiver {\n[07:59:37:INFO] sender: RTCRtpSender {\n[07:59:37:INFO] senderId: ARDAMSa0\n[07:59:37:INFO] }\n[07:59:37:INFO] receiver: RTCRtpReceiver {\n[07:59:37:INFO] receiverId: aac8edf5-3432-4199-bf0a-f872af25c469\n[07:59:37:INFO] }\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:570] [41991] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[07:59:37:INFO] [000:570] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x61400023c100): created sender: RTCRtpSender {\n[07:59:37:INFO] senderId: ARDAMSv0\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:570] [771] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x610000050650): created receiver: RTCRtpReceiver {\n[07:59:37:INFO] receiverId: c9a133f5-63d9-44ca-be2f-40c21e6273a6\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:571] [771] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x6140000598c0): created transceiver: RTCRtpTransceiver {\n[07:59:37:INFO] sender: RTCRtpSender {\n[07:59:37:INFO] senderId: ARDAMSv0\n[07:59:37:INFO] }\n[07:59:37:INFO] receiver: RTCRtpReceiver {\n[07:59:37:INFO] receiverId: c9a133f5-63d9-44ca-be2f-40c21e6273a6\n[07:59:37:INFO] }\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:571] [40455] (rtc_event_log_impl.cc:186): Starting WebRTC event log. (Timestamp, UTC) = (273507879, 1555513177134109).\n[07:59:37:INFO] [000:573] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[07:59:37:INFO] [000:576] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[07:59:37:INFO] [000:584] [23311] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds.\n[07:59:37:INFO] [000:584] [23311] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds\n[07:59:37:INFO] [000:584] [23311] (p2p_transport_channel.cc:511): Set ping most likely connection to 0\n[07:59:37:INFO] [000:584] [23311] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500\n[07:59:37:INFO] [000:584] [23311] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0\n[07:59:37:INFO] [000:584] [23311] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000\n[07:59:37:INFO] [000:584] [23311] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000\n[07:59:37:INFO] [000:585] [23311] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0\n[07:59:37:INFO] [000:586] [23311] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport db905290\n[07:59:37:INFO] [000:586] [23311] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds.\n[07:59:37:INFO] [000:586] [23311] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds\n[07:59:37:INFO] [000:586] [23311] (p2p_transport_channel.cc:511): Set ping most likely connection to 0\n[07:59:37:INFO] [000:586] [23311] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500\n[07:59:37:INFO] [000:586] [23311] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0\n[07:59:37:INFO] [000:586] [23311] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000\n[07:59:37:INFO] [000:586] [23311] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000\n[07:59:37:INFO] [000:586] [23311] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 1 transport 0\n[07:59:37:INFO] [000:587] [23311] (dtls_srtp_transport.cc:66): Setting RTP Transport on 1 transport d9e2df30\n[07:59:37:INFO] [000:587] [23311] (p2p_transport_channel.cc:432): Set ICE ufrag: 2M90 pwd: RBBwMJj6gFR0wFgy19+3OH8o on transport 0\n[07:59:37:INFO] [000:587] [23311] (p2p_transport_channel.cc:432): Set ICE ufrag: 2M90 pwd: RBBwMJj6gFR0wFgy19+3OH8o on transport 1\n[07:59:37:INFO] [000:588] [16147] (webrtc_voice_engine.cc:1502): Setting voice channel options: AudioOptions {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, }\n[07:59:37:INFO] [000:588] [16147] (webrtc_voice_engine.cc:309): WebRtcVoiceEngine::ApplyOptions: AudioOptions {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, }\n[07:59:37:INFO] [000:588] [16147] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[07:59:37:INFO] [000:588] [16147] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[07:59:37:INFO] [000:588] [16147] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[07:59:37:INFO] [000:588] [16147] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[07:59:37:INFO] [000:588] [16147] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [000:588] [16147] (audio_device_module_ios.mm:600): output: 0\n[07:59:37:INFO] [000:588] [16147] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:37:INFO] [000:588] [16147] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:37:INFO] [000:588] [16147] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:37:INFO] [000:588] [16147] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[07:59:37:INFO] [000:588] [16147] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[07:59:37:INFO] [000:588] [16147] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [000:588] [16147] (audio_device_module_ios.mm:616): output: 0\n[07:59:37:INFO] [000:588] [16147] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[07:59:37:INFO] [000:588] [16147] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [000:588] [16147] (audio_device_module_ios.mm:632): output: 0\n[07:59:37:INFO] [000:588] [16147] (apm_helpers.cc:62): NS set to 0\n[07:59:37:INFO] [000:588] [16147] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[07:59:37:INFO] [000:588] [16147] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[07:59:37:INFO] [000:588] [16147] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[07:59:37:INFO] [000:588] [16147] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[07:59:37:INFO] [000:588] [16147] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[07:59:37:INFO] [000:588] [16147] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[07:59:37:INFO] [000:588] [16147] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[07:59:37:INFO] [000:588] [16147] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[07:59:37:INFO] [000:588] [16147] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[07:59:37:INFO] [000:588] [16147] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:37:INFO] [000:588] [16147] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:37:INFO] [000:588] [16147] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:37:INFO] [000:589] [16147] (webrtc_voice_engine.cc:1520): Set voice channel options. Current options: AudioOptions {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, }\n[07:59:37:INFO] [000:589] [16147] (channel.cc:144): Created channel for 0\n[07:59:37:INFO] [000:591] [16147] (channel.cc:212): BaseChannel::Init_w, media_transport=0\n[07:59:37:INFO] [000:591] [16147] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [000:592] [16147] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [000:592] [16147] (webrtc_video_engine.cc:483): CreateMediaChannel. Options: VideoOptions {}\n[07:59:37:INFO] [000:593] [16147] (channel.cc:144): Created channel for 1\n[07:59:37:INFO] [000:594] [16147] (channel.cc:212): BaseChannel::Init_w, media_transport=0\n[07:59:37:INFO] [000:594] [16147] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [000:594] [16147] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [000:594] [23563] (peer_connection.cc:4173): Session: 892703080514118717 Old state: kStable New state: kHaveLocalOffer\n[07:59:37:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 1\n[07:59:37:INFO] [000:594] [16147] (channel.cc:871): Setting local voice description\n[07:59:37:INFO] [000:595] [16147] (webrtc_voice_engine.cc:1341): 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[102:ILBC: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}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}]}\n[07:59:37:INFO] [000:595] [16147] (webrtc_voice_engine.cc:1530): Setting receive voice codecs.\n[07:59:37:INFO] [000:596] [16147] (webrtc_voice_engine.cc:1801): AddSendStream: {id:ARDAMSa0;ssrcs:[2609444602];ssrc_groups:;cname:SVCyZMBheTeYRPl/;stream_ids:ARDAMS;}\n[07:59:37:INFO] [000:603] [16147] (neteq_impl.cc:123): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=200, min_delay_ms=0, enable_fast_accelerate=false, enable_muted_state=false, enable_rtx_handling=false\n[07:59:37:INFO] [000:605] [16147] (audio_coding_module.cc:333): Created\n[07:59:37:INFO] [000:609] [16147] (channel_send.cc:708): Not setting media_transport_ rate observers.\n[07:59:37:INFO] [000:610] [16147] (audio_send_stream.cc:142): AudioSendStream: 2609444602\n[07:59:37:INFO] [000:610] [16147] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2609444602, extmap-allow-mixed: false, extensions: [], c_name: SVCyZMBheTeYRPl/}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: <unset>}\n[07:59:37:INFO] [000:612] [16147] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [000:612] [16147] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [000:612] [16147] (channel.cc:676): Add send stream ssrc: 2609444602\n[07:59:37:INFO] [000:612] [16147] (channel.cc:863): Changing voice state, recv=0 send=0\n[07:59:37:INFO] [000:612] [16147] (channel.cc:1009): Setting local video description\n[07:59:37:INFO] [000:613] [16147] (webrtc_video_engine.cc:956): SetRecvParameters: {codecs: [VideoCodec[96:H264], VideoCodec[97:rtx], VideoCodec[98:H264], VideoCodec[99:rtx], VideoCodec[100:VP8], VideoCodec[101:rtx], VideoCodec[127:VP9], VideoCodec[123:rtx], VideoCodec[125:red], VideoCodec[122:rtx], VideoCodec[124:ulpfec]], extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {uri: urn:3gpp:video-orientation, id: 12}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}]}\n[07:59:37:INFO] [000:614] [16147] (webrtc_video_engine.cc:971): Changing recv codecs from {VideoCodec[96:H264], VideoCodec[98:H264], VideoCodec[100:VP8], VideoCodec[102:VP9]} to {VideoCodec[96:H264], VideoCodec[98:H264], VideoCodec[100:VP8], VideoCodec[127:VP9]}\n[07:59:37:INFO] [000:614] [16147] (webrtc_video_engine.cc:1072): AddSendStream: {id:ARDAMSv0;ssrcs:[2869273597,673344836];ssrc_groups:{semantics:FID;ssrcs:[2869273597,673344836]};cname:SVCyZMBheTeYRPl/;stream_ids:ARDAMS;}\n[07:59:37:INFO] [000:614] [16147] (webrtc_video_engine.cc:1111): SetLocalSsrc on all the receive streams because we added a send stream.\n[07:59:37:INFO] [000:614] [16147] (channel.cc:676): Add send stream ssrc: 2869273597\n[07:59:37:INFO] [000:614] [16147] (channel.cc:996): Changing video state, send=0\n[07:59:37:INFO] [000:615] [23563] (peer_connection.cc:5512): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport.\n[07:59:37:INFO] [000:615] [16147] (webrtc_voice_engine.cc:1502): Setting voice channel options: AudioOptions {}\n[07:59:37:INFO] [000:615] [16147] (webrtc_voice_engine.cc:309): WebRtcVoiceEngine::ApplyOptions: AudioOptions {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, }\n[07:59:37:INFO] [000:615] [16147] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[07:59:37:INFO] [000:615] [16147] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[07:59:37:INFO] [000:615] [16147] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[07:59:37:INFO] [000:615] [16147] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[07:59:37:INFO] [000:615] [16147] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [000:615] [16147] (audio_device_module_ios.mm:600): output: 0\n[07:59:37:INFO] [000:615] [16147] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:37:INFO] [000:615] [16147] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:37:INFO] [000:615] [16147] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:37:INFO] [000:615] [16147] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[07:59:37:INFO] [000:615] [16147] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[07:59:37:INFO] [000:615] [16147] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [000:615] [16147] (audio_device_module_ios.mm:616): output: 0\n[07:59:37:INFO] [000:615] [16147] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[07:59:37:INFO] [000:615] [16147] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [000:615] [16147] (audio_device_module_ios.mm:632): output: 0\n[07:59:37:INFO] [000:615] [16147] (apm_helpers.cc:62): NS set to 0\n[07:59:37:INFO] [000:615] [16147] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[07:59:37:INFO] [000:616] [16147] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[07:59:37:INFO] [000:616] [16147] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[07:59:37:INFO] [000:616] [16147] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[07:59:37:INFO] [000:616] [16147] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[07:59:37:INFO] [000:616] [16147] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[07:59:37:INFO] [000:616] [16147] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[07:59:37:INFO] [000:616] [16147] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[07:59:37:INFO] [000:616] [16147] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[07:59:37:INFO] [000:616] [16147] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:37:INFO] [000:616] [16147] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:37:INFO] [000:616] [16147] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:37:INFO] [000:616] [16147] (webrtc_voice_engine.cc:1520): Set voice channel options. Current options: AudioOptions {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, }\n[07:59:37:INFO] [000:616] [16147] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 2869273597, options: VideoOptions {noise reduction: false, is_screencast : false, }, source = (source))\n[07:59:37:INFO] [000:617] [23311] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled\n[07:59:37:INFO] [000:617] [23311] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled\n[07:59:37:INFO] [000:617] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[07:59:37:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 1\n[07:59:37:INFO] [000:618] [23311] (basic_port_allocator.cc:108): Filtered out ignored networks:\n[07:59:37:INFO] [000:618] [23311] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3]\n[07:59:37:INFO] [000:618] [23311] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2]\n[07:59:37:INFO] [000:618] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[07:59:37:INFO] [000:618] [23311] (basic_port_allocator.cc:851): Network manager has started\n[07:59:37:INFO] [000:618] [23311] (basic_port_allocator.cc:108): Filtered out ignored networks:\n[07:59:37:INFO] [000:618] [23311] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3]\n[07:59:37:INFO] [000:618] [23311] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2]\n[07:59:37:INFO] [000:618] [23311] (basic_port_allocator.cc:851): Network manager has started\n[07:59:37:INFO] [000:619] [23311] (basic_port_allocator.cc:108): Filtered out ignored networks:\n[07:59:37:INFO] [000:619] [23311] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3]\n[07:59:37:INFO] [000:619] [23311] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2]\n[07:59:37:INFO] [000:619] [23311] (basic_port_allocator.cc:766): Allocate ports on 1 networks\n[07:59:37:INFO] [000:620] [23311] (basic_port_allocator.cc:108): Filtered out ignored networks:\n[07:59:37:INFO] [000:620] [23311] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3]\n[07:59:37:INFO] [000:620] [23311] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2]\n[07:59:37:INFO] [000:620] [23311] (basic_port_allocator.cc:766): Allocate ports on 1 networks\n[07:59:37:INFO] [000:620] [23311] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Udp\n[07:59:37:INFO] [000:620] [22539] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds.\n[07:59:37:INFO] [000:620] [22539] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds\n[07:59:37:INFO] [000:621] [22539] (p2p_transport_channel.cc:511): Set ping most likely connection to 0\n[07:59:37:INFO] [000:621] [22539] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500\n[07:59:37:INFO] [000:621] [23311] (port.cc:320): Port[dc04c000::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10\n[07:59:37:INFO] [000:621] [22539] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0\n[07:59:37:INFO] [000:621] [22539] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000\n[07:59:37:INFO] [000:621] [22539] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000\n[07:59:37:INFO] [000:621] [22539] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0\n[07:59:37:INFO] [000:621] [22539] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport db81e280\n[07:59:37:INFO] [000:621] [22539] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds.\n[07:59:37:INFO] [000:621] [22539] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds\n[07:59:37:INFO] [000:621] [22539] (p2p_transport_channel.cc:511): Set ping most likely connection to 0\n[07:59:37:INFO] [000:621] [22539] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500\n[07:59:37:INFO] [000:621] [22539] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0\n[07:59:37:INFO] [000:621] [22539] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000\n[07:59:37:INFO] [000:621] [22539] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000\n[07:59:37:INFO] [000:621] [22539] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 1 transport 0\n[07:59:37:INFO] [000:621] [22539] (dtls_srtp_transport.cc:66): Setting RTP Transport on 1 transport db81e880\n[07:59:37:INFO] [000:621] [22539] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=2M90, renomination enabled\n[07:59:37:INFO] [000:621] [22539] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=2M90, renomination enabled\n[07:59:37:INFO] [000:622] [40455] (webrtc_voice_engine.cc:1502): Setting voice channel options: AudioOptions {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, }\n[07:59:37:INFO] [000:622] [40455] (webrtc_voice_engine.cc:309): WebRtcVoiceEngine::ApplyOptions: AudioOptions {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, }\n[07:59:37:INFO] [000:622] [23311] (basic_port_allocator.cc:875): Adding allocated port for 0\n[07:59:37:INFO] [000:622] [40455] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[07:59:37:INFO] [000:622] [40455] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[07:59:37:INFO] [000:622] [40455] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[07:59:37:INFO] [000:622] [40455] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[07:59:37:INFO] [000:622] [40455] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [000:622] [23311] (basic_port_allocator.cc:894): Port[dc04c000:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator\n[07:59:37:INFO] [000:622] [40455] (audio_device_module_ios.mm:600): output: 0\n[07:59:37:INFO] [000:622] [40455] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:37:INFO] [000:623] [23311] (basic_port_allocator.cc:912): Port[dc04c000:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:199186083:1:udp:2122260223:192.168.144.88:52603:local::0:2M90:RBBwMJj6gFR0wFgy19+3OH8o:1:10:0]\n[07:59:37:INFO] [000:623] [40455] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:37:INFO] [000:623] [23311] (basic_port_allocator.cc:940): Port[dc04c000:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready.\n[07:59:37:INFO] [000:623] [40455] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:37:INFO] [000:623] [23311] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported.\n[07:59:37:INFO] [000:623] [40455] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[07:59:37:INFO] [000:623] [23311] (p2p_transport_channel.cc:799): Port[dc04c000:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0\n[07:59:37:INFO] [000:623] [40455] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[07:59:37:INFO] [000:623] [40455] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [000:623] [40455] (audio_device_module_ios.mm:616): output: 0\n[07:59:37:INFO] [000:623] [40455] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[07:59:37:INFO] [000:623] [40455] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [000:623] [40455] (audio_device_module_ios.mm:632): output: 0\n[07:59:37:INFO] [000:623] [40455] (apm_helpers.cc:62): NS set to 0\n[07:59:37:INFO] [000:623] [40455] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[07:59:37:INFO] [000:623] [40455] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[07:59:37:INFO] [000:623] [40455] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[07:59:37:INFO] [000:623] [23311] (basic_port_allocator.cc:1017): Port[dc04c000:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates.\n[07:59:37:INFO] [000:623] [40455] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[07:59:37:INFO] [000:623] [40455] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[07:59:37:INFO] [000:623] [40455] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[07:59:37:INFO] [000:624] [40455] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[07:59:37:INFO] [000:624] [23311] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Udp\n[07:59:37:INFO] [000:624] [40455] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[07:59:37:INFO] [000:624] [40455] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[07:59:37:INFO] [000:624] [23311] (port.cc:320): Port[dc845a00::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10\n[07:59:37:INFO] [000:624] [40455] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:37:INFO] [000:624] [23311] (basic_port_allocator.cc:875): Adding allocated port for 1\n[07:59:37:INFO] [000:624] [23311] (basic_port_allocator.cc:894): Port[dc845a00:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator\n[07:59:37:INFO] [000:624] [40455] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:37:INFO] [000:624] [40455] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:37:INFO] [000:624] [40455] (webrtc_voice_engine.cc:1520): Set voice channel options. Current options: AudioOptions {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, }\n[07:59:37:INFO] [000:624] [23311] (basic_port_allocator.cc:912): Port[dc845a00:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:199186083:1:udp:2122260223:192.168.144.88:61528:local::0:2M90:RBBwMJj6gFR0wFgy19+3OH8o:1:10:0]\n[07:59:37:INFO] [000:624] [40455] (channel.cc:144): Created channel for 0\n[07:59:37:INFO] [000:624] [23311] (basic_port_allocator.cc:940): Port[dc845a00:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready.\n[07:59:37:INFO] [000:624] [23311] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported.\n[07:59:37:INFO] [000:624] [23311] (p2p_transport_channel.cc:799): Port[dc845a00:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0\n[07:59:37:INFO] [000:624] [40455] (channel.cc:212): BaseChannel::Init_w, media_transport=0\n[07:59:37:INFO] [000:624] [23311] (basic_port_allocator.cc:1017): Port[dc845a00:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates.\n[07:59:37:INFO] [000:624] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [000:624] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [000:625] [40455] (webrtc_video_engine.cc:483): CreateMediaChannel. Options: VideoOptions {}\n[07:59:37:INFO] [000:625] [40455] (channel.cc:144): Created channel for 1\n[07:59:37:INFO] [000:626] [40455] (channel.cc:212): BaseChannel::Init_w, media_transport=0\n[07:59:37:INFO] [000:626] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [000:626] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [000:626] [41991] (peer_connection.cc:4173): Session: 2723856753833851992 Old state: kStable New state: kHaveRemoteOffer\n[07:59:37:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 3\n[07:59:37:INFO] [000:626] [40455] (channel.cc:923): Setting remote voice description\n[07:59:37:INFO] [000:626] [40455] (webrtc_voice_engine.cc:1296): 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[102:ILBC: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}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: 0, options: AudioOptions {}}\n[07:59:37:INFO] [000:627] [40455] (webrtc_voice_engine.cc:1719): Recreate all the receive streams because the send codec has changed.\n[07:59:37:INFO] [000:627] [40455] (webrtc_voice_engine.cc:2180): WebRtcVoiceMediaChannel::SetMaxSendBitrate.\n[07:59:37:INFO] [000:627] [40455] (webrtc_voice_engine.cc:1502): Setting voice channel options: AudioOptions {}\n[07:59:37:INFO] [000:627] [40455] (webrtc_voice_engine.cc:309): WebRtcVoiceEngine::ApplyOptions: AudioOptions {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, }\n[07:59:37:INFO] [000:627] [40455] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[07:59:37:INFO] [000:627] [40455] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[07:59:37:INFO] [000:627] [40455] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[07:59:37:INFO] [000:627] [40455] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[07:59:37:INFO] [000:627] [40455] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [000:627] [40455] (audio_device_module_ios.mm:600): output: 0\n[07:59:37:INFO] [000:627] [40455] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:37:INFO] [000:628] [40455] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:37:INFO] [000:628] [40455] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:37:INFO] [000:628] [40455] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[07:59:37:INFO] [000:628] [40455] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[07:59:37:INFO] [000:628] [40455] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [000:628] [40455] (audio_device_module_ios.mm:616): output: 0\n[07:59:37:INFO] [000:628] [40455] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[07:59:37:INFO] [000:628] [40455] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [000:628] [40455] (audio_device_module_ios.mm:632): output: 0\n[07:59:37:INFO] [000:628] [40455] (apm_helpers.cc:62): NS set to 0\n[07:59:37:INFO] [000:628] [40455] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[07:59:37:INFO] [000:628] [40455] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[07:59:37:INFO] [000:628] [40455] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[07:59:37:INFO] [000:628] [40455] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[07:59:37:INFO] [000:628] [40455] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[07:59:37:INFO] [000:628] [40455] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[07:59:37:INFO] [000:628] [40455] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[07:59:37:INFO] [000:628] [40455] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[07:59:37:INFO] [000:628] [40455] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[07:59:37:INFO] [000:628] [40455] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:37:INFO] [000:628] [40455] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:37:INFO] [000:628] [40455] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:37:INFO] [000:628] [40455] (webrtc_voice_engine.cc:1520): Set voice channel options. Current options: AudioOptions {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, }\n[07:59:37:INFO] [000:628] [40455] (webrtc_voice_engine.cc:1866): AddRecvStream: {id:ARDAMSa0;ssrcs:[2609444602];ssrc_groups:;cname:SVCyZMBheTeYRPl/;stream_ids:ARDAMS;}\n[07:59:37:INFO] [000:629] [40455] (neteq_impl.cc:123): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=200, min_delay_ms=0, enable_fast_accelerate=false, enable_muted_state=true, enable_rtx_handling=false\n[07:59:37:INFO] [000:629] [40455] (audio_coding_module.cc:333): Created\n[07:59:37:INFO] [000:629] [40455] (audio_receive_stream.cc:117): AudioReceiveStream: 2609444602\n[07:59:37:INFO] [000:630] [40455] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2609444602, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS}\n[07:59:37:INFO] [000:630] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [000:630] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [000:630] [40455] (channel.cc:719): Add remote ssrc: 2609444602\n[07:59:37:INFO] [000:630] [40455] (channel.cc:863): Changing voice state, recv=0 send=0\n[07:59:37:INFO] [000:630] [40455] (channel.cc:1061): Setting remote video description\n[07:59:37:INFO] [000:630] [40455] (webrtc_video_engine.cc:681): SetSendParameters: {codecs: [VideoCodec[96:H264], VideoCodec[97:rtx], VideoCodec[98:H264], VideoCodec[99:rtx], VideoCodec[100:VP8], VideoCodec[101:rtx], VideoCodec[127:VP9], VideoCodec[123:rtx], VideoCodec[125:red], VideoCodec[122:rtx], VideoCodec[124:ulpfec]], conference_mode: no, extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {uri: urn:3gpp:video-orientation, id: 12}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: 1}\n[07:59:37:INFO] [000:631] [40455] (webrtc_video_engine.cc:690): Using codec: VideoCodec[96:H264]\n[07:59:37:INFO] [000:631] [40455] (webrtc_video_engine.cc:758): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed.\n[07:59:37:INFO] [000:631] [40455] (webrtc_video_engine.cc:1174): AddRecvStream: {id:ARDAMSv0;ssrcs:[2869273597,673344836];ssrc_groups:{semantics:FID;ssrcs:[2869273597,673344836]};cname:SVCyZMBheTeYRPl/;stream_ids:ARDAMS;}\n[07:59:37:INFO] [000:640] [40455] (video_receive_stream.cc:222): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 102, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 2869273597, 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_payload_type: 106, red_type: 104, rtx_ssrc: 673344836, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 103 (pt) -> 102 (apt), 105 (pt) -> 104 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0}\n[07:59:37:INFO] [000:641] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [000:641] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [000:645] [40455] (channel.cc:719): Add remote ssrc: 2869273597\n[07:59:37:INFO] [000:645] [40455] (channel.cc:996): Changing video state, send=0\n[07:59:37:INFO] [000:645] [41991] (peer_connection.cc:5512): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport.\n[07:59:37:INFO] [000:645] [41991] (peer_connection.cc:2745): Processing the MSIDs for MID=0 (streams=[ARDAMS]).\n[07:59:37:INFO] [000:646] [41991] (peer_connection.cc:2756): Processing the addition of a remote track for MID=0.\n[07:59:37:INFO] [000:646] [40455] (webrtc_voice_engine.cc:1979): SetOutputVolume() to 1 for recv stream with ssrc 2609444602\n[07:59:37:INFO] [000:646] [41991] (peer_connection.cc:2745): Processing the MSIDs for MID=1 (streams=[ARDAMS]).\n[07:59:37:INFO] [000:646] [41991] (peer_connection.cc:2756): Processing the addition of a remote track for MID=1.\n[07:59:37:INFO] [000:646] [40455] (webrtc_video_engine.cc:1304): SetSink: ssrc:2869273597 (ptr)\n[07:59:37:INFO] [000:647] [41991] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60000000a400): created DTMF sender: RTCDtmfSender {\n[07:59:37:INFO] remainingTones:\n[07:59:37:INFO] duration: 0.100000 sec\n[07:59:37:INFO] interToneGap: 0.050000 sec\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:647] [41991] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600000035320): created sender: RTCRtpSender {\n[07:59:37:INFO] senderId: ARDAMSa0\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:647] [41991] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x600000051a90): created receiver: RTCRtpReceiver {\n[07:59:37:INFO] receiverId: aac8edf5-3432-4199-bf0a-f872af25c469\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:647] [41991] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x600000050650): created transceiver: RTCRtpTransceiver {\n[07:59:37:INFO] sender: RTCRtpSender {\n[07:59:37:INFO] senderId: ARDAMSa0\n[07:59:37:INFO] }\n[07:59:37:INFO] receiver: RTCRtpReceiver {\n[07:59:37:INFO] receiverId: aac8edf5-3432-4199-bf0a-f872af25c469\n[07:59:37:INFO] }\n[07:59:37:INFO] }\n[07:59:37:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving audio on track aac8edf5-3432-4199-bf0a-f872af25c469.\n[07:59:37:INFO] [000:647] [41991] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[07:59:37:INFO] [000:647] [41991] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x608000036820): created sender: RTCRtpSender {\n[07:59:37:INFO] senderId: ARDAMSv0\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:647] [41991] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x6080000551b0): created receiver: RTCRtpReceiver {\n[07:59:37:INFO] receiverId: c9a133f5-63d9-44ca-be2f-40c21e6273a6\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:647] [41991] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x608000055030): created transceiver: RTCRtpTransceiver {\n[07:59:37:INFO] sender: RTCRtpSender {\n[07:59:37:INFO] senderId: ARDAMSv0\n[07:59:37:INFO] }\n[07:59:37:INFO] receiver: RTCRtpReceiver {\n[07:59:37:INFO] receiverId: c9a133f5-63d9-44ca-be2f-40c21e6273a6\n[07:59:37:INFO] }\n[07:59:37:INFO] }\n[07:59:37:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving video on track c9a133f5-63d9-44ca-be2f-40c21e6273a6.\n[07:59:37:INFO] (RTCLogging.mm:33): (ARDAppClient.m:381 -[ARDAppClient peerConnection:didAddStream:]): Stream with 1 video tracks and 1 audio tracks was added.\n[07:59:37:INFO] [000:648] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60400000b100): created DTMF sender: RTCDtmfSender {\n[07:59:37:INFO] remainingTones:\n[07:59:37:INFO] duration: 0.100000 sec\n[07:59:37:INFO] interToneGap: 0.050000 sec\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:648] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x6040000386e0): created sender: RTCRtpSender {\n[07:59:37:INFO] senderId: ARDAMSa0\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:648] [23563] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[07:59:37:INFO] [000:649] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x618000034f20): created sender: RTCRtpSender {\n[07:59:37:INFO] senderId: ARDAMSv0\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:651] [41991] (peer_connection.cc:6210): 0 is not ready to use the remote candidate because the local or remote description is not set.\n[07:59:37:INFO] [000:652] [41991] (peer_connection.cc:6210): 1 is not ready to use the remote candidate because the local or remote description is not set.\n[07:59:37:INFO] [000:652] [41991] (peer_connection.cc:2110): CreateAnswer: offer_to_receive_audio is not supported with Unified Plan semantics. Use the RtpTransceiver API instead.\n[07:59:37:INFO] [000:652] [41991] (peer_connection.cc:2115): CreateAnswer: offer_to_receive_video is not supported with Unified Plan semantics. Use the RtpTransceiver API instead.\n[07:59:37:INFO] [000:652] [41991] (peer_connection.cc:5540): Local and Remote descriptions must be applied to get the SSL Role of the session.\n[07:59:37:INFO] [000:652] [41991] (peer_connection.cc:5540): Local and Remote descriptions must be applied to get the SSL Role of the session.\n[07:59:37:INFO] [000:654] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[07:59:37:INFO] [000:655] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[07:59:37:INFO] [000:658] [22539] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0\n[07:59:37:INFO] [000:658] [22539] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport db81e280\n[07:59:37:INFO] [000:659] [22539] (p2p_transport_channel.cc:432): Set ICE ufrag: jBb/ pwd: 5uHv6ftS/m4trn98qL447dNW on transport 0\n[07:59:37:INFO] [000:660] [22539] (dtls_transport.cc:370): DtlsTransport[0|1|__]: DTLS setup complete.\n[07:59:37:INFO] [000:660] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [000:660] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [000:660] [41991] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv.\n[07:59:37:INFO] [000:660] [41991] (rtp_transceiver.cc:169): Changing transceiver (MID=1) current direction from <not set> to kSendRecv.\n[07:59:37:INFO] [000:660] [40455] (channel.cc:553): Channel enabled\n[07:59:37:INFO] [000:660] [40455] (channel.cc:863): Changing voice state, recv=0 send=0\n[07:59:37:INFO] [000:661] [40455] (channel.cc:553): Channel enabled\n[07:59:37:INFO] [000:661] [40455] (channel.cc:996): Changing video state, send=0\n[07:59:37:INFO] [000:661] [41991] (peer_connection.cc:4173): Session: 2723856753833851992 Old state: kHaveRemoteOffer New state: kStable\n[07:59:37:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 0\n[07:59:37:INFO] [000:661] [40455] (channel.cc:871): Setting local voice description\n[07:59:37:INFO] [000:661] [40455] (webrtc_voice_engine.cc:1341): 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[102:ILBC: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}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}]}\n[07:59:37:INFO] [000:661] [40455] (webrtc_voice_engine.cc:1530): Setting receive voice codecs.\n[07:59:37:INFO] [000:662] [40455] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2609444602, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS}\n[07:59:37:INFO] [000:662] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [000:662] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [000:662] [40455] (audio_receive_stream.cc:140): ~AudioReceiveStream: 2609444602\n[07:59:37:INFO] [000:662] [40455] (neteq_impl.cc:123): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=200, min_delay_ms=0, enable_fast_accelerate=false, enable_muted_state=true, enable_rtx_handling=false\n[07:59:37:INFO] [000:662] [40455] (audio_coding_module.cc:333): Created\n[07:59:37:INFO] [000:663] [40455] (audio_receive_stream.cc:117): AudioReceiveStream: 2609444602\n[07:59:37:INFO] [000:663] [40455] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2609444602, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS}\n[07:59:37:INFO] [000:663] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [000:663] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [000:663] [40455] (webrtc_voice_engine.cc:1801): AddSendStream: {id:ARDAMSa0;ssrcs:[4074878902];ssrc_groups:;cname:MTT4IWF1V9de/uJX;stream_ids:ARDAMS;}\n[07:59:37:INFO] [000:663] [40455] (neteq_impl.cc:123): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=200, min_delay_ms=0, enable_fast_accelerate=false, enable_muted_state=false, enable_rtx_handling=false\n[07:59:37:INFO] [000:664] [40455] (audio_coding_module.cc:333): Created\n[07:59:37:INFO] [000:664] [40455] (channel_send.cc:708): Not setting media_transport_ rate observers.\n[07:59:37:INFO] [000:664] [40455] (audio_send_stream.cc:142): AudioSendStream: 4074878902\n[07:59:37:INFO] [000:664] [40455] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 4074878902, extmap-allow-mixed: false, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], c_name: MTT4IWF1V9de/uJX}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: 32000, max_bitrate_bps: 32000, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, cng_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {minptime: 10, useinbandfec: 1}}}}\n[07:59:37:INFO] [000:669] [40455] (audio_encoder_opus.cc:758): Set Opus bitrate to 32000 bps.\n[07:59:37:INFO] [000:669] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [000:669] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [000:670] [40455] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2609444602, local_ssrc: 4074878902, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS}\n[07:59:37:INFO] [000:670] [40455] (channel.cc:676): Add send stream ssrc: 4074878902\n[07:59:37:INFO] [000:670] [40455] (audio_device_module_ios.mm:543): Playing\n[07:59:37:INFO] [000:670] [40455] (audio_device_module_ios.mm:480): InitPlayout\n[07:59:37:INFO] [000:670] [40455] (audio_device_module_ios.mm:506): PlayoutIsInitialized\n[07:59:37:INFO] [000:670] [40455] (audio_device_ios.mm:185): AudioDeviceIOS::InitPlayout\n[07:59:37:INFO] [000:670] [40455] (audio_device_ios.mm:859): AudioDeviceIOS::InitPlayOrRecord\n[07:59:37:INFO] [000:674] [23311] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Relay\n[07:59:37:INFO] [000:674] [23311] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Relay\n[07:59:37:INFO] [000:719] [40455] (RTCLogging.mm:33): (RTCAudioSession.mm:99 -[RTCAudioSession initWithAudioSession:]): RTCAudioSession (0x608000112360): init.\n[07:59:37:INFO] [000:720] [40455] (RTCLogging.mm:33): (audio_device_ios.mm:824 ConfigureAudioSession): Configuring audio session.\n[07:59:37:INFO] [000:720] [40455] (RTCLogging.mm:33): (RTCAudioSession.mm:699 -[RTCAudioSession configureWebRTCSession:]): Configuring audio session for WebRTC.\n[07:59:37:INFO] [000:720] [40455] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:64 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set category to: AVAudioSessionCategoryPlayAndRecord\n[07:59:37:INFO] [000:720] [40455] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:75 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set mode to: AVAudioSessionModeVoiceChat\n[07:59:37:INFO] [000:720] [40455] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:103 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred sample rate to: 48000.00\n[07:59:37:INFO] [000:720] [40455] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:116 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred IO buffer duration to: 0.020000\n[07:59:37:INFO] [000:720] [40455] (RTCLogging.mm:33): (RTCAudioSession.mm:624 -[RTCAudioSession incrementActivationCount]): Incrementing activation count.\n[07:59:37:INFO] [000:720] [40455] (RTCLogging.mm:33): (RTCAudioSession.mm:377 -[RTCAudioSession setActive:error:]): Number of current activations: 1\n[07:59:37:INFO] [000:721] [40455] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:145 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set input number of channels to: 1\n[07:59:37:INFO] [000:721] [40455] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:158 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set output number of channels to: 1\n[07:59:37:INFO] [000:721] [40455] (RTCLogging.mm:33): (audio_device_ios.mm:835 ConfigureAudioSession): Configured audio session.\n[07:59:37:INFO] [000:721] [40455] (audio_device_ios.mm:678): AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession\n[07:59:37:INFO] [000:721] [40455] (RTCLogging.mm:33): (audio_device_ios.mm:683 SetupAudioBuffersForActiveAudioSession): RTCAudioSession: {\n[07:59:37:INFO] category: AVAudioSessionCategoryPlayAndRecord\n[07:59:37:INFO] categoryOptions: 4\n[07:59:37:INFO] mode: AVAudioSessionModeVoiceChat\n[07:59:37:INFO] isActive: 1\n[07:59:37:INFO] sampleRate: 48000.00\n[07:59:37:INFO] IOBufferDuration: 0.010667\n[07:59:37:INFO] outputNumberOfChannels: 2\n[07:59:37:INFO] inputNumberOfChannels: 2\n[07:59:37:INFO] outputLatency: 0.010000\n[07:59:37:INFO] inputLatency: 0.010000\n[07:59:37:INFO] outputVolume: 0.600000\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:721] [40455] (audio_device_ios.mm:714): frames per I/O buffer: 512\n[07:59:37:INFO] [000:721] [40455] (audio_device_ios.mm:715): bytes per I/O buffer: 1024\n[07:59:37:INFO] [000:721] [40455] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer\n[07:59:37:INFO] [000:721] [40455] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000)\n[07:59:37:INFO] [000:721] [40455] (audio_device_buffer.cc:206): SetPlayoutChannels(1)\n[07:59:37:INFO] [000:721] [40455] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000)\n[07:59:37:INFO] [000:721] [40455] (audio_device_buffer.cc:200): SetRecordingChannels(1)\n[07:59:37:INFO] [000:721] [40455] (fine_audio_buffer.cc:32): FineAudioBuffer\n[07:59:37:INFO] [000:721] [40455] (fine_audio_buffer.cc:34): playout_samples_per_channel_10ms: 480\n[07:59:37:INFO] [000:721] [40455] (fine_audio_buffer.cc:36): playout_channels: 1\n[07:59:37:INFO] [000:721] [40455] (fine_audio_buffer.cc:39): record_samples_per_channel_10ms: 480\n[07:59:37:INFO] [000:722] [40455] (fine_audio_buffer.cc:41): record_channels: 1\n[07:59:37:INFO] [000:722] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:196 Initialize): Initializing audio unit with sample rate: 48000.000000\n[07:59:37:INFO] [000:722] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:43 LogStreamDescription): AudioStreamBasicDescription: {\n[07:59:37:INFO] mSampleRate: 48000.00\n[07:59:37:INFO] formatIDString: lpcm\n[07:59:37:INFO] mFormatFlags: 0xC\n[07:59:37:INFO] mBytesPerPacket: 2\n[07:59:37:INFO] mFramesPerPacket: 1\n[07:59:37:INFO] mBytesPerFrame: 2\n[07:59:37:INFO] mChannelsPerFrame: 1\n[07:59:37:INFO] mBitsPerChannel: 16\n[07:59:37:INFO] mReserved: 28672\n[07:59:37:INFO] }\n[07:59:37:INFO] [000:724] [23311] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Tcp\n[07:59:37:INFO] [000:724] [23311] (port.cc:320): Port[dd81a200::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10\n[07:59:37:INFO] [000:724] [23311] (basic_port_allocator.cc:875): Adding allocated port for 0\n[07:59:37:INFO] [000:724] [23311] (basic_port_allocator.cc:894): Port[dd81a200:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator\n[07:59:37:INFO] [000:724] [23311] (basic_port_allocator.cc:912): Port[dd81a200:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:1163700819:1:tcp:1518280447:192.168.144.88:51165:local::0:2M90:RBBwMJj6gFR0wFgy19+3OH8o:1:10:0]\n[07:59:37:INFO] [000:724] [23311] (basic_port_allocator.cc:940): Port[dd81a200:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready.\n[07:59:37:INFO] [000:724] [23311] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported.\n[07:59:37:INFO] [000:724] [23311] (p2p_transport_channel.cc:799): Port[dd81a200:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0\n[07:59:37:INFO] [000:725] [23311] (basic_port_allocator.cc:1017): Port[dd81a200:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates.\n[07:59:37:INFO] [000:725] [23311] (basic_port_allocator.cc:1119): All candidates gathered for 0:1:0\n[07:59:37:INFO] [000:725] [23311] (p2p_transport_channel.cc:854): P2PTransportChannel: 0, component 1 gathering complete\n[07:59:37:INFO] [000:725] [23311] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Tcp\n[07:59:37:INFO] [000:725] [23311] (port.cc:320): Port[dc042800::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10\n[07:59:37:INFO] [000:725] [23311] (basic_port_allocator.cc:875): Adding allocated port for 1\n[07:59:37:INFO] [000:725] [23311] (basic_port_allocator.cc:894): Port[dc042800:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator\n[07:59:37:INFO] [000:726] [23311] (basic_port_allocator.cc:912): Port[dc042800:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:1163700819:1:tcp:1518280447:192.168.144.88:51166:local::0:2M90:RBBwMJj6gFR0wFgy19+3OH8o:1:10:0]\n[07:59:37:INFO] [000:726] [23311] (basic_port_allocator.cc:940): Port[dc042800:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready.\n[07:59:37:INFO] [000:726] [23311] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported.\n[07:59:37:INFO] [000:726] [23311] (p2p_transport_channel.cc:799): Port[dc042800:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0\n[07:59:37:INFO] [000:726] [23311] (basic_port_allocator.cc:1017): Port[dc042800:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates.\n[07:59:37:INFO] [000:726] [23311] (basic_port_allocator.cc:1119): All candidates gathered for 1:1:0\n[07:59:37:INFO] [000:726] [23311] (p2p_transport_channel.cc:854): P2PTransportChannel: 1, component 1 gathering complete\n[07:59:37:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 2\n[07:59:37:INFO] [000:734] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851.\n[07:59:37:INFO] [000:734] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again...\n[07:59:37:INFO] [000:835] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851.\n[07:59:37:INFO] [000:835] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again...\n[07:59:37:INFO] [000:938] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851.\n[07:59:37:INFO] [000:938] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again...\n[07:59:37:INFO] [001:038] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851.\n[07:59:37:INFO] [001:038] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again...\n[07:59:37:INFO] [001:144] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851.\n[07:59:37:INFO] [001:144] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:242 Initialize): Too many initialization attempts.\n[07:59:37:INFO] [001:144] [40455] (audio_device_module_ios.mm:486): output: 0\n[07:59:37:INFO] [001:144] [40455] (audio_device_module_ios.mm:518): StartPlayout\n[07:59:37:INFO] [001:144] [40455] (audio_device_module_ios.mm:543): Playing\n[07:59:37:INFO] [001:144] [40455] (audio_device_buffer.cc:98): StartPlayout\n[07:59:37:INFO] [001:144] [40455] (audio_device_ios.mm:227): AudioDeviceIOS::StartPlayout\n[07:59:37:INFO] [001:144] [40455] (audio_device_module_ios.mm:525): output: 0\n[07:59:37:INFO] [001:145] [40455] (channel.cc:863): Changing voice state, recv=1 send=0\n[07:59:37:INFO] [001:145] [40455] (message_queue.cc:517): Message took 484ms to dispatch. Posted from: SetLocalContent@../../pc/channel.cc:293\n[07:59:37:INFO] [001:145] [40455] (channel.cc:1009): Setting local video description\n[07:59:37:INFO] [001:145] [40455] (webrtc_video_engine.cc:956): SetRecvParameters: {codecs: [VideoCodec[96:H264], VideoCodec[97:rtx], VideoCodec[98:H264], VideoCodec[99:rtx], VideoCodec[100:VP8], VideoCodec[101:rtx], VideoCodec[127:VP9], VideoCodec[123:rtx], VideoCodec[125:red], VideoCodec[122:rtx], VideoCodec[124:ulpfec]], extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {uri: urn:3gpp:video-orientation, id: 12}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}]}\n[07:59:37:INFO] [001:147] [40455] (webrtc_video_engine.cc:971): Changing recv codecs from {VideoCodec[96:H264], VideoCodec[98:H264], VideoCodec[100:VP8], VideoCodec[102:VP9]} to {VideoCodec[96:H264], VideoCodec[98:H264], VideoCodec[100:VP8], VideoCodec[127:VP9]}\n[07:59:37:INFO] [001:147] [40455] (webrtc_video_engine.cc:2542): MaybeRecreateWebRtcFlexfecStream (recv) because of SetRecvParameters\n[07:59:37:INFO] [001:147] [40455] (webrtc_video_engine.cc:2547): RecreateWebRtcVideoStream (recv) because of SetRecvParameters\n[07:59:37:INFO] [001:147] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [001:147] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [001:147] [40455] (video_receive_stream.cc:291): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 102, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 2869273597, 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_payload_type: 106, red_type: 104, rtx_ssrc: 673344836, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 103 (pt) -> 102 (apt), 105 (pt) -> 104 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0}\n[07:59:37:INFO] [001:149] [40455] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0\n[07:59:37:INFO] [001:149] [40455] (receive_statistics_proxy.cc:487): Frames decoded 0\n[07:59:37:INFO] WebRTC.Video.DroppedFrames.Receiver 0\n[07:59:37:INFO] \n[07:59:37:INFO] [001:150] [40455] (video_receive_stream.cc:222): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 127, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 2869273597, 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_payload_type: 124, red_type: 125, rtx_ssrc: 673344836, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 122 (pt) -> 125 (apt), 123 (pt) -> 127 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 12}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0}\n[07:59:37:INFO] [001:150] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [001:150] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [001:151] [40455] (webrtc_video_engine.cc:1072): AddSendStream: {id:ARDAMSv0;ssrcs:[3763443946,154119722];ssrc_groups:{semantics:FID;ssrcs:[3763443946,154119722]};cname:MTT4IWF1V9de/uJX;stream_ids:ARDAMS;}\n[07:59:37:INFO] [001:151] [40455] (webrtc_video_engine.cc:1883): RecreateWebRtcStream (send) because of SetCodec.\n[07:59:37:INFO] [001:152] [40455] (alr_experiment.cc:78): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR start bandwidth usage percent: 80, ALR end budget level percent: 40, ALR end budget level percent: -60, ALR experiment group ID: 3\n[07:59:37:INFO] [001:154] [9987] (rtp_video_sender.cc:516): Transmitting payload type without picture ID using NACK+ULPFEC is a waste of bandwidth since ULPFEC packets also have to be retransmitted. Disabling ULPFEC.\n[07:59:37:INFO] [001:154] [9987] (video_send_stream_impl.cc:250): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [3763443946], rtcp_mode: RtcpMode::kReducedSize, max_packet_size: 1200, extmap-allow-mixed: false, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 12}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 124, red_payload_type: 125, red_rtx_payload_type: 122}, payload_name: H264, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [154119722], payload_type: 97}, c_name: MTT4IWF1V9de/uJX}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off}\n[07:59:37:INFO] [001:154] [9987] (video_send_stream_impl.cc:278): ERROR: Initial encoder max bitrate = -1 which is <= 0!\n[07:59:37:INFO] [001:154] [9987] (video_send_stream_impl.cc:280): Using default encoder max bitrate = 10 Mbps\n[07:59:37:INFO] [001:154] [9987] (video_stream_encoder.cc:628): ConfigureEncoder requested.\n[07:59:37:INFO] [001:154] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [001:154] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [001:155] [40455] (video_send_stream.cc:160): VideoSendStream::Stop\n[07:59:37:INFO] [001:156] [40455] (webrtc_video_engine.cc:1111): SetLocalSsrc on all the receive streams because we added a send stream.\n[07:59:37:INFO] [001:156] [9987] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[07:59:37:INFO] [001:156] [40455] (webrtc_video_engine.cc:2483): RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc=3763443946\n[07:59:37:INFO] [001:156] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [001:156] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [001:156] [40455] (video_receive_stream.cc:291): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 127, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 2869273597, 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_payload_type: 124, red_type: 125, rtx_ssrc: 673344836, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 122 (pt) -> 125 (apt), 123 (pt) -> 127 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 12}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0}\n[07:59:37:INFO] [001:158] [40455] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0\n[07:59:37:INFO] [001:158] [40455] (receive_statistics_proxy.cc:487): Frames decoded 0\n[07:59:37:INFO] WebRTC.Video.DroppedFrames.Receiver 0\n[07:59:37:INFO] \n[07:59:37:INFO] [001:159] [40455] (video_receive_stream.cc:222): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 127, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 2869273597, local_ssrc: 3763443946, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 124, red_type: 125, rtx_ssrc: 673344836, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 122 (pt) -> 125 (apt), 123 (pt) -> 127 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 12}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0}\n[07:59:37:INFO] [001:159] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [001:159] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [001:160] [40455] (channel.cc:676): Add send stream ssrc: 3763443946\n[07:59:37:INFO] [001:160] [40455] (video_send_stream.cc:160): VideoSendStream::Stop\n[07:59:37:INFO] [001:160] [40455] (channel.cc:996): Changing video state, send=0\n[07:59:37:INFO] [001:160] [9987] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[07:59:37:INFO] [001:160] [41991] (peer_connection.cc:4109): Changing IceConnectionState 0 => 1\n[07:59:37:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 1\n[07:59:37:INFO] [001:160] [22539] (jsep_transport_controller.cc:318): Not adding candidate because the JsepTransport doesn't exist. Ignore it.\n[07:59:37:INFO] [001:160] [41991] (peer_connection.cc:5518): Non-rejected SCTP m= section is needed to get the SSL Role of the SCTP transport.\n[07:59:37:INFO] [001:160] [40455] (webrtc_voice_engine.cc:1502): Setting voice channel options: AudioOptions {}\n[07:59:37:INFO] [001:160] [40455] (webrtc_voice_engine.cc:309): WebRtcVoiceEngine::ApplyOptions: AudioOptions {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, }\n[07:59:37:INFO] [001:160] [40455] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[07:59:37:INFO] [001:160] [40455] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[07:59:37:INFO] [001:161] [40455] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[07:59:37:INFO] [001:161] [40455] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[07:59:37:INFO] [001:161] [40455] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [001:162] [40455] (audio_device_module_ios.mm:600): output: 0\n[07:59:37:INFO] [001:162] [40455] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:37:INFO] [001:163] [40455] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:37:INFO] [001:163] [40455] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:37:INFO] [001:163] [40455] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[07:59:37:INFO] [001:163] [40455] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[07:59:37:INFO] [001:163] [40455] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [001:163] [40455] (audio_device_module_ios.mm:616): output: 0\n[07:59:37:INFO] [001:163] [40455] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[07:59:37:INFO] [001:163] [40455] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [001:163] [40455] (audio_device_module_ios.mm:632): output: 0\n[07:59:37:INFO] [001:163] [40455] (apm_helpers.cc:62): NS set to 0\n[07:59:37:INFO] [001:163] [40455] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[07:59:37:INFO] [001:163] [40455] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[07:59:37:INFO] [001:163] [40455] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[07:59:37:INFO] [001:163] [40455] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[07:59:37:INFO] [001:163] [40455] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[07:59:37:INFO] [001:163] [40455] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[07:59:37:INFO] [001:163] [40455] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[07:59:37:INFO] [001:163] [40455] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[07:59:37:INFO] [001:163] [40455] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[07:59:37:INFO] [001:163] [40455] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:37:INFO] [001:164] [40455] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:37:INFO] [001:164] [40455] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:37:INFO] [001:164] [40455] (webrtc_voice_engine.cc:1520): Set voice channel options. Current options: AudioOptions {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, }\n[07:59:37:INFO] [001:164] [40455] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 3763443946, options: VideoOptions {noise reduction: false, is_screencast : false, }, source = (source))\n[07:59:37:INFO] [001:164] [9987] (video_stream_encoder.cc:628): ConfigureEncoder requested.\n[07:59:37:INFO] [001:164] [22539] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled\n[07:59:37:INFO] [001:165] [22539] (basic_port_allocator.cc:108): Filtered out ignored networks:\n[07:59:37:INFO] [001:165] [22539] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3]\n[07:59:37:INFO] [001:165] [22539] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2]\n[07:59:37:INFO] [001:165] [22539] (basic_port_allocator.cc:851): Network manager has started\n[07:59:37:INFO] [001:165] [22539] (basic_port_allocator.cc:108): Filtered out ignored networks:\n[07:59:37:INFO] [001:165] [22539] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3]\n[07:59:37:INFO] [001:165] [22539] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2]\n[07:59:37:INFO] [001:165] [22539] (basic_port_allocator.cc:766): Allocate ports on 1 networks\n[07:59:37:INFO] [001:165] [22539] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Udp\n[07:59:37:INFO] [001:165] [22539] (port.cc:320): Port[da817000::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10\n[07:59:37:INFO] [001:165] [22539] (basic_port_allocator.cc:875): Adding allocated port for 0\n[07:59:37:INFO] [001:165] [22539] (basic_port_allocator.cc:894): Port[da817000:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator\n[07:59:37:INFO] [001:165] [41991] (message_queue.cc:517): Message took 507ms to dispatch. Posted from: SetLocalDescription@../../api/peer_connection_proxy.h:100\n[07:59:37:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 1\n[07:59:37:INFO] [001:165] [22539] (basic_port_allocator.cc:912): Port[da817000:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:199186083:1:udp:2122260223:192.168.144.88:51546:local::0:jBb/:5uHv6ftS/m4trn98qL447dNW:1:10:0]\n[07:59:37:INFO] [001:165] [22539] (basic_port_allocator.cc:940): Port[da817000:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready.\n[07:59:37:INFO] [001:165] [22539] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported.\n[07:59:37:INFO] [001:165] [22539] (p2p_transport_channel.cc:799): Port[da817000:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0\n[07:59:37:INFO] [001:165] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[07:59:37:INFO] [001:165] [22539] (port.cc:1128): Conn[da862200:0:Net[en0:192.168.144.0/22:Wifi:id=1]:JgbNGUAH:1:0:local:udp:192.168.144.88:51546->UdYfm5Uq:1:2122260223:local:udp:192.168.144.88:52603|C--W|-|0|0|9115038255631187454|-]: Connection created\n[07:59:37:INFO] [001:166] [22539] (p2p_transport_channel.cc:1264): Channel[0|1|__]: Created connection with origin: 2, total: 1\n[07:59:37:INFO] [001:166] [22539] (p2p_transport_channel.cc:1914): Channel[0|1|__]: Transport channel state changed from 0 to 2\n[07:59:37:INFO] [001:166] [22539] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete.\n[07:59:37:INFO] [001:166] [22539] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete.\n[07:59:37:INFO] [001:166] [22539] (p2p_transport_channel.cc:1476): Channel[0|1|__]: Have a pingable connection for the first time; starting to ping.\n[07:59:37:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 1\n[07:59:37:INFO] [001:166] [22539] (basic_port_allocator.cc:1017): Port[da817000:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates.\n[07:59:37:INFO] [001:166] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[07:59:37:INFO] [001:167] [22539] (port.cc:1776): Conn[da862200:0:Net[en0:192.168.144.0/22:Wifi:id=1]:JgbNGUAH:1:0:local:udp:192.168.144.88:51546->UdYfm5Uq:1:2122260223:local:udp:192.168.144.88:52603|C--W|-|0|0|9115038255631187454|-]: Sent STUN ping, id=6d6a5430614d6165502b6777, use_candidate=0, nomination=0\n[07:59:37:INFO] [001:167] [23311] (port.cc:530): Received STUN ping id=6d6a5430614d6165502b6777 from unknown address 192.168.144.88:51546\n[07:59:37:INFO] [001:167] [23311] (port.cc:1128): Conn[dc049c00:0:Net[en0:192.168.144.0/22:Wifi:id=1]:Uzl2s3kZ:1:0:local:udp:192.168.144.88:52603->R+HoTWme:1:1853824767:prflx:udp:192.168.144.88:51546|C--W|-|0|0|7962116751024340479|-]: Connection created\n[07:59:37:INFO] [001:167] [23311] (p2p_transport_channel.cc:988): Adding connection from peer reflexive candidate: Cand[:2279935700:1:udp:1853824767:192.168.144.88:51546:prflx::0:jBb/::1:10:0]\n[07:59:37:INFO] [001:168] [23311] (port.cc:855): Port[dc04c000:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Sent STUN ping response, to=192.168.144.88:51546, id=6d6a5430614d6165502b6777\n[07:59:37:INFO] [001:168] [23311] (p2p_transport_channel.cc:1914): Channel[0|1|R_]: Transport channel state changed from 0 to 2\n[07:59:37:INFO] [001:168] [23311] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete.\n[07:59:37:INFO] [001:168] [23311] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete.\n[07:59:37:INFO] [001:168] [22539] (port.cc:1724): Conn[da862200:0:Net[en0:192.168.144.0/22:Wifi:id=1]:JgbNGUAH:1:0:local:udp:192.168.144.88:51546->UdYfm5Uq:1:2122260223:local:udp:192.168.144.88:52603|C--I|-|0|0|9115038255631187454|-]: Received STUN ping response, id=6d6a5430614d6165502b6777, code=0, rtt=1, pings_since_last_response=6d6a5430614d6165502b6777\n[07:59:37:INFO] [001:169] [22539] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed\n[07:59:37:INFO] [001:169] [22539] (p2p_transport_channel.cc:1852): Channel[0|1|__]: New selected connection: Conn[da862200:0:Net[en0:192.168.144.0/22:Wifi:id=1]:JgbNGUAH:1:0:local:udp:192.168.144.88:51546->UdYfm5Uq:1:2122260223:local:udp:192.168.144.88:52603|CRWS|S|0|0|9115038255631187454|1]\n[07:59:37:INFO] [001:169] [22539] (channel.cc:376): Network route was changed.\n[07:59:37:INFO] [001:169] [22539] (channel.cc:376): Network route was changed.\n[07:59:37:INFO] [001:169] [22539] (dtls_transport.cc:811): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 1\n[07:59:37:INFO] [001:169] [22539] (openssl_stream_adapter.cc:777): BeginSSL with peer.\n[07:59:37:INFO] [001:171] [16147] (webrtc_video_engine.cc:1349): Call stats: 274107, {send_bw_bps: 0, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1}\n[07:59:37:INFO] [001:173] [23311] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0\n[07:59:37:INFO] [001:173] [23311] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport db905290\n[07:59:37:INFO] [001:173] [23311] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=jBb/, renomination enabled\n[07:59:37:INFO] [001:173] [23311] (dtls_transport.cc:370): DtlsTransport[0|1|__]: DTLS setup complete.\n[07:59:37:INFO] [001:173] [16147] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [001:173] [16147] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [001:173] [23311] (p2p_transport_channel.cc:1476): Channel[0|1|R_]: Have a pingable connection for the first time; starting to ping.\n[07:59:37:INFO] [001:173] [23311] (p2p_transport_channel.cc:2440): Selecting connection for triggered check: Conn[dc049c00:0:Net[en0:192.168.144.0/22:Wifi:id=1]:Uzl2s3kZ:1:0:local:udp:192.168.144.88:52603->R+HoTWme:1:1853824767:prflx:udp:192.168.144.88:51546|CR-W|-|0|0|7962116751024340479|-]\n[07:59:37:INFO] [001:173] [16147] (channel.cc:553): Channel enabled\n[07:59:37:INFO] [001:173] [23311] (port.cc:1776): Conn[dc049c00:0:Net[en0:192.168.144.0/22:Wifi:id=1]:Uzl2s3kZ:1:0:local:udp:192.168.144.88:52603->R+HoTWme:1:1853824767:prflx:udp:192.168.144.88:51546|CR-W|-|0|0|7962116751024340479|-]: Sent STUN ping, id=5249346a545734354a594274, use_candidate=0, nomination=0\n[07:59:37:INFO] [001:174] [16147] (channel.cc:863): Changing voice state, recv=1 send=0\n[07:59:37:INFO] [001:174] [16147] (channel.cc:553): Channel enabled\n[07:59:37:INFO] [001:174] [16147] (channel.cc:996): Changing video state, send=0\n[07:59:37:INFO] [001:174] [23563] (peer_connection.cc:4173): Session: 892703080514118717 Old state: kHaveLocalOffer New state: kStable\n[07:59:37:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 0\n[07:59:37:INFO] [001:174] [16147] (channel.cc:923): Setting remote voice description\n[07:59:37:INFO] [001:174] [16147] (webrtc_voice_engine.cc:1296): 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[102:ILBC: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}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: 0, options: AudioOptions {}}\n[07:59:37:INFO] [001:175] [16147] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2609444602, extmap-allow-mixed: false, extensions: [], c_name: SVCyZMBheTeYRPl/}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: 32000, max_bitrate_bps: 32000, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, cng_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {minptime: 10, useinbandfec: 1}}}}\n[07:59:37:INFO] [001:175] [16147] (audio_encoder_opus.cc:758): Set Opus bitrate to 32000 bps.\n[07:59:37:INFO] [001:175] [16147] (webrtc_voice_engine.cc:1719): Recreate all the receive streams because the send codec has changed.\n[07:59:37:INFO] [001:175] [16147] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2609444602, extmap-allow-mixed: false, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], c_name: SVCyZMBheTeYRPl/}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: 32000, max_bitrate_bps: 32000, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, cng_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {minptime: 10, useinbandfec: 1}}}}\n[07:59:37:INFO] [001:175] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client enter_early_data\n[07:59:37:INFO] [001:175] [16147] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2609444602, extmap-allow-mixed: false, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], c_name: SVCyZMBheTeYRPl/}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: 32000, max_bitrate_bps: 32000, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, cng_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {minptime: 10, useinbandfec: 1}}}}\n[07:59:37:INFO] [001:175] [16147] (webrtc_voice_engine.cc:2180): WebRtcVoiceMediaChannel::SetMaxSendBitrate.\n[07:59:37:INFO] [001:175] [16147] (webrtc_voice_engine.cc:1502): Setting voice channel options: AudioOptions {}\n[07:59:37:INFO] [001:175] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client read_hello_verify_request\n[07:59:37:INFO] [001:176] [23311] (dtls_transport.cc:546): DtlsTransport[0|1|__]: Packet received before DTLS started.\n[07:59:37:INFO] [001:176] [16147] (webrtc_voice_engine.cc:309): WebRtcVoiceEngine::ApplyOptions: AudioOptions {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, }\n[07:59:37:INFO] [001:176] [22539] (openssl_adapter.cc:794): SSL_connect:error in TLS client read_hello_verify_request\n[07:59:37:INFO] [001:176] [23311] (dtls_transport.cc:555): DtlsTransport[0|1|__]: Caching DTLS ClientHello packet until DTLS is started.\n[07:59:37:INFO] [001:176] [16147] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[07:59:37:INFO] [001:176] [22539] (dtls_transport.cc:698): DtlsTransport[0|1|__]: DtlsTransport: Started DTLS handshake\n[07:59:37:INFO] [001:176] [16147] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[07:59:37:INFO] [001:176] [16147] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[07:59:37:INFO] [001:176] [16147] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[07:59:37:INFO] [001:176] [16147] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [001:176] [22539] (srtp_transport.cc:364): The params in SRTP transport are reset.\n[07:59:37:INFO] [001:176] [16147] (audio_device_module_ios.mm:600): output: 0\n[07:59:37:INFO] [001:176] [16147] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:37:INFO] [001:176] [22539] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete.\n[07:59:37:INFO] [001:177] [22539] (basic_port_allocator.cc:1119): All candidates gathered for 0:1:0\n[07:59:37:INFO] [001:177] [22539] (p2p_transport_channel.cc:854): P2PTransportChannel: 0, component 1 gathering complete\n[07:59:37:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 2\n[07:59:37:INFO] [001:177] [16147] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:37:INFO] [001:177] [16147] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:37:INFO] [001:177] [16147] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[07:59:37:INFO] [001:177] [16147] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[07:59:37:INFO] [001:177] [16147] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [001:177] [23311] (port.cc:1724): Conn[dc049c00:0:Net[en0:192.168.144.0/22:Wifi:id=1]:Uzl2s3kZ:1:0:local:udp:192.168.144.88:52603->R+HoTWme:1:1853824767:prflx:udp:192.168.144.88:51546|CR-I|-|0|0|7962116751024340479|-]: Received STUN ping response, id=5249346a545734354a594274, code=0, rtt=4, pings_since_last_response=5249346a545734354a594274\n[07:59:37:INFO] [001:177] [16147] (audio_device_module_ios.mm:616): output: 0\n[07:59:37:INFO] [001:177] [16147] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[07:59:37:INFO] [001:177] [16147] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[07:59:37:INFO] [001:177] [16147] (audio_device_module_ios.mm:632): output: 0\n[07:59:37:INFO] [001:177] [16147] (apm_helpers.cc:62): NS set to 0\n[07:59:37:INFO] [001:177] [16147] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[07:59:37:INFO] [001:177] [16147] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[07:59:37:INFO] [001:177] [16147] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[07:59:37:INFO] [001:177] [16147] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[07:59:37:INFO] [001:177] [23311] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed\n[07:59:37:INFO] [001:177] [16147] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[07:59:37:INFO] [001:177] [16147] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[07:59:37:INFO] [001:177] [16147] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[07:59:37:INFO] [001:177] [16147] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[07:59:37:INFO] [001:177] [23311] (p2p_transport_channel.cc:1852): Channel[0|1|R_]: New selected connection: Conn[dc049c00:0:Net[en0:192.168.144.0/22:Wifi:id=1]:Uzl2s3kZ:1:0:local:udp:192.168.144.88:52603->R+HoTWme:1:1853824767:prflx:udp:192.168.144.88:51546|CRWS|S|0|0|7962116751024340479|4]\n[07:59:37:INFO] [001:178] [16147] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[07:59:37:INFO] [001:178] [23311] (channel.cc:376): Network route was changed.\n[07:59:37:INFO] [001:178] [16147] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:37:INFO] [001:178] [23311] (channel.cc:376): Network route was changed.\n[07:59:37:INFO] [001:178] [23311] (dtls_transport.cc:811): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 4\n[07:59:37:INFO] [001:178] [16147] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:37:INFO] [001:178] [23311] (openssl_stream_adapter.cc:777): BeginSSL with peer.\n[07:59:37:INFO] [001:178] [16147] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:37:INFO] [001:178] [16147] (webrtc_voice_engine.cc:1520): Set voice channel options. Current options: AudioOptions {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, }\n[07:59:37:INFO] [001:178] [16147] (webrtc_voice_engine.cc:1866): AddRecvStream: {id:ARDAMSa0;ssrcs:[4074878902];ssrc_groups:;cname:MTT4IWF1V9de/uJX;stream_ids:ARDAMS;}\n[07:59:37:INFO] [001:178] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_hello\n[07:59:37:INFO] [001:178] [23311] (openssl_adapter.cc:794): SSL_accept:error in TLS server read_client_hello\n[07:59:37:INFO] [001:178] [16147] (neteq_impl.cc:123): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=200, min_delay_ms=0, enable_fast_accelerate=false, enable_muted_state=true, enable_rtx_handling=false\n[07:59:37:INFO] [001:178] [23311] (dtls_transport.cc:698): DtlsTransport[0|1|__]: DtlsTransport: Started DTLS handshake\n[07:59:37:INFO] [001:178] [23311] (srtp_transport.cc:364): The params in SRTP transport are reset.\n[07:59:37:INFO] [001:178] [16147] (audio_coding_module.cc:333): Created\n[07:59:37:INFO] [001:178] [23311] (dtls_transport.cc:704): DtlsTransport[0|1|__]: Handling cached DTLS ClientHello packet.\n[07:59:37:INFO] [001:179] [16147] (audio_receive_stream.cc:117): AudioReceiveStream: 4074878902\n[07:59:37:INFO] [001:179] [16147] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 4074878902, local_ssrc: 2609444602, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS}\n[07:59:37:INFO] [001:179] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server select_certificate\n[07:59:37:INFO] [001:179] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server select_parameters\n[07:59:37:INFO] [001:179] [16147] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:37:INFO] [001:179] [16147] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:37:INFO] [001:179] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_hello\n[07:59:37:INFO] [001:179] [16147] (audio_device_module_ios.mm:543): Playing\n[07:59:37:INFO] [001:179] [16147] (audio_device_module_ios.mm:480): InitPlayout\n[07:59:37:INFO] [001:179] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_certificate\n[07:59:37:INFO] [001:179] [16147] (audio_device_module_ios.mm:506): PlayoutIsInitialized\n[07:59:37:INFO] [001:179] [16147] (audio_device_ios.mm:185): AudioDeviceIOS::InitPlayout\n[07:59:37:INFO] [001:179] [16147] (audio_device_ios.mm:859): AudioDeviceIOS::InitPlayOrRecord\n[07:59:37:INFO] [001:180] [16147] (RTCLogging.mm:33): (audio_device_ios.mm:824 ConfigureAudioSession): Configuring audio session.\n[07:59:37:INFO] [001:180] [16147] (RTCLogging.mm:33): (RTCAudioSession.mm:699 -[RTCAudioSession configureWebRTCSession:]): Configuring audio session for WebRTC.\n[07:59:37:INFO] [001:180] [16147] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:116 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred IO buffer duration to: 0.020000\n[07:59:37:INFO] [001:180] [16147] (RTCLogging.mm:33): (RTCAudioSession.mm:624 -[RTCAudioSession incrementActivationCount]): Incrementing activation count.\n[07:59:37:INFO] [001:180] [16147] (RTCLogging.mm:33): (RTCAudioSession.mm:377 -[RTCAudioSession setActive:error:]): Number of current activations: 2\n[07:59:37:INFO] [001:180] [16147] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:145 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set input number of channels to: 1\n[07:59:37:INFO] [001:180] [16147] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:158 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set output number of channels to: 1\n[07:59:37:INFO] [001:180] [16147] (RTCLogging.mm:33): (audio_device_ios.mm:835 ConfigureAudioSession): Configured audio session.\n[07:59:37:INFO] [001:180] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_key_exchange\n[07:59:37:INFO] [001:180] [16147] (audio_device_ios.mm:678): AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession\n[07:59:37:INFO] [001:180] [16147] (RTCLogging.mm:33): (audio_device_ios.mm:683 SetupAudioBuffersForActiveAudioSession): RTCAudioSession: {\n[07:59:37:INFO] category: AVAudioSessionCategoryPlayAndRecord\n[07:59:37:INFO] categoryOptions: 4\n[07:59:37:INFO] mode: AVAudioSessionModeVoiceChat\n[07:59:37:INFO] isActive: 1\n[07:59:37:INFO] sampleRate: 48000.00\n[07:59:37:INFO] IOBufferDuration: 0.010667\n[07:59:37:INFO] outputNumberOfChannels: 2\n[07:59:37:INFO] inputNumberOfChannels: 2\n[07:59:37:INFO] outputLatency: 0.010000\n[07:59:37:INFO] inputLatency: 0.010000\n[07:59:37:INFO] outputVolume: 0.600000\n[07:59:37:INFO] }\n[07:59:37:INFO] [001:180] [16147] (audio_device_ios.mm:714): frames per I/O buffer: 512\n[07:59:37:INFO] [001:180] [16147] (audio_device_ios.mm:715): bytes per I/O buffer: 1024\n[07:59:37:INFO] [001:180] [16147] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer\n[07:59:37:INFO] [001:180] [16147] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000)\n[07:59:37:INFO] [001:181] [16147] (audio_device_buffer.cc:206): SetPlayoutChannels(1)\n[07:59:37:INFO] [001:181] [16147] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000)\n[07:59:37:INFO] [001:181] [16147] (audio_device_buffer.cc:200): SetRecordingChannels(1)\n[07:59:37:INFO] [001:181] [16147] (fine_audio_buffer.cc:32): FineAudioBuffer\n[07:59:37:INFO] [001:181] [16147] (fine_audio_buffer.cc:34): playout_samples_per_channel_10ms: 480\n[07:59:37:INFO] [001:181] [16147] (fine_audio_buffer.cc:36): playout_channels: 1\n[07:59:37:INFO] [001:181] [16147] (fine_audio_buffer.cc:39): record_samples_per_channel_10ms: 480\n[07:59:37:INFO] [001:181] [16147] (fine_audio_buffer.cc:41): record_channels: 1\n[07:59:37:INFO] [001:181] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:196 Initialize): Initializing audio unit with sample rate: 48000.000000\n[07:59:37:INFO] [001:181] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:43 LogStreamDescription): AudioStreamBasicDescription: {\n[07:59:37:INFO] mSampleRate: 48000.00\n[07:59:37:INFO] formatIDString: lpcm\n[07:59:37:INFO] mFormatFlags: 0xC\n[07:59:37:INFO] mBytesPerPacket: 2\n[07:59:37:INFO] mFramesPerPacket: 1\n[07:59:37:INFO] mBytesPerFrame: 2\n[07:59:37:INFO] mChannelsPerFrame: 1\n[07:59:37:INFO] mBitsPerChannel: 16\n[07:59:37:INFO] mReserved: 28672\n[07:59:37:INFO] }\n[07:59:37:INFO] [001:181] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851.\n[07:59:37:INFO] [001:181] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again...\n[07:59:37:INFO] [001:183] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_hello_done\n[07:59:37:INFO] [001:183] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_certificate\n[07:59:37:INFO] [001:183] [23311] (openssl_adapter.cc:794): SSL_accept:error in TLS server read_client_certificate\n[07:59:37:INFO] [001:183] [23311] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete.\n[07:59:37:INFO] [001:183] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_hello\n[07:59:37:INFO] [001:183] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_certificate\n[07:59:37:INFO] [001:183] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client read_certificate_status\n[07:59:37:INFO] [001:183] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client verify_server_certificate\n[07:59:37:INFO] [001:184] [22539] (openssl_stream_adapter.cc:1049): Accepted peer certificate.\n[07:59:37:INFO] [001:184] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_key_exchange\n[07:59:37:INFO] [001:184] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client read_certificate_request\n[07:59:37:INFO] [001:184] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_hello_done\n[07:59:37:INFO] [001:184] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_certificate\n[07:59:37:INFO] [001:184] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_key_exchange\n[07:59:37:INFO] [001:185] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_certificate_verify\n[07:59:37:INFO] [001:188] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_finished\n[07:59:37:INFO] [001:188] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client finish_flight\n[07:59:37:INFO] [001:189] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client read_session_ticket\n[07:59:37:INFO] [001:189] [22539] (openssl_adapter.cc:794): SSL_connect:error in TLS client read_session_ticket\n[07:59:37:INFO] [001:189] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server verify_client_certificate\n[07:59:37:INFO] [001:189] [23311] (openssl_stream_adapter.cc:1049): Accepted peer certificate.\n[07:59:37:INFO] [001:189] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_key_exchange\n[07:59:37:INFO] [001:191] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_certificate_verify\n[07:59:37:INFO] [001:191] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server read_change_cipher_spec\n[07:59:37:INFO] [001:191] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server process_change_cipher_spec\n[07:59:37:INFO] [001:191] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server read_next_proto\n[07:59:37:INFO] [001:191] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server read_channel_id\n[07:59:37:INFO] [001:191] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_finished\n[07:59:37:INFO] [001:191] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_finished\n[07:59:37:INFO] [001:191] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server finish_server_handshake\n[07:59:37:INFO] [001:192] [23311] (openssl_adapter.cc:784): SSL_accept:TLS server done\n[07:59:37:INFO] [001:192] [23311] (dtls_transport.cc:630): DtlsTransport[0|1|__]: DTLS handshake complete.\n[07:59:37:INFO] [001:192] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client process_change_cipher_spec\n[07:59:37:INFO] [001:192] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_finished\n[07:59:37:INFO] [001:192] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client finish_client_handshake\n[07:59:37:INFO] [001:192] [23311] (jsep_transport_controller.cc:1213): Transport 0 writability changed to 1.\n[07:59:37:INFO] [001:192] [22539] (openssl_adapter.cc:784): SSL_connect:TLS client done\n[07:59:37:INFO] [001:192] [22539] (dtls_transport.cc:630): DtlsTransport[0|1|__]: DTLS handshake complete.\n[07:59:38:INFO] [001:192] [23311] (dtls_srtp_transport.cc:219): Extracting keys from transport: 0\n[07:59:38:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 2\n[07:59:38:INFO] [001:192] [22539] (jsep_transport_controller.cc:1213): Transport 0 writability changed to 1.\n[07:59:38:INFO] [001:192] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[07:59:38:INFO] [001:192] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[07:59:38:INFO] [001:192] [22539] (dtls_srtp_transport.cc:219): Extracting keys from transport: 0\n[07:59:38:INFO] [001:192] [41991] (peer_connection.cc:6044): Changing to ICE connected state because all transports are writable.\n[07:59:38:INFO] [001:192] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[07:59:38:INFO] [001:192] [5635] (rtp_transport_controller_send.cc:459): Creating fallback congestion controller\n[07:59:38:INFO] [001:194] [41991] (peer_connection.cc:4109): Changing IceConnectionState 1 => 2\n[07:59:38:INFO] [001:194] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[07:59:38:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 2\n[07:59:38:INFO] [001:194] [5635] (alr_experiment.cc:78): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR start bandwidth usage percent: 80, ALR end budget level percent: 40, ALR end budget level percent: -60, ALR experiment group ID: 3\n[07:59:38:INFO] [001:194] [5635] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85\n[07:59:38:INFO] [001:194] [5635] (delay_based_bwe.cc:107): Using Trendline filter for delay change estimation with window size 20\n[07:59:38:INFO] [001:196] [5635] (delay_based_bwe.cc:297): BWE Setting start bitrate to: 300 kbps\n[07:59:38:INFO] [001:197] [5635] (probe_controller.cc:266): Measured bitrate: 300000 Minimum to probe further: 1260000\n[07:59:38:INFO] [001:197] [5635] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (900000:1687:5)\n[07:59:38:INFO] [001:197] [5635] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5)\n[07:59:38:INFO] [001:197] [5635] (bitrate_allocator.cc:115): Current BWE 300000\n[07:59:38:INFO] [001:197] [23311] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1\n[07:59:38:INFO] [001:197] [22539] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1\n[07:59:38:INFO] [001:197] [23311] (channel.cc:583): Channel writable (0) for the first time\n[07:59:38:INFO] [001:197] [22539] (channel.cc:583): Channel writable (0) for the first time\n[07:59:38:INFO] [001:197] [23311] (channel.cc:583): Channel writable (1) for the first time\n[07:59:38:INFO] [001:197] [22539] (channel.cc:583): Channel writable (1) for the first time\n[07:59:38:INFO] [001:198] [40455] (webrtc_voice_engine.cc:309): WebRtcVoiceEngine::ApplyOptions: AudioOptions {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, }\n[07:59:38:INFO] [001:198] [40455] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[07:59:38:INFO] [001:198] [40455] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[07:59:38:INFO] [001:198] [40455] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[07:59:38:INFO] [001:198] [40455] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[07:59:38:INFO] [001:198] [40455] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[07:59:38:INFO] [001:198] [40455] (audio_device_module_ios.mm:600): output: 0\n[07:59:38:INFO] [001:198] [40455] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:38:INFO] [001:198] [40455] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:38:INFO] [001:198] [40455] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:38:INFO] [001:198] [40455] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[07:59:38:INFO] [001:199] [40455] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[07:59:38:INFO] [001:199] [40455] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[07:59:38:INFO] [001:199] [40455] (audio_device_module_ios.mm:616): output: 0\n[07:59:38:INFO] [001:199] [40455] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[07:59:38:INFO] [001:199] [40455] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[07:59:38:INFO] [001:199] [40455] (audio_device_module_ios.mm:632): output: 0\n[07:59:38:INFO] [001:199] [40455] (apm_helpers.cc:62): NS set to 0\n[07:59:38:INFO] [001:199] [40455] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[07:59:38:INFO] [001:199] [40455] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[07:59:38:INFO] [001:199] [40455] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[07:59:38:INFO] [001:199] [40455] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[07:59:38:INFO] [001:199] [40455] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[07:59:38:INFO] [001:199] [40455] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[07:59:38:INFO] [001:199] [40455] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[07:59:38:INFO] [001:199] [40455] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[07:59:38:INFO] [001:199] [40455] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[07:59:38:INFO] [001:201] [40455] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:38:INFO] [001:201] [40455] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:38:INFO] [001:201] [40455] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:38:INFO] [001:201] [40455] (audio_device_module_ios.mm:512): RecordingIsInitialized\n[07:59:38:INFO] [001:201] [5635] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 32000bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 32000bps\n[07:59:38:INFO] [001:201] [40455] (audio_device_module_ios.mm:574): Recording\n[07:59:38:INFO] [001:201] [40455] (audio_device_module_ios.mm:493): InitRecording\n[07:59:38:INFO] [001:201] [40455] (audio_device_module_ios.mm:512): RecordingIsInitialized\n[07:59:38:INFO] [001:201] [40455] (audio_device_module_ios.mm:549): StartRecording\n[07:59:38:INFO] [001:201] [40455] (audio_device_module_ios.mm:574): Recording\n[07:59:38:INFO] [001:201] [40455] (audio_device_buffer.cc:117): StartRecording\n[07:59:38:INFO] [001:202] [40455] (audio_device_ios.mm:280): AudioDeviceIOS::StartRecording\n[07:59:38:INFO] [001:202] [40455] (audio_device_module_ios.mm:556): output: 0\n[07:59:38:INFO] [001:202] [40455] (channel.cc:863): Changing voice state, recv=1 send=1\n[07:59:38:INFO] [001:202] [40455] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers\n[07:59:38:INFO] [001:202] [5635] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers\n[07:59:38:INFO] [001:202] [5635] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 32000bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10032000bps\n[07:59:38:INFO] [001:202] [40455] (channel.cc:996): Changing video state, send=1\n[07:59:38:INFO] [001:202] [9987] (video_stream_encoder.cc:1568): Video suspend state changed to: not suspended\n[07:59:38:INFO] [001:283] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851.\n[07:59:38:INFO] [001:283] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again...\n[07:59:38:INFO] [001:384] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851.\n[07:59:38:INFO] [001:384] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again...\n[07:59:38:INFO] [001:485] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851.\n[07:59:38:INFO] [001:485] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again...\n[07:59:38:INFO] [001:586] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851.\n[07:59:38:INFO] [001:586] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:242 Initialize): Too many initialization attempts.\n[07:59:38:INFO] [001:586] [16147] (audio_device_module_ios.mm:486): output: 0\n[07:59:38:INFO] [001:586] [16147] (audio_device_module_ios.mm:518): StartPlayout\n[07:59:38:INFO] [001:586] [16147] (audio_device_module_ios.mm:543): Playing\n[07:59:38:INFO] [001:586] [16147] (audio_device_buffer.cc:98): StartPlayout\n[07:59:38:INFO] [001:586] [16147] (audio_device_ios.mm:227): AudioDeviceIOS::StartPlayout\n[07:59:38:INFO] [001:586] [16147] (audio_device_module_ios.mm:525): output: 0\n[07:59:38:INFO] [001:586] [16147] (channel.cc:719): Add remote ssrc: 4074878902\n[07:59:38:INFO] [001:586] [16147] (webrtc_voice_engine.cc:309): WebRtcVoiceEngine::ApplyOptions: AudioOptions {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, }\n[07:59:38:INFO] [001:586] [16147] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[07:59:38:INFO] [001:586] [16147] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[07:59:38:INFO] [001:586] [16147] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[07:59:38:INFO] [001:586] [16147] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[07:59:38:INFO] [001:587] [16147] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[07:59:38:INFO] [001:587] [16147] (audio_device_module_ios.mm:600): output: 0\n[07:59:38:INFO] [001:587] [16147] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:38:INFO] [001:587] [16147] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:38:INFO] [001:587] [16147] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:38:INFO] [001:587] [16147] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[07:59:38:INFO] [001:587] [16147] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[07:59:38:INFO] [001:587] [16147] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[07:59:38:INFO] [001:587] [16147] (audio_device_module_ios.mm:616): output: 0\n[07:59:38:INFO] [001:587] [16147] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[07:59:38:INFO] [001:587] [16147] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[07:59:38:INFO] [001:587] [16147] (audio_device_module_ios.mm:632): output: 0\n[07:59:38:INFO] [001:587] [16147] (apm_helpers.cc:62): NS set to 0\n[07:59:38:INFO] [001:587] [16147] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[07:59:38:INFO] [001:587] [16147] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[07:59:38:INFO] [001:587] [16147] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[07:59:38:INFO] [001:587] [16147] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[07:59:38:INFO] [001:587] [16147] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[07:59:38:INFO] [001:587] [16147] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[07:59:38:INFO] [001:587] [16147] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[07:59:38:INFO] [001:587] [16147] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[07:59:38:INFO] [001:587] [16147] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[07:59:38:INFO] [001:587] [16147] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[07:59:38:INFO] [001:587] [16147] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[07:59:38:INFO] [001:587] [16147] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[07:59:38:INFO] [001:588] [16147] (audio_device_module_ios.mm:512): RecordingIsInitialized\n[07:59:38:INFO] [001:588] [5635] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 32000bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 32000bps\n[07:59:38:INFO] [001:588] [16147] (audio_device_module_ios.mm:574): Recording\n[07:59:38:INFO] [001:588] [16147] (audio_device_module_ios.mm:493): InitRecording\n[07:59:38:INFO] [001:588] [16147] (audio_device_module_ios.mm:512): RecordingIsInitialized\n[07:59:38:INFO] [001:588] [16147] (audio_device_module_ios.mm:549): StartRecording\n[07:59:38:INFO] [001:588] [16147] (audio_device_module_ios.mm:574): Recording\n[07:59:38:INFO] [001:588] [16147] (audio_device_buffer.cc:117): StartRecording\n[07:59:38:INFO] [001:588] [16147] (audio_device_ios.mm:280): AudioDeviceIOS::StartRecording\n[07:59:38:INFO] [001:588] [16147] (audio_device_module_ios.mm:556): output: 0\n[07:59:38:INFO] [001:588] [16147] (channel.cc:863): Changing voice state, recv=1 send=1\n[07:59:38:INFO] [001:588] [16147] (message_queue.cc:517): Message took 414ms to dispatch. Posted from: SetRemoteContent@../../pc/channel.cc:302\n[07:59:38:INFO] [001:588] [16147] (channel.cc:1061): Setting remote video description\n[07:59:38:INFO] [001:588] [16147] (webrtc_video_engine.cc:681): SetSendParameters: {codecs: [VideoCodec[96:H264], VideoCodec[97:rtx], VideoCodec[98:H264], VideoCodec[99:rtx], VideoCodec[100:VP8], VideoCodec[101:rtx], VideoCodec[127:VP9], VideoCodec[123:rtx], VideoCodec[125:red], VideoCodec[122:rtx], VideoCodec[124:ulpfec]], conference_mode: no, extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {uri: urn:3gpp:video-orientation, id: 12}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: 1}\n[07:59:38:INFO] [001:589] [16147] (webrtc_video_engine.cc:690): Using codec: VideoCodec[96:H264]\n[07:59:38:INFO] [001:589] [16147] (webrtc_video_engine.cc:1883): RecreateWebRtcStream (send) because of SetCodec.\n[07:59:38:INFO] [001:589] [16147] (alr_experiment.cc:78): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR start bandwidth usage percent: 80, ALR end budget level percent: 40, ALR end budget level percent: -60, ALR experiment group ID: 3\n[07:59:38:INFO] [001:590] [5635] (rtp_video_sender.cc:516): Transmitting payload type without picture ID using NACK+ULPFEC is a waste of bandwidth since ULPFEC packets also have to be retransmitted. Disabling ULPFEC.\n[07:59:38:INFO] [001:590] [5635] (video_send_stream_impl.cc:250): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [2869273597], rtcp_mode: RtcpMode::kReducedSize, max_packet_size: 1200, extmap-allow-mixed: false, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 12}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 124, red_payload_type: 125, red_rtx_payload_type: 122}, payload_name: H264, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [673344836], payload_type: 97}, c_name: SVCyZMBheTeYRPl/}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off}\n[07:59:38:INFO] [001:590] [5635] (video_send_stream_impl.cc:278): ERROR: Initial encoder max bitrate = -1 which is <= 0!\n[07:59:38:INFO] [001:590] [5635] (video_send_stream_impl.cc:280): Using default encoder max bitrate = 10 Mbps\n[07:59:38:INFO] [001:590] [16147] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:38:INFO] [001:590] [5635] (video_stream_encoder.cc:628): ConfigureEncoder requested.\n[07:59:38:INFO] [001:590] [16147] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:38:INFO] [001:590] [16147] (video_send_stream.cc:160): VideoSendStream::Stop\n[07:59:38:INFO] [001:590] [16147] (webrtc_video_engine.cc:758): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed.\n[07:59:38:INFO] [001:590] [5635] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[07:59:38:INFO] [001:590] [16147] (webrtc_video_engine.cc:1174): AddRecvStream: {id:ARDAMSv0;ssrcs:[3763443946,154119722];ssrc_groups:{semantics:FID;ssrcs:[3763443946,154119722]};cname:MTT4IWF1V9de/uJX;stream_ids:ARDAMS;}\n[07:59:38:INFO] [001:592] [16147] (video_receive_stream.cc:222): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 127, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 3763443946, local_ssrc: 2869273597, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 124, red_type: 125, rtx_ssrc: 154119722, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 122 (pt) -> 125 (apt), 123 (pt) -> 127 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 12}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0}\n[07:59:38:INFO] [001:592] [16147] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:38:INFO] [001:592] [16147] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:38:INFO] [001:592] [16147] (channel.cc:719): Add remote ssrc: 3763443946\n[07:59:38:INFO] [001:593] [16147] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers\n[07:59:38:INFO] [001:593] [5635] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers\n[07:59:38:INFO] [001:593] [5635] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 32000bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10032000bps\n[07:59:38:INFO] [001:593] [16147] (channel.cc:996): Changing video state, send=1\n[07:59:38:INFO] [001:593] [16147] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[07:59:38:INFO] [001:593] [23563] (peer_connection.cc:4109): Changing IceConnectionState 0 => 1\n[07:59:38:INFO] [001:593] [16147] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[07:59:38:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 1\n[07:59:38:INFO] [001:593] [16147] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[07:59:38:INFO] [001:593] [23563] (peer_connection.cc:5518): Non-rejected SCTP m= section is needed to get the SSL Role of the SCTP transport.\n[07:59:38:INFO] [001:593] [5635] (rtp_transport_controller_send.cc:459): Creating fallback congestion controller\n[07:59:38:INFO] [001:593] [16147] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[07:59:38:INFO] [001:593] [23563] (peer_connection.cc:2745): Processing the MSIDs for MID=0 (streams=[ARDAMS]).\n[07:59:38:INFO] [001:593] [5635] (alr_experiment.cc:78): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR start bandwidth usage percent: 80, ALR end budget level percent: 40, ALR end budget level percent: -60, ALR experiment group ID: 3\n[07:59:38:INFO] [001:593] [5635] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85\n[07:59:38:INFO] [001:593] [23563] (peer_connection.cc:2756): Processing the addition of a remote track for MID=0.\n[07:59:38:INFO] [001:593] [5635] (delay_based_bwe.cc:107): Using Trendline filter for delay change estimation with window size 20\n[07:59:38:INFO] [001:593] [16147] (channel.cc:863): Changing voice state, recv=1 send=1\n[07:59:38:INFO] [001:593] [23563] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv.\n[07:59:38:INFO] [001:593] [5635] (delay_based_bwe.cc:297): BWE Setting start bitrate to: 300 kbps\n[07:59:38:INFO] [001:594] [5635] (probe_controller.cc:266): Measured bitrate: 300000 Minimum to probe further: 1260000\n[07:59:38:INFO] [001:594] [5635] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (900000:1687:5)\n[07:59:38:INFO] [001:594] [5635] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5)\n[07:59:38:INFO] [001:594] [16147] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers\n[07:59:38:INFO] [001:594] [5635] (bitrate_allocator.cc:115): Current BWE 300000\n[07:59:38:INFO] [001:594] [9987] (video_stream_encoder.cc:1568): Video suspend state changed to: not suspended\n[07:59:38:INFO] [001:594] [5635] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers\n[07:59:38:INFO] [001:594] [16147] (channel.cc:996): Changing video state, send=1\n[07:59:38:INFO] [001:594] [16147] (webrtc_voice_engine.cc:1979): SetOutputVolume() to 1 for recv stream with ssrc 4074878902\n[07:59:38:INFO] [001:594] [23563] (peer_connection.cc:2745): Processing the MSIDs for MID=1 (streams=[ARDAMS]).\n[07:59:38:INFO] [001:594] [23563] (peer_connection.cc:2756): Processing the addition of a remote track for MID=1.\n[07:59:38:INFO] [001:594] [23563] (rtp_transceiver.cc:169): Changing transceiver (MID=1) current direction from <not set> to kSendRecv.\n[07:59:38:INFO] [001:594] [16147] (webrtc_video_engine.cc:1304): SetSink: ssrc:3763443946 (ptr)\n[07:59:38:INFO] [001:595] [23563] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x614000015530): created DTMF sender: RTCDtmfSender {\n[07:59:38:INFO] remainingTones:\n[07:59:38:INFO] duration: 0.100000 sec\n[07:59:38:INFO] interToneGap: 0.050000 sec\n[07:59:38:INFO] }\n[07:59:38:INFO] [001:595] [23563] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x61400023c6e0): created sender: RTCRtpSender {\n[07:59:38:INFO] senderId: ARDAMSa0\n[07:59:38:INFO] }\n[07:59:38:INFO] [001:595] [23563] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x61400005a340): created receiver: RTCRtpReceiver {\n[07:59:38:INFO] receiverId: 02e154fc-5824-4435-8f57-eb1d2c6dbfc1\n[07:59:38:INFO] }\n[07:59:38:INFO] [001:595] [23563] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x61400005a490): created transceiver: RTCRtpTransceiver {\n[07:59:38:INFO] sender: RTCRtpSender {\n[07:59:38:INFO] senderId: ARDAMSa0\n[07:59:38:INFO] }\n[07:59:38:INFO] receiver: RTCRtpReceiver {\n[07:59:38:INFO] receiverId: 02e154fc-5824-4435-8f57-eb1d2c6dbfc1\n[07:59:38:INFO] }\n[07:59:38:INFO] }\n[07:59:38:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving audio on track 02e154fc-5824-4435-8f57-eb1d2c6dbfc1.\n[07:59:38:INFO] [001:595] [23563] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[07:59:38:INFO] [001:595] [23563] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x61400023c340): created sender: RTCRtpSender {\n[07:59:38:INFO] senderId: ARDAMSv0\n[07:59:38:INFO] }\n[07:59:38:INFO] [001:595] [23563] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x61400005a280): created receiver: RTCRtpReceiver {\n[07:59:38:INFO] receiverId: ef7ff4ed-10d0-4351-8398-2efa37dc555c\n[07:59:38:INFO] }\n[07:59:38:INFO] [001:595] [23563] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x61400005a490): created transceiver: RTCRtpTransceiver {\n[07:59:38:INFO] sender: RTCRtpSender {\n[07:59:38:INFO] senderId: ARDAMSv0\n[07:59:38:INFO] }\n[07:59:38:INFO] receiver: RTCRtpReceiver {\n[07:59:38:INFO] receiverId: ef7ff4ed-10d0-4351-8398-2efa37dc555c\n[07:59:38:INFO] }\n[07:59:38:INFO] }\n[07:59:38:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving video on track ef7ff4ed-10d0-4351-8398-2efa37dc555c.\n[07:59:38:INFO] (RTCLogging.mm:33): (ARDAppClient.m:381 -[ARDAppClient peerConnection:didAddStream:]): Stream with 1 video tracks and 1 audio tracks was added.\n[07:59:38:INFO] [001:596] [23563] (message_queue.cc:517): Message took 428ms to dispatch. Posted from: SetRemoteDescription@../../api/peer_connection_proxy.h:104\n[07:59:38:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 1\n[07:59:38:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 2\n[07:59:38:INFO] [001:596] [23563] (peer_connection.cc:6050): Changing to ICE completed state because all transports are complete.\n[07:59:38:INFO] [001:596] [23563] (peer_connection.cc:4109): Changing IceConnectionState 1 => 2\n[07:59:38:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 2\n[07:59:38:INFO] [001:596] [23563] (peer_connection.cc:4109): Changing IceConnectionState 2 => 3\n[07:59:38:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 3\n[07:59:38:INFO] [001:596] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x61000000a2c0): created DTMF sender: RTCDtmfSender {\n[07:59:38:INFO] remainingTones:\n[07:59:38:INFO] duration: 0.100000 sec\n[07:59:38:INFO] interToneGap: 0.050000 sec\n[07:59:38:INFO] }\n[07:59:38:INFO] [001:596] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x618000034c00): created sender: RTCRtpSender {\n[07:59:38:INFO] senderId: ARDAMSa0\n[07:59:38:INFO] }\n[07:59:38:INFO] [001:596] [41991] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[07:59:38:INFO] [001:596] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600000035d40): created sender: RTCRtpSender {\n[07:59:38:INFO] senderId: ARDAMSv0\n[07:59:38:INFO] }\n[07:59:38:INFO] [001:598] [22539] (jsep_transport_controller.cc:318): Not adding candidate because the JsepTransport doesn't exist. Ignore it.\n[07:59:38:INFO] [001:601] [40455] (rtc_event_log_impl.cc:211): Stopping WebRTC event log.\n[07:59:38:INFO] [001:601] [40455] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped.\n[07:59:38:INFO] [001:603] [40455] (webrtc_video_engine.cc:2234): VideoSendStream stats: 274539, {input_fps: 0, encode_fps: 0, encode_ms: 0, encode_usage_perc: 0, target_bps: 268000, media_bps: 0, suspended: false, bw_adapted: false} {ssrc: 3763443946, width: 0, height: 0, key: 0, delta: 0, total_bps: 0, retransmit_bps: 0, avg_delay_ms: 0, max_delay_ms: 0, cum_loss: 0, max_ext_seq: 0, nack: 0, fir: 0, pli: 0}\n[07:59:38:INFO] [001:603] [40455] (webrtc_video_engine.cc:2731): VideoReceiveStream stats: 274539, {ssrc: 2869273597, total_bps: 0, width: 0, height: 0, key: 0, delta: 0, network_fps: 0, decode_fps: 0, render_fps: 0, decode_ms: 0, max_decode_ms: 0, first_frame_received_to_decoded_ms: -1, cur_delay_ms: 0, targ_delay_ms: 0, jb_delay_ms: 0, min_playout_delay_ms: 0, sync_offset_ms: 2147483647, cum_loss: 0, max_ext_seq: 0, nack: 0, fir: 0, pli: 0}\n[07:59:38:INFO] [001:603] [40455] (webrtc_video_engine.cc:1349): Call stats: 274539, {send_bw_bps: 300000, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1}\n[07:59:38:INFO] [001:603] [41991] (peer_connection.cc:4173): Session: 2723856753833851992 Old state: kStable New state: kClosed\n[07:59:38:INFO] [001:603] [5635] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps\n[07:59:38:INFO] [001:604] [40455] (audio_device_module_ios.mm:563): StopRecording\n[07:59:38:INFO] [001:604] [40455] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording\n[07:59:38:INFO] [001:604] [40455] (audio_device_buffer.cc:153): StopRecording\n[07:59:38:INFO] [001:604] [40455] (audio_device_buffer.cc:176): total recording time: 402\n[07:59:38:INFO] [001:604] [40455] (audio_device_module_ios.mm:567): output: 0\n[07:59:38:INFO] [001:604] [40455] (webrtc_voice_engine.cc:1979): SetOutputVolume() to 0 for recv stream with ssrc 2609444602\n[07:59:38:INFO] [001:604] [40455] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 3763443946, options: nullptr, source = nullptr)\n[07:59:38:INFO] [001:604] [40455] (webrtc_video_engine.cc:1304): SetSink: ssrc:2869273597 nullptr\n[07:59:38:INFO] [001:605] [40455] (channel.cc:563): Channel disabled\n[07:59:38:INFO] [001:605] [40455] (video_send_stream.cc:160): VideoSendStream::Stop\n[07:59:38:INFO] [001:605] [40455] (channel.cc:996): Changing video state, send=0\n[07:59:38:INFO] [001:605] [5635] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[07:59:38:INFO] [001:605] [5635] (rtp_rtcp_impl.cc:360): Failed to send RTCP BYE\n[07:59:38:INFO] [001:605] [5635] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps\n[07:59:38:INFO] [001:605] [5635] (video_stream_encoder.cc:1568): Video suspend state changed to: suspended\n[07:59:38:INFO] [001:605] [40455] (video_send_stream.cc:160): VideoSendStream::Stop\n[07:59:38:INFO] [001:605] [5635] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[07:59:38:INFO] [001:605] [5635] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[07:59:38:INFO] [001:605] [5635] (video_send_stream_impl.cc:338): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [3763443946], rtcp_mode: RtcpMode::kReducedSize, max_packet_size: 1200, extmap-allow-mixed: false, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 12}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 124, red_payload_type: 125, red_rtx_payload_type: 122}, payload_name: H264, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [154119722], payload_type: 97}, c_name: MTT4IWF1V9de/uJX}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off}\n[07:59:38:INFO] [001:606] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[07:59:38:INFO] [001:606] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[07:59:38:INFO] [001:607] [40455] (send_statistics_proxy.cc:655): Frames encoded 0\n[07:59:38:INFO] WebRTC.Video.DroppedFrames.Capturer 0\n[07:59:38:INFO] WebRTC.Video.DroppedFrames.EncoderQueue 0\n[07:59:38:INFO] WebRTC.Video.DroppedFrames.Encoder 0\n[07:59:38:INFO] WebRTC.Video.DroppedFrames.Ratelimiter 0\n[07:59:38:INFO] [001:607] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[07:59:38:INFO] [001:607] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[07:59:38:INFO] [001:607] [40455] (video_receive_stream.cc:291): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 127, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 2869273597, local_ssrc: 3763443946, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 124, red_type: 125, rtx_ssrc: 673344836, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 122 (pt) -> 125 (apt), 123 (pt) -> 127 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 12}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0}\n[07:59:38:INFO] [001:608] [40455] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0\n[07:59:38:INFO] [001:608] [40455] (receive_statistics_proxy.cc:487): Frames decoded 0\n[07:59:38:INFO] WebRTC.Video.DroppedFrames.Receiver 0\n[07:59:38:INFO] \n[07:59:38:INFO] [001:608] [40455] (channel.cc:162): Destroyed channel: 1\n[07:59:38:INFO] [001:608] [40455] (channel.cc:563): Channel disabled\n[07:59:38:INFO] [001:608] [40455] (audio_device_module_ios.mm:532): StopPlayout\n[07:59:38:INFO] [001:608] [40455] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout\n[07:59:38:INFO] [001:608] [40455] (audio_device_ios.mm:903): AudioDeviceIOS::ShutdownPlayOrRecord\n[07:59:38:INFO] [001:608] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:331 Stop): Stopping audio unit.\n[07:59:38:INFO] [001:608] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:338 Stop): Stopped audio unit\n[07:59:38:INFO] [001:608] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:347 Uninitialize): Unintializing audio unit.\n[07:59:38:INFO] [001:608] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:354 Uninitialize): Uninitialized audio unit.\n[07:59:38:INFO] [001:608] [40455] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:459 DisposeAudioUnit): Disposing audio unit.\n[07:59:38:INFO] [001:609] [40455] (RTCLogging.mm:33): (RTCAudioSession.mm:196 -[RTCAudioSession removeDelegate:]): Removing delegate: (0x60000000a1f0)\n[07:59:38:INFO] [001:609] [40455] (RTCLogging.mm:33): (audio_device_ios.mm:844 UnconfigureAudioSession): Unconfiguring audio session.\n[07:59:38:INFO] [001:609] [40455] (RTCLogging.mm:33): (RTCAudioSession.mm:762 -[RTCAudioSession unconfigureWebRTCSession:]): Unconfiguring audio session for WebRTC.\n[07:59:38:INFO] [001:609] [40455] (RTCLogging.mm:33): (RTCAudioSession.mm:629 -[RTCAudioSession decrementActivationCount]): Decrementing activation count.\n[07:59:38:INFO] [001:610] [40455] (RTCLogging.mm:33): (RTCAudioSession.mm:377 -[RTCAudioSession setActive:error:]): Number of current activations: 1\n[07:59:38:INFO] [001:610] [40455] (RTCLogging.mm:33): (audio_device_ios.mm:855 UnconfigureAudioSession): Unconfigured audio session.\n[07:59:38:INFO] [001:610] [40455] (RTCLogging.mm:33): (audio_device_ios.mm:271 StopPlayout): Average number of playout callbacks between glitches: 100000\n[07:59:38:INFO] [001:610] [40455] (audio_device_buffer.cc:139): StopPlayout\n[07:59:38:INFO] [001:610] [40455] (audio_device_buffer.cc:145): total playout time: 466\n[07:59:38:INFO] [001:610] [40455] (audio_device_module_ios.mm:536): output: 0\n[07:59:38:INFO] [001:610] [40455] (channel.cc:863): Changing voice state, recv=0 send=0\n[07:59:38:INFO] [001:611] [40455] (webrtc_voice_engine.cc:1840): RemoveSendStream: 4074878902\n[07:59:38:INFO] [001:611] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[07:59:38:INFO] [001:611] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[07:59:38:INFO] [001:611] [40455] (audio_send_stream.cc:179): ~AudioSendStream: 4074878902\n[07:59:38:INFO] [001:611] [40455] (webrtc_voice_engine.cc:1917): RemoveRecvStream: 2609444602\n[07:59:38:INFO] [001:611] [40455] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:38:INFO] [001:611] [40455] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:38:INFO] [001:611] [40455] (audio_receive_stream.cc:140): ~AudioReceiveStream: 2609444602\n[07:59:38:INFO] [001:611] [9735] (paced_sender.cc:106): PacedSender paused.\n[07:59:38:INFO] [001:611] [9735] (control_handler.cc:77): Bitrate estimate state changed, BWE: 300 kbps.\n[07:59:38:INFO] [001:611] [40455] (channel.cc:162): Destroyed channel: 0\n[07:59:38:INFO] [001:611] [22539] (openssl_stream_adapter.cc:896): Cleanup\n[07:59:38:INFO] [001:611] [22539] (openssl_adapter.cc:787): SSL3 alert write:warning:close notify\n[07:59:38:INFO] [001:612] [23311] (openssl_adapter.cc:787): SSL3 alert read:warning:close notify\n[07:59:38:INFO] [001:612] [23311] (openssl_stream_adapter.cc:896): Cleanup\n[07:59:38:INFO] [001:612] [23311] (openssl_adapter.cc:787): SSL3 alert write:warning:close notify\n[07:59:38:INFO] [001:612] [23311] (dtls_transport.cc:651): DtlsTransport[0|1|_W]: DTLS transport closed\n[07:59:38:INFO] [001:612] [23311] (jsep_transport_controller.cc:1213): Transport 0 writability changed to 0.\n[07:59:38:INFO] [001:612] [23311] (channel.cc:596): Channel not writable (0)\n[07:59:38:INFO] [001:612] [23563] (peer_connection.cc:4109): Changing IceConnectionState 3 => 5\n[07:59:38:INFO] [001:612] [23311] (channel.cc:596): Channel not writable (1)\n[07:59:38:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 5\n[07:59:38:INFO] [001:612] [23311] (srtp_transport.cc:364): The params in SRTP transport are reset.\n[07:59:38:INFO] [001:612] [16147] (channel.cc:863): Changing voice state, recv=1 send=1\n[07:59:38:INFO] [001:613] [16147] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers\n[07:59:38:INFO] [001:613] [9735] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers\n[07:59:38:INFO] [001:613] [16147] (channel.cc:996): Changing video state, send=1\n[07:59:38:INFO] [001:613] [40455] (paced_sender.cc:418): ProcessThreadAttached 0x0\n[07:59:38:INFO] [001:613] [40455] (paced_sender.cc:418): ProcessThreadAttached 0x0\n[07:59:38:INFO] [001:613] [40455] (rtc_event_log_impl.cc:211): Stopping WebRTC event log.\n[07:59:38:INFO] [001:613] [40455] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped.\n[07:59:38:INFO] [001:613] [41991] (peer_connection.cc:6776): Usage signature is 4088\n[07:59:38:INFO] [001:614] [41991] (peer_connection.cc:915): Session: 2723856753833851992 is destroyed.\n[07:59:38:INFO] [001:614] [41991] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue.\n[07:59:38:INFO] [001:614] [40455] (webrtc_video_engine.cc:474): WebRtcVideoEngine::~WebRtcVideoEngine\n[07:59:38:INFO] [001:614] [40455] (webrtc_voice_engine.cc:205): WebRtcVoiceEngine::~WebRtcVoiceEngine\n[07:59:38:INFO] [001:614] [40455] (audio_device_module_ios.mm:532): StopPlayout\n[07:59:38:INFO] [001:614] [40455] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout\n[07:59:38:INFO] [001:614] [40455] (audio_device_module_ios.mm:536): output: 0\n[07:59:38:INFO] [001:614] [40455] (audio_device_module_ios.mm:563): StopRecording\n[07:59:38:INFO] [001:614] [40455] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording\n[07:59:38:INFO] [001:614] [40455] (audio_device_module_ios.mm:567): output: 0\n[07:59:38:INFO] [001:614] [40455] (audio_device_module_ios.mm:581): RegisterAudioCallback\n[07:59:38:INFO] [001:614] [40455] (audio_device_buffer.cc:81): RegisterAudioCallback\n[07:59:38:INFO] [001:614] [40455] (audio_device_module_ios.mm:92): Terminate\n[07:59:38:INFO] [001:615] [40455] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate\n[07:59:38:INFO] [001:615] [40455] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout\n[07:59:38:INFO] [001:615] [40455] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording\n[07:59:38:INFO] [001:616] [40455] (audio_device_module_ios.mm:55): ~AudioDeviceModuleIOS\n[07:59:38:INFO] [001:616] [40455] (audio_device_buffer.cc:75): AudioDeviceBuffer::~dtor\n[07:59:38:INFO] [001:616] [40455] (audio_device_ios.mm:127): AudioDeviceIOS::~dtor<NSThread: 0x600000069400>{number = 4, name = (null)}\n[07:59:38:INFO] [001:616] [40455] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate\n[07:59:38:INFO] [001:616] [40455] (fine_audio_buffer.cc:46): ~FineAudioBuffer\n[07:59:38:INFO] Test Case '-[ARDAppClientTest testSession]' passed (1.631 seconds).\n[07:59:38:INFO] Test Suite 'ARDAppClientTest' passed at 2019-04-17 07:59:38.180.\n[07:59:38:INFO] \t Executed 1 test, with 0 failures (0 unexpected) in 1.631 (1.631) seconds\n[07:59:38:INFO] Test Suite 'ARDFileCaptureControllerTests' started at 2019-04-17 07:59:38.181\n[07:59:38:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStarted]' started.\n[07:59:38:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStarted]' passed (0.001 seconds).\n[07:59:38:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStoped]' started.\n[07:59:38:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStoped]' passed (0.001 seconds).\n[07:59:38:INFO] Test Suite 'ARDFileCaptureControllerTests' passed at 2019-04-17 07:59:38.183.\n[07:59:38:INFO] \t Executed 2 tests, with 0 failures (0 unexpected) in 0.001 (0.002) seconds\n[07:59:38:INFO] Test Suite 'ARDSettingsModelTests' started at 2019-04-17 07:59:38.184\n[07:59:38:INFO] Test Case '-[ARDSettingsModelTests testConstraintComponentIsNilWhenInvalidConstraintString]' started.\n[07:59:38:INFO] Test Case '-[ARDSettingsModelTests testConstraintComponentIsNilWhenInvalidConstraintString]' passed (0.001 seconds).\n[07:59:38:INFO] Test Case '-[ARDSettingsModelTests testHeightConstraintFromStore]' started.\n[07:59:38:INFO] Test Case '-[ARDSettingsModelTests testHeightConstraintFromStore]' passed (0.001 seconds).\n[07:59:38:INFO] Test Case '-[ARDSettingsModelTests testRetrievingSetting]' started.\n[07:59:38:INFO] Test Case '-[ARDSettingsModelTests testRetrievingSetting]' passed (0.001 seconds).\n[07:59:38:INFO] Test Case '-[ARDSettingsModelTests testReturningDefaultCallOption]' started.\n[07:59:38:INFO] Test Case '-[ARDSettingsModelTests testReturningDefaultCallOption]' passed (0.001 seconds).\n[07:59:38:INFO] Test Case '-[ARDSettingsModelTests testStoringAudioSetting]' started.\n[07:59:38:INFO] Test Case '-[ARDSettingsModelTests testStoringAudioSetting]' passed (0.001 seconds).\n[07:59:38:INFO] Test Case '-[ARDSettingsModelTests testStoringInvalidConstraintReturnsNo]' started.\n[07:59:38:INFO] Test Case '-[ARDSettingsModelTests testStoringInvalidConstraintReturnsNo]' passed (0.001 seconds).\n[07:59:38:INFO] Test Case '-[ARDSettingsModelTests testWidthConstraintFromStore]' started.\n[07:59:38:INFO] Test Case '-[ARDSettingsModelTests testWidthConstraintFromStore]' passed (0.001 seconds).\n[07:59:38:INFO] Test Suite 'ARDSettingsModelTests' passed at 2019-04-17 07:59:38.193.\n[07:59:38:INFO] \t Executed 7 tests, with 0 failures (0 unexpected) in 0.007 (0.010) seconds\n[07:59:38:INFO] Test Suite 'apprtcmobile_tests.app' passed at 2019-04-17 07:59:38.194.\n[07:59:38:INFO] \t Executed 10 tests, with 0 failures (0 unexpected) in 1.639 (1.645) seconds\n[07:59:38:INFO] Test Suite 'All tests' passed at 2019-04-17 07:59:38.194.\n[07:59:38:INFO] \t Executed 10 tests, with 0 failures (0 unexpected) in 1.639 (1.647) seconds\n[07:59:38:INFO] [001:633] [16147] (rtc_event_log_impl.cc:211): Stopping WebRTC event log.\n[07:59:38:INFO] [001:633] [16147] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped.\n[07:59:38:INFO] [001:635] [23563] (peer_connection.cc:4173): Session: 892703080514118717 Old state: kStable New state: kClosed\n[07:59:38:INFO] [001:635] [9735] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps\n[07:59:38:INFO] [001:635] [16147] (audio_device_module_ios.mm:563): StopRecording\n[07:59:38:INFO] [001:635] [16147] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording\n[07:59:38:INFO] [001:635] [16147] (audio_device_buffer.cc:153): StopRecording\n[07:59:38:INFO] [001:635] [16147] (audio_device_buffer.cc:176): total recording time: 47\n[07:59:38:INFO] [001:635] [16147] (audio_device_module_ios.mm:567): output: 0\n[07:59:38:INFO] [001:635] [16147] (webrtc_voice_engine.cc:1979): SetOutputVolume() to 0 for recv stream with ssrc 4074878902\n[07:59:38:INFO] [001:635] [16147] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 2869273597, options: nullptr, source = nullptr)\n[07:59:38:INFO] [001:635] [16147] (webrtc_video_engine.cc:1304): SetSink: ssrc:3763443946 nullptr\n[07:59:38:INFO] [001:636] [16147] (channel.cc:563): Channel disabled\n[07:59:38:INFO] [001:636] [16147] (video_send_stream.cc:160): VideoSendStream::Stop\n[07:59:38:INFO] [001:636] [16147] (channel.cc:996): Changing video state, send=0\n[07:59:38:INFO] [001:636] [9735] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[07:59:38:INFO] [001:636] [9735] (rtp_rtcp_impl.cc:360): Failed to send RTCP BYE\n[07:59:38:INFO] [001:636] [9735] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps\n[07:59:38:INFO] [001:636] [9735] (video_stream_encoder.cc:1568): Video suspend state changed to: suspended\n[07:59:38:INFO] [001:636] [16147] (video_send_stream.cc:160): VideoSendStream::Stop\n[07:59:38:INFO] [001:636] [9735] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[07:59:38:INFO] [001:636] [3075] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[07:59:38:INFO] [001:636] [3075] (video_send_stream_impl.cc:338): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [2869273597], rtcp_mode: RtcpMode::kReducedSize, max_packet_size: 1200, extmap-allow-mixed: false, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 12}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 124, red_payload_type: 125, red_rtx_payload_type: 122}, payload_name: H264, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [673344836], payload_type: 97}, c_name: SVCyZMBheTeYRPl/}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off}\n[07:59:38:INFO] [001:637] [16147] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[07:59:38:INFO] [001:637] [16147] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[07:59:38:INFO] [001:637] [16147] (send_statistics_proxy.cc:655): Frames encoded 0\n[07:59:38:INFO] WebRTC.Video.DroppedFrames.Capturer 0\n[07:59:38:INFO] WebRTC.Video.DroppedFrames.EncoderQueue 0\n[07:59:38:INFO] WebRTC.Video.DroppedFrames.Encoder 0\n[07:59:38:INFO] WebRTC.Video.DroppedFrames.Ratelimiter 0\n[07:59:38:INFO] [001:637] [16147] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[07:59:38:INFO] [001:637] [16147] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[07:59:38:INFO] [001:637] [16147] (video_receive_stream.cc:291): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 127, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 3763443946, local_ssrc: 2869273597, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 124, red_type: 125, rtx_ssrc: 154119722, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 122 (pt) -> 125 (apt), 123 (pt) -> 127 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 8}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 13}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 9}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 11}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 7}, {uri: urn:3gpp:video-orientation, id: 12}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 3}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 5}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 4}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0}\n[07:59:38:INFO] [001:637] [16147] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0\n[07:59:38:INFO] [001:637] [16147] (receive_statistics_proxy.cc:487): Frames decoded 0\n[07:59:38:INFO] WebRTC.Video.DroppedFrames.Receiver 0\n[07:59:38:INFO] \n[07:59:38:INFO] [001:638] [16147] (channel.cc:162): Destroyed channel: 1\n[07:59:38:INFO] [001:638] [16147] (channel.cc:563): Channel disabled\n[07:59:38:INFO] [001:638] [16147] (audio_device_module_ios.mm:532): StopPlayout\n[07:59:38:INFO] [001:638] [16147] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout\n[07:59:38:INFO] [001:638] [16147] (audio_device_ios.mm:903): AudioDeviceIOS::ShutdownPlayOrRecord\n[07:59:38:INFO] [001:638] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:331 Stop): Stopping audio unit.\n[07:59:38:INFO] [001:638] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:338 Stop): Stopped audio unit\n[07:59:38:INFO] [001:638] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:347 Uninitialize): Unintializing audio unit.\n[07:59:38:INFO] [001:638] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:354 Uninitialize): Uninitialized audio unit.\n[07:59:38:INFO] [001:638] [16147] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:459 DisposeAudioUnit): Disposing audio unit.\n[07:59:38:INFO] [001:638] [16147] (RTCLogging.mm:33): (RTCAudioSession.mm:196 -[RTCAudioSession removeDelegate:]): Removing delegate: (0x61000000a3a0)\n[07:59:38:INFO] [001:638] [16147] (RTCLogging.mm:33): (audio_device_ios.mm:844 UnconfigureAudioSession): Unconfiguring audio session.\n[07:59:38:INFO] [001:638] [16147] (RTCLogging.mm:33): (RTCAudioSession.mm:762 -[RTCAudioSession unconfigureWebRTCSession:]): Unconfiguring audio session for WebRTC.\n[07:59:38:INFO] [001:638] [16147] (RTCLogging.mm:33): (RTCAudioSession.mm:629 -[RTCAudioSession decrementActivationCount]): Decrementing activation count.\n[07:59:38:INFO] [001:638] [16147] (RTCLogging.mm:33): (RTCAudioSession.mm:377 -[RTCAudioSession setActive:error:]): Number of current activations: 0\n[07:59:38:INFO] [001:638] [16147] (RTCLogging.mm:33): (audio_device_ios.mm:855 UnconfigureAudioSession): Unconfigured audio session.\n[07:59:38:INFO] [001:638] [16147] (RTCLogging.mm:33): (audio_device_ios.mm:271 StopPlayout): Average number of playout callbacks between glitches: 100000\n[07:59:38:INFO] [001:639] [16147] (audio_device_buffer.cc:139): StopPlayout\n[07:59:38:INFO] [001:639] [16147] (audio_device_buffer.cc:145): total playout time: 53\n[07:59:38:INFO] [001:639] [16147] (audio_device_module_ios.mm:536): output: 0\n[07:59:38:INFO] [001:639] [16147] (channel.cc:863): Changing voice state, recv=0 send=0\n[07:59:38:INFO] [001:639] [16147] (webrtc_voice_engine.cc:1840): RemoveSendStream: 2609444602\n[07:59:38:INFO] [001:639] [16147] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[07:59:38:INFO] [001:639] [16147] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[07:59:38:INFO] [001:639] [16147] (audio_send_stream.cc:179): ~AudioSendStream: 2609444602\n[07:59:38:INFO] [001:640] [16147] (webrtc_voice_engine.cc:1917): RemoveRecvStream: 4074878902\n[07:59:38:INFO] [001:640] [16147] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[07:59:38:INFO] [001:640] [16147] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[07:59:38:INFO] [001:640] [16147] (audio_receive_stream.cc:140): ~AudioReceiveStream: 4074878902\n[07:59:38:INFO] [001:640] [3075] (paced_sender.cc:106): PacedSender paused.\n[07:59:38:INFO] [001:640] [3075] (control_handler.cc:77): Bitrate estimate state changed, BWE: 300 kbps.\n[07:59:38:INFO] [001:640] [16147] (channel.cc:162): Destroyed channel: 0\n[07:59:38:INFO] [001:640] [23311] (openssl_stream_adapter.cc:896): Cleanup\n[07:59:38:INFO] [001:641] [16147] (paced_sender.cc:418): ProcessThreadAttached 0x0\n[07:59:38:INFO] [001:641] [16147] (paced_sender.cc:418): ProcessThreadAttached 0x0\n[07:59:38:INFO] [001:641] [16147] (rtc_event_log_impl.cc:211): Stopping WebRTC event log.\n[07:59:38:INFO] [001:641] [16147] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped.\n[07:59:38:INFO] [001:641] [23563] (peer_connection.cc:6776): Usage signature is 4088\n[07:59:38:INFO] [001:641] [23563] (peer_connection.cc:915): Session: 892703080514118717 is destroyed.\n[07:59:38:INFO] [001:641] [23563] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue.\n[07:59:38:INFO] [001:642] [16147] (webrtc_video_engine.cc:474): WebRtcVideoEngine::~WebRtcVideoEngine\n[07:59:38:INFO] [001:642] [16147] (webrtc_voice_engine.cc:205): WebRtcVoiceEngine::~WebRtcVoiceEngine\n[07:59:38:INFO] [001:642] [16147] (audio_device_module_ios.mm:532): StopPlayout\n[07:59:38:INFO] [001:642] [16147] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout\n[07:59:38:INFO] [001:642] [16147] (audio_device_module_ios.mm:536): output: 0\n[07:59:38:INFO] [001:642] [16147] (audio_device_module_ios.mm:563): StopRecording\n[07:59:38:INFO] [001:642] [16147] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording\n[07:59:38:INFO] [001:642] [16147] (audio_device_module_ios.mm:567): output: 0\n[07:59:38:INFO] [001:642] [16147] (audio_device_module_ios.mm:581): RegisterAudioCallback\n[07:59:38:INFO] [001:642] [16147] (audio_device_buffer.cc:81): RegisterAudioCallback\n[07:59:38:INFO] [001:642] [16147] (audio_device_module_ios.mm:92): Terminate\n[07:59:38:INFO] [001:642] [16147] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate\n[07:59:38:INFO] [001:642] [16147] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout\n[07:59:38:INFO] [001:642] [16147] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording\n[07:59:38:INFO] [001:643] [16147] (audio_device_module_ios.mm:55): ~AudioDeviceModuleIOS\n[07:59:38:INFO] [001:643] [16147] (audio_device_buffer.cc:75): AudioDeviceBuffer::~dtor\n[07:59:38:INFO] [001:643] [16147] (audio_device_ios.mm:127): AudioDeviceIOS::~dtor<NSThread: 0x610000068300>{number = 3, name = (null)}\n[07:59:38:INFO] [001:643] [16147] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate\n[07:59:38:INFO] [001:643] [16147] (fine_audio_buffer.cc:46): ~FineAudioBuffer\n[07:59:38:INFO] \n[07:59:38:INFO] \n[07:59:38:INFO] Test session results and logs:\n[07:59:38:INFO] \t/Users/chrome-bot/Library/Developer/Xcode/DerivedData/temporary-artzivirmkrfysewqiyzwhklfbfy/Logs/Test/Test-Transient Testing-2019.04.17_07-59-19--0700.xcresult\n[07:59:38:INFO] \n[07:59:38:INFO] 2019-04-17 07:59:38.562 xcodebuild[890:11868] [MT] IDETestOperationsObserverDebug: 19.261 elapsed -- Testing started completed.\n[07:59:38:INFO] 2019-04-17 07:59:38.562 xcodebuild[890:11868] [MT] IDETestOperationsObserverDebug: 0.000 sec, +0.000 sec -- start\n[07:59:38:INFO] 2019-04-17 07:59:38.562 xcodebuild[890:11868] [MT] IDETestOperationsObserverDebug: 19.261 sec, +19.261 sec -- end\n[07:59:38:INFO] ** TEST EXECUTE SUCCEEDED **\n[07:59:38:INFO] \n[07:59:44:DEBUG] Finished print_process_output.\n[07:59:44:INFO] Waiting for test process to terminate.\n[07:59:44:INFO] Test process terminated.\n[07:59:44:DEBUG] Setting sigterm handler.\n[07:59:44:DEBUG] Stdout flushed after test process.\n[07:59:44:DEBUG] Processing test results.\n[07:59:44:INFO] /b/s/w/ir/src/out/Debug-iphonesimulator/iossim returned 0\n\n[07:59:44:DEBUG] Extracting test data.\n[07:59:45:DEBUG] Retrieving crash reports.\n[07:59:45:DEBUG] Retrieving derived data.\n[07:59:45:DEBUG] Making desktop screenshots.\n[07:59:45:DEBUG] Killing simulators.\n[07:59:50:DEBUG] Wiping simulator.\nDevice wiped.\n[07:59:52:DEBUG] End of tear_down.\n", "outputs_ref": { "isolated": "0b347b87056f9a65b6a982faef78d03c6499eaf9", "isolatedserver": "https://isolateserver.appspot.com", "namespace": "default-gzip" }, "run_id": "44449fdc51a4c611", "server_versions": [ "4260-fb32526" ], "started_ts": "2019-04-17T14:58:55.977165", "state": "COMPLETED", "tags": [ "build_is_experimental:false", "buildername:iOS64 Sim Debug (iOS 11)", "buildnumber:829", "data:59830afca40bce17e4fb0a5978c0f5d86651f570", "device_type:iPhone 6s", "ios_version:11.4", "master:client.webrtc", "name:apprtcmobile_tests", "os:Mac-10.13", "platform:simulator", "pool:Chrome", "priority:30", "project:webrtc", "purpose:CI", "purpose:luci", "purpose:post-commit", "service_account:ios-isolated-tester@chops-service-accounts.iam.gserviceaccount.com", "slavename:build141-m9", "spec_name:webrtc.ci:iOS64 Sim Debug (iOS 11)", "stepname:apprtcmobile_tests (iPhone 6s iOS 11.4)", "swarming.pool.template:none", "swarming.pool.version:5e3aa1067cab64ab93b35d7f6148b82e1a0987f1", "test:apprtcmobile_tests", "user:None" ], "task_id": "44449fdc51a4c610", "try_number": "1" } ] }