Back to build
Raw log | Switch to full mode | Login

{ "shards": [ { "bot_dimensions": [ { "key": "caches", "value": [ "swarming_module_cache_vpython", "xcode_ios_10b61", "xcode_ios_10l232m", "xcode_ios_10o45e" ] }, { "key": "cores", "value": [ "4" ] }, { "key": "cpu", "value": [ "x86", "x86-64", "x86-64-i7-4578U" ] }, { "key": "gce", "value": [ "0" ] }, { "key": "gpu", "value": [ "8086", "8086:0a2e" ] }, { "key": "hidpi", "value": [ "0" ] }, { "key": "id", "value": [ "build17-m9" ] }, { "key": "mac_model", "value": [ "Macmini7,1" ] }, { "key": "machine_type", "value": [ "n1-standard-4" ] }, { "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": [ "4267-53b17d5" ] }, { "key": "ssd", "value": [ "1" ] }, { "key": "zone", "value": [ "us", "us-golo", "us-golo-9" ] } ], "bot_id": "build17-m9", "bot_version": "bd68e04e43fc4d1b7b5b65890a29c5d5bae55ab1423851733f92abd7e13dda14", "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": "2019-04-19T23:11:00.119069", "costs_usd": [ 0.010261170812036842 ], "created_ts": "2019-04-19T23:09:37.100753", "duration": 48.089081048965454, "modified_ts": "2019-04-19T23:11:00.119069", "name": "apprtcmobile_tests (iPhone 6s iOS 11.4)/Mac-10.13/04a2fc63de/iOS64 Sim Debug (iOS 11)/850", "output": "[I 2019-04-19 16:09:45] About to install Xcode 10l232m in Xcode.app for ios\n[04:09:47:INFO] Using Xcode version 10.0 build 10L232m at /b/s/w/ir/Xcode.app/Contents/Developer\nDevice wiped.\n[04:09:56:DEBUG] Setting sigterm handler.\n[04:09:57:INFO] User defaults from command line:\n[04:09:57:INFO] IDETestRunSpecificationPath = /var/folders/2j/22s2gz0s7hn48k32d47clxf80000gm/T/6DF31FC7-AE69-4098-A780-44E62AF009C8\n[04:09:57:INFO] \n[04:09:57:INFO] Testing started on 'iPhone 6s'\n[04:09:57:INFO] 2019-04-19 16:09:57.708 xcodebuild[7348:72805] IDETestOperationsObserverDebug: Writing diagnostic log for test session to:\n[04:09:57:INFO] /Users/chrome-bot/Library/Developer/Xcode/DerivedData/temporary-bzpukcbbdwktbsdaclipyolfvwfq/Logs/Test/Test-Transient Testing-2019.04.19_16-09-57--0700.xcresult/1_Test/Diagnostics/apprtcmobile_tests_module-FC74725B-AACA-4FD1-A9FF-2ABD56C9215A/apprtcmobile_tests_module-5AE6C737-E003-4058-8EAA-29BD21A71859/Session-apprtcmobile_tests_module-2019-04-19_160957-0Wpq0j.log\n[04:09:57:INFO] 2019-04-19 16:09:57.708 xcodebuild[7348:72779] [MT] IDETestOperationsObserverDebug: (5BFC188E-A716-42BB-A4A6-676CBBED6383) Beginning test session apprtcmobile_tests_module-5BFC188E-A716-42BB-A4A6-676CBBED6383 at 2019-04-19 16:09:57.709 with Xcode 10L232m on target <DVTiPhoneSimulator: 0x7f82066a6900> {\n[04:09:57:INFO] \t\tSimDevice: iPhone 6s (D0835354-E76F-4A91-8536-57615807E34E, iOS 11.4, Shutdown)\n[04:09:57:INFO] } (11.4 (15F79))\n[04:10:17:INFO] objc[7465]: Class RTCDispatcher is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114b268a0) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/apprtcmobile_tests (0x10f426330). One of the two will be used. Which one is undefined.\n[04:10:17:INFO] objc[7465]: Class RTCCameraPreviewView is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114b268c8) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/apprtcmobile_tests (0x10f426358). One of the two will be used. Which one is undefined.\n[04:10:17:INFO] objc[7465]: Class RTCEncodedImage is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114b25770) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/apprtcmobile_tests (0x10f4263d0). One of the two will be used. Which one is undefined.\n[04:10:17:INFO] objc[7465]: Class RTCRtpFragmentationHeader is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114b257c0) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/apprtcmobile_tests (0x10f426420). One of the two will be used. Which one is undefined.\n[04:10:17:INFO] objc[7465]: Class RTCVideoCapturer is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114b257e8) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/apprtcmobile_tests (0x10f426448). One of the two will be used. Which one is undefined.\n[04:10:17:INFO] objc[7465]: Class RTCVideoCodecInfo is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114b25838) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/apprtcmobile_tests (0x10f426498). One of the two will be used. Which one is undefined.\n[04:10:17:INFO] objc[7465]: Class RTCVideoEncoderQpThresholds is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114b25888) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/apprtcmobile_tests (0x10f4264e8). One of the two will be used. Which one is undefined.\n[04:10:17:INFO] objc[7465]: Class RTCVideoEncoderSettings is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114b25900) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/apprtcmobile_tests (0x10f426560). One of the two will be used. Which one is undefined.\n[04:10:17:INFO] objc[7465]: Class RTCVideoFrame is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x114b25928) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D0835354-E76F-4A91-8536-57615807E34E/data/Containers/Bundle/Application/2E7CE041-88F0-422E-A625-CBB6254D95C6/apprtcmobile_tests.app/apprtcmobile_tests (0x10f426588). One of the two will be used. Which one is undefined.\n[04:10:17:INFO] 2019-04-19 16:10:17.743 apprtcmobile_tests[7465:74234] +[CATransaction synchronize] called within transaction\n[04:10:17:INFO] Test Suite 'All tests' started at 2019-04-19 16:10:17.986\n[04:10:17:INFO] Test Suite 'apprtcmobile_tests.app' started at 2019-04-19 16:10:17.987\n[04:10:17:INFO] Test Suite 'ARDAppClientTest' started at 2019-04-19 16:10:17.987\n[04:10:17:INFO] Test Case '-[ARDAppClientTest testSession]' started.\n[04:10:17:INFO] [000:000] [771] (audio_device_module.mm:21): CreateAudioDeviceModule\n[04:10:17:INFO] [000:001] [771] (audio_device_module_ios.mm:44): current platform is IOS\n[04:10:18:INFO] [000:001] [771] (audio_device_module_ios.mm:45): iPhone Audio APIs will be utilized.\n[04:10:18:INFO] [000:012] [771] (audio_processing_impl.cc:438): Capture analyzer activated: 0\n[04:10:18:INFO] Capture post processor activated: 0\n[04:10:18:INFO] Render pre processor activated: 0\n[04:10:18:INFO] [000:014] [771] (webrtc_video_engine.cc:470): WebRtcVideoEngine::WebRtcVideoEngine()\n[04:10:18:INFO] [000:017] [771] (webrtc_voice_engine.cc:196): WebRtcVoiceEngine::WebRtcVoiceEngine\n[04:10:18:INFO] [000:022] [18711] (webrtc_voice_engine.cc:219): WebRtcVoiceEngine::Init\n[04:10:18:INFO] [000:022] [18711] (webrtc_voice_engine.cc:227): Supported send codecs in order of preference:\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:230): opus/48000/2 { minptime=10 useinbandfec=1 } (111)\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:230): ISAC/16000/1 (103)\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:230): ISAC/32000/1 (104)\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:230): G722/8000/1 (9)\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:230): ILBC/8000/1 (102)\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:230): PCMU/8000/1 (0)\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:230): PCMA/8000/1 (8)\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:230): CN/32000/1 (106)\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:230): CN/16000/1 (105)\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:230): CN/8000/1 (13)\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:230): telephone-event/48000/1 (110)\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:230): telephone-event/32000/1 (112)\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:230): telephone-event/16000/1 (113)\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:230): telephone-event/8000/1 (126)\n[04:10:18:INFO] [000:028] [18711] (webrtc_voice_engine.cc:233): Supported recv codecs in order of preference:\n[04:10:18:INFO] [000:029] [18711] (webrtc_voice_engine.cc:236): opus/48000/2 { minptime=10 useinbandfec=1 } (111)\n[04:10:18:INFO] [000:029] [18711] (webrtc_voice_engine.cc:236): ISAC/16000/1 (103)\n[04:10:18:INFO] [000:029] [18711] (webrtc_voice_engine.cc:236): ISAC/32000/1 (104)\n[04:10:18:INFO] [000:029] [18711] (webrtc_voice_engine.cc:236): G722/8000/1 (9)\n[04:10:18:INFO] [000:029] [18711] (webrtc_voice_engine.cc:236): ILBC/8000/1 (102)\n[04:10:18:INFO] [000:029] [18711] (webrtc_voice_engine.cc:236): PCMU/8000/1 (0)\n[04:10:18:INFO] [000:029] [18711] (webrtc_voice_engine.cc:236): PCMA/8000/1 (8)\n[04:10:18:INFO] [000:029] [18711] (webrtc_voice_engine.cc:236): CN/32000/1 (106)\n[04:10:18:INFO] [000:029] [18711] (webrtc_voice_engine.cc:236): CN/16000/1 (105)\n[04:10:18:INFO] [000:029] [18711] (webrtc_voice_engine.cc:236): CN/8000/1 (13)\n[04:10:18:INFO] [000:029] [18711] (webrtc_voice_engine.cc:236): telephone-event/48000/1 (110)\n[04:10:18:INFO] [000:029] [18711] (webrtc_voice_engine.cc:236): telephone-event/32000/1 (112)\n[04:10:18:INFO] [000:029] [18711] (webrtc_voice_engine.cc:236): telephone-event/16000/1 (113)\n[04:10:18:INFO] [000:029] [18711] (webrtc_voice_engine.cc:236): telephone-event/8000/1 (126)\n[04:10:18:INFO] [000:029] [18711] (audio_device_module_ios.mm:69): Init\n[04:10:18:INFO] [000:030] [18711] (audio_device_buffer.cc:63): AudioDeviceBuffer::ctor\n[04:10:18:INFO] [000:031] [18711] (audio_device_ios.mm:117): AudioDeviceIOS::ctor<NSThread: 0x604000874c00>{number = 3, name = (null)}\n[04:10:18:INFO] [000:031] [18711] (audio_device_module_ios.mm:49): AttachAudioBuffer\n[04:10:18:INFO] [000:031] [18711] (audio_device_ios.mm:133): AudioDeviceIOS::AttachAudioBuffer\n[04:10:18:INFO] [000:031] [18711] (audio_device_ios.mm:140): AudioDeviceIOS::Init\n[04:10:18:INFO] [000:031] [18711] (audio_device_ios.mm:85): LogDeviceInfo\n[04:10:18:INFO] [000:031] [18711] (audio_device_ios.mm:87): system name: iOS\n[04:10:18:INFO] [000:031] [18711] (audio_device_ios.mm:88): system version: 11.4\n[04:10:18:INFO] [000:031] [18711] (audio_device_ios.mm:89): device type: iPhone\n[04:10:18:INFO] [000:031] [18711] (audio_device_ios.mm:90): device name: x86_64\n[04:10:18:INFO] [000:032] [18711] (audio_device_ios.mm:91): process name: apprtcmobile_tests\n[04:10:18:INFO] [000:032] [18711] (audio_device_ios.mm:92): process ID: 7465\n[04:10:18:INFO] [000:032] [18711] (audio_device_ios.mm:93): OS version: Version 11.4 (Build 15F79)\n[04:10:18:INFO] [000:032] [18711] (audio_device_ios.mm:94): processing cores: 4\n[04:10:18:INFO] [000:032] [18711] (audio_device_ios.mm:95): low power mode: 0\n[04:10:18:INFO] [000:032] [18711] (audio_device_ios.mm:97): TARGET_IPHONE_SIMULATOR is defined\n[04:10:18:INFO] [000:032] [18711] (audio_device_ios.mm:99): DeviceIsSimulator: 1\n[04:10:18:INFO] [000:033] [18711] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer\n[04:10:18:INFO] [000:033] [18711] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000)\n[04:10:18:INFO] [000:033] [18711] (audio_device_buffer.cc:206): SetPlayoutChannels(1)\n[04:10:18:INFO] [000:033] [18711] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000)\n[04:10:18:INFO] [000:033] [18711] (audio_device_buffer.cc:200): SetRecordingChannels(1)\n[04:10:18:INFO] [000:034] [18711] (audio_device_module_ios.mm:406): SetPlayoutDevice(0)\n[04:10:18:INFO] [000:034] [18711] (audio_device_ios.mm:1006): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetPlayoutDevice(uint16_t): Not implemented\n[04:10:18:INFO] [000:034] [18711] (audio_device_module_ios.mm:108): InitSpeaker\n[04:10:18:INFO] [000:034] [18711] (audio_device_module_ios.mm:310): StereoPlayoutIsAvailable\n[04:10:18:INFO] [000:034] [18711] (audio_device_module_ios.mm:317): output: 0\n[04:10:18:INFO] [000:034] [18711] (audio_device_module_ios.mm:322): SetStereoPlayout(0)\n[04:10:18:INFO] [000:034] [18711] (audio_device_ios.mm:1059): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetStereoPlayout(bool): Not implemented\n[04:10:18:INFO] [000:034] [18711] (audio_device_module_ios.mm:330): stereo playout is not supported\n[04:10:18:INFO] [000:034] [18711] (adm_helpers.cc:57): Failed to set stereo playout mode.\n[04:10:18:INFO] [000:035] [18711] (audio_device_module_ios.mm:468): SetRecordingDevice(0)\n[04:10:18:INFO] [000:035] [18711] (audio_device_ios.mm:1108): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetRecordingDevice(uint16_t): Not implemented\n[04:10:18:INFO] [000:035] [18711] (audio_device_module_ios.mm:114): InitMicrophone\n[04:10:18:INFO] [000:035] [18711] (audio_device_module_ios.mm:277): StereoRecordingIsAvailable\n[04:10:18:INFO] [000:035] [18711] (audio_device_module_ios.mm:284): output: 0\n[04:10:18:INFO] [000:035] [18711] (audio_device_module_ios.mm:289): SetStereoRecording(0)\n[04:10:18:INFO] [000:035] [18711] (adm_helpers.cc:77): Failed to set stereo recording mode.\n[04:10:18:INFO] [000:035] [18711] (apm_helpers.cc:32): Setting AGC mode to 2\n[04:10:18:INFO] [000:035] [18711] (audio_processing_impl.cc:727): Highpass filter activated: 0\n[04:10:18:INFO] [000:035] [18711] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:035] [18711] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] [000:037] [18711] (audio_device_module_ios.mm:581): RegisterAudioCallback\n[04:10:18:INFO] [000:037] [18711] (audio_device_buffer.cc:81): RegisterAudioCallback\n[04:10:18:INFO] [000:037] [18711] (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[04:10:18:INFO] [000:037] [18711] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[04:10:18:INFO] [000:037] [18711] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[04:10:18:INFO] [000:037] [18711] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[04:10:18:INFO] [000:037] [18711] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[04:10:18:INFO] [000:037] [18711] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:037] [18711] (audio_device_module_ios.mm:600): output: 0\n[04:10:18:INFO] [000:037] [18711] (audio_processing_impl.cc:727): Highpass filter activated: 0\n[04:10:18:INFO] [000:037] [18711] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:037] [18711] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] [000:037] [18711] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[04:10:18:INFO] [000:037] [18711] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[04:10:18:INFO] [000:038] [18711] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:038] [18711] (audio_device_module_ios.mm:616): output: 0\n[04:10:18:INFO] [000:038] [18711] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[04:10:18:INFO] [000:038] [18711] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:038] [18711] (audio_device_module_ios.mm:632): output: 0\n[04:10:18:INFO] [000:038] [18711] (audio_processing_impl.cc:727): Highpass filter activated: 0\n[04:10:18:INFO] [000:038] [18711] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:038] [18711] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] [000:038] [18711] (apm_helpers.cc:62): NS set to 0\n[04:10:18:INFO] [000:038] [18711] (webrtc_voice_engine.cc:447): Stereo swapping enabled? 0\n[04:10:18:INFO] [000:038] [18711] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[04:10:18:INFO] [000:038] [18711] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[04:10:18:INFO] [000:038] [18711] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[04:10:18:INFO] [000:038] [18711] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[04:10:18:INFO] [000:038] [18711] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[04:10:18:INFO] [000:038] [18711] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[04:10:18:INFO] [000:038] [18711] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[04:10:18:INFO] [000:038] [18711] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[04:10:18:INFO] [000:038] [18711] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[04:10:18:INFO] [000:038] [18711] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:18:INFO] [000:039] [18711] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:039] [18711] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] (RTCLogging.mm:33): (ARDAppClient.m:270 -[ARDAppClient connectToRoomWithId:settings:isLoopback:]_block_invoke): Joined room:testRoom on room server.\n[04:10:18:INFO] [000:047] [771] (openssl_identity.cc:44): Making key pair\n[04:10:18:INFO] [000:206] [771] (openssl_identity.cc:92): Returning key pair\n[04:10:18:INFO] [000:207] [771] (openssl_certificate.cc:58): Making certificate for WebRTC\n[04:10:18:INFO] [000:216] [771] (openssl_certificate.cc:108): Returning certificate\n[04:10:18:INFO] [000:219] [771] (RTCCertificate.mm:63): CERT PEM\n[04:10:18:INFO] [000:219] [771] (RTCCertificate.mm:64): -----BEGIN CERTIFICATE-----\n[04:10:18:INFO] MIIBnjCCAQegAwIBAgIJALGoGtR3HJQMMA0GCSqGSIb3DQEBCwUAMBExDzANBgNV\n[04:10:18:INFO] BAMMBldlYlJUQzAeFw0xOTA0MTgyMzEwMThaFw0xOTA0MTkyMzExNThaMBExDzAN\n[04:10:18:INFO] BgNVBAMMBldlYlJUQzCBnzANBgkqhkiG9w0BAQEFAAOBjQAwgYkCgYEAvF9n1+cP\n[04:10:18:INFO] BByKnYGMGWQmGGt6fxjAOty14BXPOgkTArzqtrQehAidXEEZO5MHTLv5QktVzUT8\n[04:10:18:INFO] tv938jceLxCIBDSz4nG36tM5+msstH9s0EVVje1NXRscKV6PzrY11fEET82Ts+b7\n[04:10:18:INFO] dQFf6Q+DkvzVQ9tAxT9HJj/Wj26nhIKLB/ECAwEAATANBgkqhkiG9w0BAQsFAAOB\n[04:10:18:INFO] gQAqNbFHmzVc+MERuMx+fFBKmJg1UTcjN5qsNwYerHe2mEal6IsP/XWibVROFzDC\n[04:10:18:INFO] nXzqUqRgjZ9OoZaBsB+pUPAA0SJiW5esgNkXDgFGM0oNhxmahZlaLCPRREFWNzPk\n[04:10:18:INFO] 82QY37x+Lnov9PfRYVdFSjd3f8dgwlqUV5+kIG+nUVaqSA==\n[04:10:18:INFO] -----END CERTIFICATE-----\n[04:10:18:INFO] \n[04:10:18:INFO] [000:219] [771] (RTCConfiguration.mm:213): Have configured cert - using it.\n[04:10:18:INFO] [000:221] [771] (RTCConfiguration.mm:218): Created cert from PEM strings.\n[04:10:18:INFO] [000:225] [18711] (rtc_event_log_impl.cc:71): Creating legacy encoder for RTC event log.\n[04:10:18:INFO] [000:226] [18711] (peer_connection_factory.cc:407): Using default network controller factory\n[04:10:18:INFO] [000:229] [18711] (bitrate_prober.cc:66): Bandwidth probing enabled, set to inactive\n[04:10:18:INFO] [000:231] [18711] (paced_sender.cc:418): ProcessThreadAttached 0x35710ba0\n[04:10:18:INFO] [000:232] [18711] (cpu_info.cc:49): Available number of cores: 4\n[04:10:18:INFO] [000:235] [18711] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85\n[04:10:18:INFO] [000:235] [18711] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating.\n[04:10:18:INFO] [000:246] [39691] (webrtc_session_description_factory.cc:314): Using certificate supplied to the constructor.\n[04:10:18:INFO] [000:249] [39691] (peer_connection.cc:1405): Adding audio transceiver in response to a call to AddTrack.\n[04:10:18:INFO] (RTCLogging.mm:33): (ARDAppClient.m:396 -[ARDAppClient peerConnectionShouldNegotiate:]): WARNING: Renegotiation needed but unimplemented.\n[04:10:18:INFO] [000:253] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60400000aec0): created DTMF sender: RTCDtmfSender {\n[04:10:18:INFO] remainingTones:\n[04:10:18:INFO] duration: 0.100000 sec\n[04:10:18:INFO] interToneGap: 0.050000 sec\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:253] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60400022e840): created sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSa0\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:254] [771] (RTCLogging.mm:33): (RTCFileVideoCapturer.m:103 -[RTCFileVideoCapturer stopCapture]): File capturer stopped.\n[04:10:18:INFO] [000:254] [39691] (peer_connection.cc:1405): Adding video transceiver in response to a call to AddTrack.\n[04:10:18:INFO] [000:255] [39691] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[04:10:18:INFO] [000:255] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x6000002273e0): created sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSv0\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:255] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60400000aed0): created DTMF sender: RTCDtmfSender {\n[04:10:18:INFO] remainingTones:\n[04:10:18:INFO] duration: 0.100000 sec\n[04:10:18:INFO] interToneGap: 0.050000 sec\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:255] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60400022e880): created sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSa0\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:256] [771] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x604000056050): created receiver: RTCRtpReceiver {\n[04:10:18:INFO] receiverId: 77f96473-0b6f-4fd9-9c66-b60f97444634\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:256] [771] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x604000055fc0): created transceiver: RTCRtpTransceiver {\n[04:10:18:INFO] sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSa0\n[04:10:18:INFO] }\n[04:10:18:INFO] receiver: RTCRtpReceiver {\n[04:10:18:INFO] receiverId: 77f96473-0b6f-4fd9-9c66-b60f97444634\n[04:10:18:INFO] }\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:256] [39691] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[04:10:18:INFO] [000:256] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60400022e940): created sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSv0\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:256] [771] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x6040000560e0): created receiver: RTCRtpReceiver {\n[04:10:18:INFO] receiverId: 3efb1fbd-6f62-4b42-9491-83041d61cca3\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:256] [771] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x604000056080): created transceiver: RTCRtpTransceiver {\n[04:10:18:INFO] sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSv0\n[04:10:18:INFO] }\n[04:10:18:INFO] receiver: RTCRtpReceiver {\n[04:10:18:INFO] receiverId: 3efb1fbd-6f62-4b42-9491-83041d61cca3\n[04:10:18:INFO] }\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:260] [39691] (media_session.cc:335): Duplicate id found. Reassigning from 102 to 127\n[04:10:18:INFO] [000:260] [39691] (media_session.cc:335): Duplicate id found. Reassigning from 104 to 125\n[04:10:18:INFO] [000:260] [39691] (media_session.cc:335): Duplicate id found. Reassigning from 106 to 124\n[04:10:18:INFO] [000:260] [39691] (media_session.cc:335): Duplicate id found. Reassigning from 103 to 123\n[04:10:18:INFO] [000:260] [39691] (media_session.cc:335): Duplicate id found. Reassigning from 105 to 122\n[04:10:18:INFO] [000:261] [39691] (media_session.cc:335): Duplicate id found. Reassigning from 1 to 14\n[04:10:18:INFO] [000:261] [39691] (media_session.cc:335): Duplicate id found. Reassigning from 2 to 13\n[04:10:18:INFO] [000:261] [39691] (media_session.cc:335): Duplicate id found. Reassigning from 3 to 12\n[04:10:18:INFO] [000:261] [39691] (media_session.cc:335): Duplicate id found. Reassigning from 5 to 11\n[04:10:18:INFO] [000:266] [18711] (rtc_event_log_impl.cc:186): Starting WebRTC event log. (Timestamp, UTC) = (7457152683, 1555715418263901).\n[04:10:18:INFO] [000:267] [771] (audio_device_module.mm:21): CreateAudioDeviceModule\n[04:10:18:INFO] [000:267] [771] (audio_device_module_ios.mm:44): current platform is IOS\n[04:10:18:INFO] [000:267] [771] (audio_device_module_ios.mm:45): iPhone Audio APIs will be utilized.\n[04:10:18:INFO] [000:268] [771] (audio_processing_impl.cc:438): Capture analyzer activated: 0\n[04:10:18:INFO] Capture post processor activated: 0\n[04:10:18:INFO] Render pre processor activated: 0\n[04:10:18:INFO] [000:268] [771] (webrtc_video_engine.cc:470): WebRtcVideoEngine::WebRtcVideoEngine()\n[04:10:18:INFO] [000:268] [771] (webrtc_voice_engine.cc:196): WebRtcVoiceEngine::WebRtcVoiceEngine\n[04:10:18:INFO] [000:269] [41991] (webrtc_voice_engine.cc:219): WebRtcVoiceEngine::Init\n[04:10:18:INFO] [000:269] [41991] (webrtc_voice_engine.cc:227): Supported send codecs in order of preference:\n[04:10:18:INFO] [000:269] [41991] (webrtc_voice_engine.cc:230): opus/48000/2 { minptime=10 useinbandfec=1 } (111)\n[04:10:18:INFO] [000:269] [41991] (webrtc_voice_engine.cc:230): ISAC/16000/1 (103)\n[04:10:18:INFO] [000:269] [41991] (webrtc_voice_engine.cc:230): ISAC/32000/1 (104)\n[04:10:18:INFO] [000:269] [41991] (webrtc_voice_engine.cc:230): G722/8000/1 (9)\n[04:10:18:INFO] [000:269] [41991] (webrtc_voice_engine.cc:230): ILBC/8000/1 (102)\n[04:10:18:INFO] [000:269] [41991] (webrtc_voice_engine.cc:230): PCMU/8000/1 (0)\n[04:10:18:INFO] [000:269] [41991] (webrtc_voice_engine.cc:230): PCMA/8000/1 (8)\n[04:10:18:INFO] [000:269] [41991] (webrtc_voice_engine.cc:230): CN/32000/1 (106)\n[04:10:18:INFO] [000:269] [41991] (webrtc_voice_engine.cc:230): CN/16000/1 (105)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:230): CN/8000/1 (13)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:230): telephone-event/48000/1 (110)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:230): telephone-event/32000/1 (112)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:230): telephone-event/16000/1 (113)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:230): telephone-event/8000/1 (126)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:233): Supported recv codecs in order of preference:\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:236): opus/48000/2 { minptime=10 useinbandfec=1 } (111)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:236): ISAC/16000/1 (103)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:236): ISAC/32000/1 (104)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:236): G722/8000/1 (9)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:236): ILBC/8000/1 (102)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:236): PCMU/8000/1 (0)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:236): PCMA/8000/1 (8)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:236): CN/32000/1 (106)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:236): CN/16000/1 (105)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:236): CN/8000/1 (13)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:236): telephone-event/48000/1 (110)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:236): telephone-event/32000/1 (112)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:236): telephone-event/16000/1 (113)\n[04:10:18:INFO] [000:270] [41991] (webrtc_voice_engine.cc:236): telephone-event/8000/1 (126)\n[04:10:18:INFO] [000:270] [41991] (audio_device_module_ios.mm:69): Init\n[04:10:18:INFO] [000:270] [41991] (audio_device_buffer.cc:63): AudioDeviceBuffer::ctor\n[04:10:18:INFO] [000:270] [41991] (audio_device_ios.mm:117): AudioDeviceIOS::ctor<NSThread: 0x600001272a80>{number = 4, name = (null)}\n[04:10:18:INFO] [000:270] [41991] (audio_device_module_ios.mm:49): AttachAudioBuffer\n[04:10:18:INFO] [000:270] [41991] (audio_device_ios.mm:133): AudioDeviceIOS::AttachAudioBuffer\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:140): AudioDeviceIOS::Init\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:85): LogDeviceInfo\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:87): system name: iOS\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:88): system version: 11.4\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:89): device type: iPhone\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:90): device name: x86_64\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:91): process name: apprtcmobile_tests\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:92): process ID: 7465\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:93): OS version: Version 11.4 (Build 15F79)\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:94): processing cores: 4\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:95): low power mode: 0\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:97): TARGET_IPHONE_SIMULATOR is defined\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:99): DeviceIsSimulator: 1\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer\n[04:10:18:INFO] [000:271] [41991] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000)\n[04:10:18:INFO] [000:271] [41991] (audio_device_buffer.cc:206): SetPlayoutChannels(1)\n[04:10:18:INFO] [000:271] [41991] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000)\n[04:10:18:INFO] [000:271] [41991] (audio_device_buffer.cc:200): SetRecordingChannels(1)\n[04:10:18:INFO] [000:271] [41991] (audio_device_module_ios.mm:406): SetPlayoutDevice(0)\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:1006): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetPlayoutDevice(uint16_t): Not implemented\n[04:10:18:INFO] [000:271] [41991] (audio_device_module_ios.mm:108): InitSpeaker\n[04:10:18:INFO] [000:271] [41991] (audio_device_module_ios.mm:310): StereoPlayoutIsAvailable\n[04:10:18:INFO] [000:271] [41991] (audio_device_module_ios.mm:317): output: 0\n[04:10:18:INFO] [000:271] [41991] (audio_device_module_ios.mm:322): SetStereoPlayout(0)\n[04:10:18:INFO] [000:271] [41991] (audio_device_ios.mm:1059): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetStereoPlayout(bool): Not implemented\n[04:10:18:INFO] [000:271] [41991] (audio_device_module_ios.mm:330): stereo playout is not supported\n[04:10:18:INFO] [000:271] [41991] (adm_helpers.cc:57): Failed to set stereo playout mode.\n[04:10:18:INFO] [000:271] [41991] (audio_device_module_ios.mm:468): SetRecordingDevice(0)\n[04:10:18:INFO] [000:272] [41991] (audio_device_ios.mm:1108): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetRecordingDevice(uint16_t): Not implemented\n[04:10:18:INFO] [000:272] [41991] (audio_device_module_ios.mm:114): InitMicrophone\n[04:10:18:INFO] [000:272] [41991] (audio_device_module_ios.mm:277): StereoRecordingIsAvailable\n[04:10:18:INFO] [000:272] [41991] (audio_device_module_ios.mm:284): output: 0\n[04:10:18:INFO] [000:272] [41991] (audio_device_module_ios.mm:289): SetStereoRecording(0)\n[04:10:18:INFO] [000:272] [41991] (adm_helpers.cc:77): Failed to set stereo recording mode.\n[04:10:18:INFO] [000:272] [41991] (apm_helpers.cc:32): Setting AGC mode to 2\n[04:10:18:INFO] [000:272] [41991] (audio_processing_impl.cc:727): Highpass filter activated: 0\n[04:10:18:INFO] [000:272] [41991] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:272] [41991] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] [000:272] [41991] (audio_device_module_ios.mm:581): RegisterAudioCallback\n[04:10:18:INFO] [000:272] [41991] (audio_device_buffer.cc:81): RegisterAudioCallback\n[04:10:18:INFO] [000:272] [41991] (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[04:10:18:INFO] [000:272] [41991] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[04:10:18:INFO] [000:272] [41991] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[04:10:18:INFO] [000:272] [41991] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[04:10:18:INFO] [000:272] [41991] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[04:10:18:INFO] [000:272] [41991] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:273] [41991] (audio_device_module_ios.mm:600): output: 0\n[04:10:18:INFO] [000:273] [41991] (audio_processing_impl.cc:727): Highpass filter activated: 0\n[04:10:18:INFO] [000:273] [41991] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:273] [41991] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] [000:273] [41991] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[04:10:18:INFO] [000:273] [41991] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[04:10:18:INFO] [000:273] [41991] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:273] [41991] (audio_device_module_ios.mm:616): output: 0\n[04:10:18:INFO] [000:273] [41991] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[04:10:18:INFO] [000:273] [41991] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:273] [41991] (audio_device_module_ios.mm:632): output: 0\n[04:10:18:INFO] [000:273] [41991] (audio_processing_impl.cc:727): Highpass filter activated: 0\n[04:10:18:INFO] [000:274] [41991] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:274] [41991] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] [000:274] [41991] (apm_helpers.cc:62): NS set to 0\n[04:10:18:INFO] [000:274] [41991] (webrtc_voice_engine.cc:447): Stereo swapping enabled? 0\n[04:10:18:INFO] [000:274] [41991] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[04:10:18:INFO] [000:274] [41991] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[04:10:18:INFO] [000:274] [41991] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[04:10:18:INFO] [000:274] [41991] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[04:10:18:INFO] [000:274] [41991] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[04:10:18:INFO] [000:274] [41991] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[04:10:18:INFO] [000:274] [41991] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[04:10:18:INFO] [000:274] [41991] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[04:10:18:INFO] [000:274] [41991] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[04:10:18:INFO] [000:274] [41991] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:18:INFO] [000:274] [41991] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:274] [41991] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] (RTCLogging.mm:33): (ARDAppClient.m:270 -[ARDAppClient connectToRoomWithId:settings:isLoopback:]_block_invoke): Joined room:testRoom on room server.\n[04:10:18:INFO] [000:274] [771] (openssl_identity.cc:44): Making key pair\n[04:10:18:INFO] [000:481] [771] (openssl_identity.cc:92): Returning key pair\n[04:10:18:INFO] [000:481] [771] (openssl_certificate.cc:58): Making certificate for WebRTC\n[04:10:18:INFO] [000:484] [771] (openssl_certificate.cc:108): Returning certificate\n[04:10:18:INFO] [000:484] [771] (RTCCertificate.mm:63): CERT PEM\n[04:10:18:INFO] [000:484] [771] (RTCCertificate.mm:64): -----BEGIN CERTIFICATE-----\n[04:10:18:INFO] MIIBnjCCAQegAwIBAgIJAM60FluKv1UXMA0GCSqGSIb3DQEBCwUAMBExDzANBgNV\n[04:10:18:INFO] BAMMBldlYlJUQzAeFw0xOTA0MTgyMzEwMThaFw0xOTA0MTkyMzExNThaMBExDzAN\n[04:10:18:INFO] BgNVBAMMBldlYlJUQzCBnzANBgkqhkiG9w0BAQEFAAOBjQAwgYkCgYEAzI1W10uB\n[04:10:18:INFO] Mu/GHK9IvHwf5afJQSvXI7xBrN1JX4OEDVpm2QzfcTTqL7H35h4AZYRAAeV+KYHu\n[04:10:18:INFO] wIoGmKutHQeEyCEBrF5MMlZsbGx/wO9AjltuI3uxRhywfHrVaYWsM/zA4SDh+6zW\n[04:10:18:INFO] 7ZgABLiZOYGFdgZWNJsiNhpiBmfzXjX3t7ECAwEAATANBgkqhkiG9w0BAQsFAAOB\n[04:10:18:INFO] gQA7rvdIqwtdJpPJE5Px6snpMd46o1SrkyW515hDU3KqtuDjm9gIhFdhrbkZuh70\n[04:10:18:INFO] WWpBDQ37WZKG9rA5jz1qe4ZAOUYliSpKl51J+VR2PMGJQ3k8+LL2dTJOErtZ8ifP\n[04:10:18:INFO] vUnLKnpfYSW5fxeCs2QkQUmIuX+4Zg/jFM/VN++IPuZOmg==\n[04:10:18:INFO] -----END CERTIFICATE-----\n[04:10:18:INFO] \n[04:10:18:INFO] [000:485] [771] (RTCConfiguration.mm:213): Have configured cert - using it.\n[04:10:18:INFO] [000:486] [771] (RTCConfiguration.mm:218): Created cert from PEM strings.\n[04:10:18:INFO] [000:487] [41991] (rtc_event_log_impl.cc:71): Creating legacy encoder for RTC event log.\n[04:10:18:INFO] [000:487] [41991] (peer_connection_factory.cc:407): Using default network controller factory\n[04:10:18:INFO] [000:487] [41991] (bitrate_prober.cc:66): Bandwidth probing enabled, set to inactive\n[04:10:18:INFO] [000:487] [41991] (paced_sender.cc:418): ProcessThreadAttached 0x35440fa0\n[04:10:18:INFO] [000:487] [41991] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85\n[04:10:18:INFO] [000:487] [41991] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating.\n[04:10:18:INFO] [000:488] [16139] (webrtc_session_description_factory.cc:314): Using certificate supplied to the constructor.\n[04:10:18:INFO] [000:489] [16139] (peer_connection.cc:1405): Adding audio transceiver in response to a call to AddTrack.\n[04:10:18:INFO] (RTCLogging.mm:33): (ARDAppClient.m:396 -[ARDAppClient peerConnectionShouldNegotiate:]): WARNING: Renegotiation needed but unimplemented.\n[04:10:18:INFO] [000:489] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60400000b040): created DTMF sender: RTCDtmfSender {\n[04:10:18:INFO] remainingTones:\n[04:10:18:INFO] duration: 0.100000 sec\n[04:10:18:INFO] interToneGap: 0.050000 sec\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:489] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60400022f160): created sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSa0\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:489] [771] (RTCLogging.mm:33): (RTCFileVideoCapturer.m:103 -[RTCFileVideoCapturer stopCapture]): File capturer stopped.\n[04:10:18:INFO] [000:489] [16139] (peer_connection.cc:1405): Adding video transceiver in response to a call to AddTrack.\n[04:10:18:INFO] [000:490] [16139] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[04:10:18:INFO] [000:490] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x6000002292e0): created sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSv0\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:490] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60400000b0e0): created DTMF sender: RTCDtmfSender {\n[04:10:18:INFO] remainingTones:\n[04:10:18:INFO] duration: 0.100000 sec\n[04:10:18:INFO] interToneGap: 0.050000 sec\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:490] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600000229580): created sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSa0\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:490] [771] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x604000056aa0): created receiver: RTCRtpReceiver {\n[04:10:18:INFO] receiverId: 5d7d0dc4-08fe-486e-b754-ab0bee9890ce\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:490] [771] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x60000004e9d0): created transceiver: RTCRtpTransceiver {\n[04:10:18:INFO] sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSa0\n[04:10:18:INFO] }\n[04:10:18:INFO] receiver: RTCRtpReceiver {\n[04:10:18:INFO] receiverId: 5d7d0dc4-08fe-486e-b754-ab0bee9890ce\n[04:10:18:INFO] }\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:490] [16139] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[04:10:18:INFO] [000:491] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x6000002295a0): created sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSv0\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:491] [771] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x60000004ea60): created receiver: RTCRtpReceiver {\n[04:10:18:INFO] receiverId: 532e2aea-fa78-454d-b22d-4f061c3bb6b6\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:491] [771] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x60000004ea00): created transceiver: RTCRtpTransceiver {\n[04:10:18:INFO] sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSv0\n[04:10:18:INFO] }\n[04:10:18:INFO] receiver: RTCRtpReceiver {\n[04:10:18:INFO] receiverId: 532e2aea-fa78-454d-b22d-4f061c3bb6b6\n[04:10:18:INFO] }\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:491] [41991] (rtc_event_log_impl.cc:186): Starting WebRTC event log. (Timestamp, UTC) = (7457377766, 1555715418488990).\n[04:10:18:INFO] [000:493] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[04:10:18:INFO] [000:495] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[04:10:18:INFO] [000:501] [16403] (p2p_transport_channel.cc:501): Set backup connection ping interval to 25000 milliseconds.\n[04:10:18:INFO] [000:501] [16403] (p2p_transport_channel.cc:510): Set ICE receiving timeout to 2500 milliseconds\n[04:10:18:INFO] [000:501] [16403] (p2p_transport_channel.cc:517): Set ping most likely connection to 0\n[04:10:18:INFO] [000:501] [16403] (p2p_transport_channel.cc:524): Set stable_writable_connection_ping_interval to 2500\n[04:10:18:INFO] [000:501] [16403] (p2p_transport_channel.cc:537): Set presume writable when fully relayed to 0\n[04:10:18:INFO] [000:501] [16403] (p2p_transport_channel.cc:546): Set regather_on_failed_networks_interval to 300000\n[04:10:18:INFO] [000:501] [16403] (p2p_transport_channel.cc:565): Set receiving_switching_delay to 1000\n[04:10:18:INFO] [000:502] [16403] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0\n[04:10:18:INFO] [000:502] [16403] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 3563c5b0\n[04:10:18:INFO] [000:502] [16403] (p2p_transport_channel.cc:501): Set backup connection ping interval to 25000 milliseconds.\n[04:10:18:INFO] [000:502] [16403] (p2p_transport_channel.cc:510): Set ICE receiving timeout to 2500 milliseconds\n[04:10:18:INFO] [000:502] [16403] (p2p_transport_channel.cc:517): Set ping most likely connection to 0\n[04:10:18:INFO] [000:503] [16403] (p2p_transport_channel.cc:524): Set stable_writable_connection_ping_interval to 2500\n[04:10:18:INFO] [000:503] [16403] (p2p_transport_channel.cc:537): Set presume writable when fully relayed to 0\n[04:10:18:INFO] [000:503] [16403] (p2p_transport_channel.cc:546): Set regather_on_failed_networks_interval to 300000\n[04:10:18:INFO] [000:503] [16403] (p2p_transport_channel.cc:565): Set receiving_switching_delay to 1000\n[04:10:18:INFO] [000:503] [16403] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 1 transport 0\n[04:10:18:INFO] [000:503] [16403] (dtls_srtp_transport.cc:66): Setting RTP Transport on 1 transport 3563cd60\n[04:10:18:INFO] [000:503] [16403] (p2p_transport_channel.cc:438): Set ICE ufrag: qFn3 pwd: YzaQUb5kFo04/Vz8vnZxMqt5 on transport 0\n[04:10:18:INFO] [000:503] [16403] (p2p_transport_channel.cc:438): Set ICE ufrag: qFn3 pwd: YzaQUb5kFo04/Vz8vnZxMqt5 on transport 1\n[04:10:18:INFO] [000:504] [18711] (webrtc_voice_engine.cc:1485): 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[04:10:18:INFO] [000:504] [18711] (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[04:10:18:INFO] [000:504] [18711] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[04:10:18:INFO] [000:504] [18711] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[04:10:18:INFO] [000:504] [18711] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[04:10:18:INFO] [000:504] [18711] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[04:10:18:INFO] [000:504] [18711] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:504] [18711] (audio_device_module_ios.mm:600): output: 0\n[04:10:18:INFO] [000:504] [18711] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:18:INFO] [000:504] [18711] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:504] [18711] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] [000:504] [18711] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[04:10:18:INFO] [000:504] [18711] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[04:10:18:INFO] [000:504] [18711] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:504] [18711] (audio_device_module_ios.mm:616): output: 0\n[04:10:18:INFO] [000:504] [18711] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[04:10:18:INFO] [000:504] [18711] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:504] [18711] (audio_device_module_ios.mm:632): output: 0\n[04:10:18:INFO] [000:504] [18711] (apm_helpers.cc:62): NS set to 0\n[04:10:18:INFO] [000:504] [18711] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[04:10:18:INFO] [000:504] [18711] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[04:10:18:INFO] [000:504] [18711] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[04:10:18:INFO] [000:504] [18711] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[04:10:18:INFO] [000:505] [18711] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[04:10:18:INFO] [000:505] [18711] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[04:10:18:INFO] [000:505] [18711] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[04:10:18:INFO] [000:505] [18711] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[04:10:18:INFO] [000:505] [18711] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[04:10:18:INFO] [000:505] [18711] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:18:INFO] [000:505] [18711] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:505] [18711] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] [000:505] [18711] (webrtc_voice_engine.cc:1503): 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[04:10:18:INFO] [000:505] [18711] (channel.cc:144): Created channel for 0\n[04:10:18:INFO] [000:507] [18711] (channel.cc:212): BaseChannel::Init_w, media_transport=0\n[04:10:18:INFO] [000:507] [18711] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:18:INFO] [000:507] [18711] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:18:INFO] [000:508] [18711] (webrtc_video_engine.cc:483): CreateMediaChannel. Options: VideoOptions {}\n[04:10:18:INFO] [000:508] [18711] (channel.cc:144): Created channel for 1\n[04:10:18:INFO] [000:509] [18711] (channel.cc:212): BaseChannel::Init_w, media_transport=0\n[04:10:18:INFO] [000:509] [18711] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:18:INFO] [000:509] [18711] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:18:INFO] [000:509] [39691] (peer_connection.cc:4173): Session: 2477746311721103795 Old state: kStable New state: kHaveLocalOffer\n[04:10:18:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 1\n[04:10:18:INFO] [000:509] [18711] (channel.cc:871): Setting local voice description\n[04:10:18:INFO] [000:510] [18711] (webrtc_voice_engine.cc:1324): 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[04:10:18:INFO] [000:510] [18711] (webrtc_voice_engine.cc:1513): Setting receive voice codecs.\n[04:10:18:INFO] [000:511] [18711] (webrtc_voice_engine.cc:1784): AddSendStream: {id:ARDAMSa0;ssrcs:[2982798033];ssrc_groups:;cname:S0NY9KGLI1aoJqMd;stream_ids:ARDAMS;}\n[04:10:18:INFO] [000:515] [18711] (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[04:10:18:INFO] [000:516] [18711] (audio_coding_module.cc:333): Created\n[04:10:18:INFO] [000:519] [18711] (channel_send.cc:708): Not setting media_transport_ rate observers.\n[04:10:18:INFO] [000:519] [18711] (audio_send_stream.cc:142): AudioSendStream: 2982798033\n[04:10:18:INFO] [000:519] [18711] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2982798033, extmap-allow-mixed: false, extensions: [], c_name: S0NY9KGLI1aoJqMd}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: <unset>}\n[04:10:18:INFO] [000:520] [18711] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:18:INFO] [000:520] [18711] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:18:INFO] [000:521] [18711] (channel.cc:676): Add send stream ssrc: 2982798033\n[04:10:18:INFO] [000:521] [18711] (channel.cc:863): Changing voice state, recv=0 send=0\n[04:10:18:INFO] [000:521] [18711] (channel.cc:1009): Setting local video description\n[04:10:18:INFO] [000:521] [18711] (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[04:10:18:INFO] [000:522] [18711] (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[04:10:18:INFO] [000:523] [18711] (webrtc_video_engine.cc:1072): AddSendStream: {id:ARDAMSv0;ssrcs:[97654628,3853731424];ssrc_groups:{semantics:FID;ssrcs:[97654628,3853731424]};cname:S0NY9KGLI1aoJqMd;stream_ids:ARDAMS;}\n[04:10:18:INFO] [000:523] [18711] (webrtc_video_engine.cc:1111): SetLocalSsrc on all the receive streams because we added a send stream.\n[04:10:18:INFO] [000:523] [18711] (channel.cc:676): Add send stream ssrc: 97654628\n[04:10:18:INFO] [000:523] [18711] (channel.cc:996): Changing video state, send=0\n[04:10:18:INFO] [000:523] [39691] (peer_connection.cc:5512): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport.\n[04:10:18:INFO] [000:523] [18711] (webrtc_voice_engine.cc:1485): Setting voice channel options: AudioOptions {}\n[04:10:18:INFO] [000:523] [18711] (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[04:10:18:INFO] [000:523] [18711] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[04:10:18:INFO] [000:524] [18711] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[04:10:18:INFO] [000:524] [18711] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[04:10:18:INFO] [000:524] [18711] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[04:10:18:INFO] [000:524] [18711] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:524] [18711] (audio_device_module_ios.mm:600): output: 0\n[04:10:18:INFO] [000:524] [18711] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:18:INFO] [000:524] [18711] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:524] [18711] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] [000:524] [18711] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[04:10:18:INFO] [000:524] [18711] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[04:10:18:INFO] [000:524] [18711] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:525] [18711] (audio_device_module_ios.mm:616): output: 0\n[04:10:18:INFO] [000:525] [18711] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[04:10:18:INFO] [000:525] [18711] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:525] [18711] (audio_device_module_ios.mm:632): output: 0\n[04:10:18:INFO] [000:525] [18711] (apm_helpers.cc:62): NS set to 0\n[04:10:18:INFO] [000:525] [18711] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[04:10:18:INFO] [000:525] [18711] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[04:10:18:INFO] [000:525] [18711] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[04:10:18:INFO] [000:525] [18711] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[04:10:18:INFO] [000:525] [18711] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[04:10:18:INFO] [000:525] [18711] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[04:10:18:INFO] [000:525] [18711] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[04:10:18:INFO] [000:525] [18711] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[04:10:18:INFO] [000:525] [18711] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[04:10:18:INFO] [000:525] [18711] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:18:INFO] [000:525] [18711] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:525] [18711] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] [000:525] [18711] (webrtc_voice_engine.cc:1503): 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[04:10:18:INFO] [000:525] [18711] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 97654628, options: VideoOptions {noise reduction: false, is_screencast : false, }, source = (source))\n[04:10:18:INFO] [000:526] [16403] (basic_port_allocator.cc:405): Start getting ports with prune_turn_ports disabled\n[04:10:18:INFO] [000:526] [16403] (basic_port_allocator.cc:405): Start getting ports with prune_turn_ports disabled\n[04:10:18:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 1\n[04:10:18:INFO] [000:527] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[04:10:18:INFO] [000:527] [16403] (basic_port_allocator.cc:109): Filtered out ignored networks:\n[04:10:18:INFO] [000:527] [16403] (basic_port_allocator.cc:111): Net[lo0:::1/128:Loopback:id=3]\n[04:10:18:INFO] [000:527] [16403] (basic_port_allocator.cc:111): Net[lo0:127.0.0.0/8:Loopback:id=2]\n[04:10:18:INFO] [000:528] [16403] (basic_port_allocator.cc:918): Network manager has started\n[04:10:18:INFO] [000:528] [16403] (basic_port_allocator.cc:109): Filtered out ignored networks:\n[04:10:18:INFO] [000:528] [16403] (basic_port_allocator.cc:111): Net[lo0:::1/128:Loopback:id=3]\n[04:10:18:INFO] [000:528] [16403] (basic_port_allocator.cc:111): Net[lo0:127.0.0.0/8:Loopback:id=2]\n[04:10:18:INFO] [000:528] [16403] (basic_port_allocator.cc:918): Network manager has started\n[04:10:18:INFO] [000:528] [16403] (basic_port_allocator.cc:109): Filtered out ignored networks:\n[04:10:18:INFO] [000:528] [16403] (basic_port_allocator.cc:111): Net[lo0:::1/128:Loopback:id=3]\n[04:10:18:INFO] [000:528] [16403] (basic_port_allocator.cc:111): Net[lo0:127.0.0.0/8:Loopback:id=2]\n[04:10:18:INFO] [000:528] [16403] (basic_port_allocator.cc:833): Allocate ports on 1 networks\n[04:10:18:INFO] [000:528] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[04:10:18:INFO] [000:528] [16403] (basic_port_allocator.cc:109): Filtered out ignored networks:\n[04:10:18:INFO] [000:528] [16403] (basic_port_allocator.cc:111): Net[lo0:::1/128:Loopback:id=3]\n[04:10:18:INFO] [000:528] [16403] (basic_port_allocator.cc:111): Net[lo0:127.0.0.0/8:Loopback:id=2]\n[04:10:18:INFO] [000:528] [16403] (basic_port_allocator.cc:833): Allocate ports on 1 networks\n[04:10:18:INFO] [000:529] [16403] (basic_port_allocator.cc:1372): Net[en0:192.168.72.0/22:Wifi:id=1]: Allocation Phase=Udp\n[04:10:18:INFO] [000:529] [16403] (port.cc:320): Port[3704b800::1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port created with network cost 10\n[04:10:18:INFO] [000:529] [16403] (basic_port_allocator.cc:942): Adding allocated port for 0\n[04:10:18:INFO] [000:530] [16403] (basic_port_allocator.cc:961): Port[3704b800:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Added port to allocator\n[04:10:18:INFO] [000:530] [16403] (basic_port_allocator.cc:979): Port[3704b800:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Gathered candidate: Cand[:3223547122:1:udp:2122260223:192.168.72.27:51820:local::0:qFn3:YzaQUb5kFo04/Vz8vnZxMqt5:1:10:0]\n[04:10:18:INFO] [000:531] [23047] (p2p_transport_channel.cc:501): Set backup connection ping interval to 25000 milliseconds.\n[04:10:18:INFO] [000:531] [23047] (p2p_transport_channel.cc:510): Set ICE receiving timeout to 2500 milliseconds\n[04:10:18:INFO] [000:531] [23047] (p2p_transport_channel.cc:517): Set ping most likely connection to 0\n[04:10:18:INFO] [000:531] [23047] (p2p_transport_channel.cc:524): Set stable_writable_connection_ping_interval to 2500\n[04:10:18:INFO] [000:531] [16403] (basic_port_allocator.cc:1007): Port[3704b800:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port ready.\n[04:10:18:INFO] [000:531] [23047] (p2p_transport_channel.cc:537): Set presume writable when fully relayed to 0\n[04:10:18:INFO] [000:531] [16403] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported.\n[04:10:18:INFO] [000:531] [23047] (p2p_transport_channel.cc:546): Set regather_on_failed_networks_interval to 300000\n[04:10:18:INFO] [000:531] [23047] (p2p_transport_channel.cc:565): Set receiving_switching_delay to 1000\n[04:10:18:INFO] [000:531] [16403] (p2p_transport_channel.cc:805): Port[3704b800:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0\n[04:10:18:INFO] [000:531] [23047] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0\n[04:10:18:INFO] [000:532] [23047] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 35442ec0\n[04:10:18:INFO] [000:532] [23047] (p2p_transport_channel.cc:501): Set backup connection ping interval to 25000 milliseconds.\n[04:10:18:INFO] [000:532] [23047] (p2p_transport_channel.cc:510): Set ICE receiving timeout to 2500 milliseconds\n[04:10:18:INFO] [000:532] [23047] (p2p_transport_channel.cc:517): Set ping most likely connection to 0\n[04:10:18:INFO] [000:532] [23047] (p2p_transport_channel.cc:524): Set stable_writable_connection_ping_interval to 2500\n[04:10:18:INFO] [000:532] [23047] (p2p_transport_channel.cc:537): Set presume writable when fully relayed to 0\n[04:10:18:INFO] [000:532] [23047] (p2p_transport_channel.cc:546): Set regather_on_failed_networks_interval to 300000\n[04:10:18:INFO] [000:532] [23047] (p2p_transport_channel.cc:565): Set receiving_switching_delay to 1000\n[04:10:18:INFO] [000:532] [23047] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 1 transport 0\n[04:10:18:INFO] [000:532] [23047] (dtls_srtp_transport.cc:66): Setting RTP Transport on 1 transport 3571c5b0\n[04:10:18:INFO] [000:532] [16403] (basic_port_allocator.cc:1084): Port[3704b800:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port completed gathering candidates.\n[04:10:18:INFO] [000:533] [16403] (basic_port_allocator.cc:1372): Net[en0:192.168.72.0/22:Wifi:id=1]: Allocation Phase=Udp\n[04:10:18:INFO] [000:533] [23047] (p2p_transport_channel.cc:449): Received remote ICE parameters: ufrag=qFn3, renomination enabled\n[04:10:18:INFO] [000:533] [16403] (port.cc:320): Port[37085800::1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port created with network cost 10\n[04:10:18:INFO] [000:533] [16403] (basic_port_allocator.cc:942): Adding allocated port for 1\n[04:10:18:INFO] [000:533] [23047] (p2p_transport_channel.cc:449): Received remote ICE parameters: ufrag=qFn3, renomination enabled\n[04:10:18:INFO] [000:533] [16403] (basic_port_allocator.cc:961): Port[37085800:1:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Added port to allocator\n[04:10:18:INFO] [000:533] [16403] (basic_port_allocator.cc:979): Port[37085800:1:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Gathered candidate: Cand[:3223547122:1:udp:2122260223:192.168.72.27:57750:local::0:qFn3:YzaQUb5kFo04/Vz8vnZxMqt5:1:10:0]\n[04:10:18:INFO] [000:533] [41991] (webrtc_voice_engine.cc:1485): 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[04:10:18:INFO] [000:534] [41991] (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[04:10:18:INFO] [000:534] [16403] (basic_port_allocator.cc:1007): Port[37085800:1:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port ready.\n[04:10:18:INFO] [000:534] [41991] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[04:10:18:INFO] [000:534] [41991] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[04:10:18:INFO] [000:534] [16403] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported.\n[04:10:18:INFO] [000:534] [41991] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[04:10:18:INFO] [000:534] [16403] (p2p_transport_channel.cc:805): Port[37085800:1:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0\n[04:10:18:INFO] [000:534] [41991] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[04:10:18:INFO] [000:534] [41991] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:534] [41991] (audio_device_module_ios.mm:600): output: 0\n[04:10:18:INFO] [000:534] [41991] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:18:INFO] [000:534] [16403] (basic_port_allocator.cc:1084): Port[37085800:1:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port completed gathering candidates.\n[04:10:18:INFO] [000:534] [41991] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:534] [41991] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] [000:534] [41991] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[04:10:18:INFO] [000:534] [41991] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[04:10:18:INFO] [000:534] [41991] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:534] [41991] (audio_device_module_ios.mm:616): output: 0\n[04:10:18:INFO] [000:534] [41991] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[04:10:18:INFO] [000:534] [41991] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:534] [41991] (audio_device_module_ios.mm:632): output: 0\n[04:10:18:INFO] [000:534] [41991] (apm_helpers.cc:62): NS set to 0\n[04:10:18:INFO] [000:535] [41991] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[04:10:18:INFO] [000:535] [41991] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[04:10:18:INFO] [000:535] [41991] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[04:10:18:INFO] [000:535] [41991] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[04:10:18:INFO] [000:535] [41991] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[04:10:18:INFO] [000:535] [41991] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[04:10:18:INFO] [000:535] [41991] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[04:10:18:INFO] [000:535] [41991] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[04:10:18:INFO] [000:535] [41991] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[04:10:18:INFO] [000:535] [41991] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:18:INFO] [000:535] [41991] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:535] [41991] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] [000:535] [41991] (webrtc_voice_engine.cc:1503): 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[04:10:18:INFO] [000:535] [41991] (channel.cc:144): Created channel for 0\n[04:10:18:INFO] [000:536] [41991] (channel.cc:212): BaseChannel::Init_w, media_transport=0\n[04:10:18:INFO] [000:536] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:18:INFO] [000:536] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:18:INFO] [000:536] [41991] (webrtc_video_engine.cc:483): CreateMediaChannel. Options: VideoOptions {}\n[04:10:18:INFO] [000:536] [41991] (channel.cc:144): Created channel for 1\n[04:10:18:INFO] [000:537] [41991] (channel.cc:212): BaseChannel::Init_w, media_transport=0\n[04:10:18:INFO] [000:537] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:18:INFO] [000:537] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:18:INFO] [000:537] [16139] (peer_connection.cc:4173): Session: 3921930938603699854 Old state: kStable New state: kHaveRemoteOffer\n[04:10:18:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 3\n[04:10:18:INFO] [000:537] [41991] (channel.cc:923): Setting remote voice description\n[04:10:18:INFO] [000:537] [41991] (webrtc_voice_engine.cc:1279): 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[04:10:18:INFO] [000:537] [41991] (webrtc_voice_engine.cc:1702): Recreate all the receive streams because the send codec has changed.\n[04:10:18:INFO] [000:537] [41991] (webrtc_voice_engine.cc:2163): WebRtcVoiceMediaChannel::SetMaxSendBitrate.\n[04:10:18:INFO] [000:537] [41991] (webrtc_voice_engine.cc:1485): Setting voice channel options: AudioOptions {}\n[04:10:18:INFO] [000:537] [41991] (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[04:10:18:INFO] [000:538] [41991] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[04:10:18:INFO] [000:538] [41991] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[04:10:18:INFO] [000:538] [41991] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[04:10:18:INFO] [000:538] [41991] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[04:10:18:INFO] [000:538] [41991] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:538] [41991] (audio_device_module_ios.mm:600): output: 0\n[04:10:18:INFO] [000:538] [41991] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:18:INFO] [000:538] [41991] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:538] [41991] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] [000:538] [41991] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[04:10:18:INFO] [000:538] [41991] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[04:10:18:INFO] [000:538] [41991] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:538] [41991] (audio_device_module_ios.mm:616): output: 0\n[04:10:18:INFO] [000:538] [41991] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[04:10:18:INFO] [000:538] [41991] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[04:10:18:INFO] [000:538] [41991] (audio_device_module_ios.mm:632): output: 0\n[04:10:18:INFO] [000:538] [41991] (apm_helpers.cc:62): NS set to 0\n[04:10:18:INFO] [000:538] [41991] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[04:10:18:INFO] [000:538] [41991] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[04:10:18:INFO] [000:538] [41991] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[04:10:18:INFO] [000:538] [41991] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[04:10:18:INFO] [000:538] [41991] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[04:10:18:INFO] [000:538] [41991] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[04:10:18:INFO] [000:538] [41991] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[04:10:18:INFO] [000:538] [41991] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[04:10:18:INFO] [000:538] [41991] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[04:10:18:INFO] [000:538] [41991] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:18:INFO] [000:539] [41991] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:18:INFO] [000:539] [41991] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:18:INFO] [000:539] [41991] (webrtc_voice_engine.cc:1503): 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[04:10:18:INFO] [000:539] [41991] (webrtc_voice_engine.cc:1849): AddRecvStream: {id:ARDAMSa0;ssrcs:[2982798033];ssrc_groups:;cname:S0NY9KGLI1aoJqMd;stream_ids:ARDAMS;}\n[04:10:18:INFO] [000:540] [41991] (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[04:10:18:INFO] [000:540] [41991] (audio_coding_module.cc:333): Created\n[04:10:18:INFO] [000:540] [41991] (audio_receive_stream.cc:117): AudioReceiveStream: 2982798033\n[04:10:18:INFO] [000:540] [41991] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2982798033, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS}\n[04:10:18:INFO] [000:540] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:18:INFO] [000:540] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:18:INFO] [000:541] [41991] (channel.cc:719): Add remote ssrc: 2982798033\n[04:10:18:INFO] [000:541] [41991] (channel.cc:863): Changing voice state, recv=0 send=0\n[04:10:18:INFO] [000:541] [41991] (channel.cc:1061): Setting remote video description\n[04:10:18:INFO] [000:541] [41991] (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[04:10:18:INFO] [000:542] [41991] (webrtc_video_engine.cc:690): Using codec: VideoCodec[96:H264]\n[04:10:18:INFO] [000:542] [41991] (webrtc_video_engine.cc:758): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed.\n[04:10:18:INFO] [000:542] [41991] (webrtc_video_engine.cc:1174): AddRecvStream: {id:ARDAMSv0;ssrcs:[97654628,3853731424];ssrc_groups:{semantics:FID;ssrcs:[97654628,3853731424]};cname:S0NY9KGLI1aoJqMd;stream_ids:ARDAMS;}\n[04:10:18:INFO] [000:548] [41991] (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: 97654628, 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: 3853731424, 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[04:10:18:INFO] [000:549] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:18:INFO] [000:549] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:18:INFO] [000:552] [41991] (channel.cc:719): Add remote ssrc: 97654628\n[04:10:18:INFO] [000:552] [41991] (channel.cc:996): Changing video state, send=0\n[04:10:18:INFO] [000:552] [16139] (peer_connection.cc:5512): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport.\n[04:10:18:INFO] [000:552] [16139] (peer_connection.cc:2745): Processing the MSIDs for MID=0 (streams=[ARDAMS]).\n[04:10:18:INFO] [000:552] [16139] (peer_connection.cc:2756): Processing the addition of a remote track for MID=0.\n[04:10:18:INFO] [000:553] [41991] (webrtc_voice_engine.cc:1962): SetOutputVolume() to 1 for recv stream with ssrc 2982798033\n[04:10:18:INFO] [000:553] [16139] (peer_connection.cc:2745): Processing the MSIDs for MID=1 (streams=[ARDAMS]).\n[04:10:18:INFO] [000:553] [16139] (peer_connection.cc:2756): Processing the addition of a remote track for MID=1.\n[04:10:18:INFO] [000:553] [41991] (webrtc_video_engine.cc:1304): SetSink: ssrc:97654628 (ptr)\n[04:10:18:INFO] [000:553] [16139] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60400000b680): created DTMF sender: RTCDtmfSender {\n[04:10:18:INFO] remainingTones:\n[04:10:18:INFO] duration: 0.100000 sec\n[04:10:18:INFO] interToneGap: 0.050000 sec\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:553] [16139] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60400022faa0): created sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSa0\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:553] [16139] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x60400005aa30): created receiver: RTCRtpReceiver {\n[04:10:18:INFO] receiverId: 5d7d0dc4-08fe-486e-b754-ab0bee9890ce\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:553] [16139] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x604000059800): created transceiver: RTCRtpTransceiver {\n[04:10:18:INFO] sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSa0\n[04:10:18:INFO] }\n[04:10:18:INFO] receiver: RTCRtpReceiver {\n[04:10:18:INFO] receiverId: 5d7d0dc4-08fe-486e-b754-ab0bee9890ce\n[04:10:18:INFO] }\n[04:10:18:INFO] }\n[04:10:18:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving audio on track 5d7d0dc4-08fe-486e-b754-ab0bee9890ce.\n[04:10:18:INFO] [000:554] [16139] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[04:10:18:INFO] [000:554] [16139] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x604000230160): created sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSv0\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:554] [16139] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x60400005abe0): created receiver: RTCRtpReceiver {\n[04:10:18:INFO] receiverId: 532e2aea-fa78-454d-b22d-4f061c3bb6b6\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:554] [16139] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x604000059800): created transceiver: RTCRtpTransceiver {\n[04:10:18:INFO] sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSv0\n[04:10:18:INFO] }\n[04:10:18:INFO] receiver: RTCRtpReceiver {\n[04:10:18:INFO] receiverId: 532e2aea-fa78-454d-b22d-4f061c3bb6b6\n[04:10:18:INFO] }\n[04:10:18:INFO] }\n[04:10:18:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving video on track 532e2aea-fa78-454d-b22d-4f061c3bb6b6.\n[04:10:18:INFO] (RTCLogging.mm:33): (ARDAppClient.m:381 -[ARDAppClient peerConnection:didAddStream:]): Stream with 1 video tracks and 1 audio tracks was added.\n[04:10:18:INFO] [000:554] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60400000b610): created DTMF sender: RTCDtmfSender {\n[04:10:18:INFO] remainingTones:\n[04:10:18:INFO] duration: 0.100000 sec\n[04:10:18:INFO] interToneGap: 0.050000 sec\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:554] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x604000230180): created sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSa0\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:554] [39691] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[04:10:18:INFO] [000:554] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x6040002301a0): created sender: RTCRtpSender {\n[04:10:18:INFO] senderId: ARDAMSv0\n[04:10:18:INFO] }\n[04:10:18:INFO] [000:557] [16139] (peer_connection.cc:6210): 0 is not ready to use the remote candidate because the local or remote description is not set.\n[04:10:18:INFO] [000:557] [16139] (peer_connection.cc:6210): 1 is not ready to use the remote candidate because the local or remote description is not set.\n[04:10:18:INFO] [000:557] [16139] (peer_connection.cc:2110): CreateAnswer: offer_to_receive_audio is not supported with Unified Plan semantics. Use the RtpTransceiver API instead.\n[04:10:18:INFO] [000:557] [16139] (peer_connection.cc:2115): CreateAnswer: offer_to_receive_video is not supported with Unified Plan semantics. Use the RtpTransceiver API instead.\n[04:10:18:INFO] [000:557] [16139] (peer_connection.cc:5540): Local and Remote descriptions must be applied to get the SSL Role of the session.\n[04:10:18:INFO] [000:557] [16139] (peer_connection.cc:5540): Local and Remote descriptions must be applied to get the SSL Role of the session.\n[04:10:18:INFO] [000:560] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[04:10:18:INFO] [000:561] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[04:10:18:INFO] [000:563] [23047] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0\n[04:10:18:INFO] [000:563] [23047] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 35442ec0\n[04:10:18:INFO] [000:563] [23047] (p2p_transport_channel.cc:438): Set ICE ufrag: f8qE pwd: g/tdCaL4+41pCPyYCJeeI/2k on transport 0\n[04:10:18:INFO] [000:564] [23047] (dtls_transport.cc:370): DtlsTransport[0|1|__]: DTLS setup complete.\n[04:10:18:INFO] [000:564] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:18:INFO] [000:564] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:18:INFO] [000:564] [16139] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv.\n[04:10:18:INFO] [000:564] [16139] (rtp_transceiver.cc:169): Changing transceiver (MID=1) current direction from <not set> to kSendRecv.\n[04:10:18:INFO] [000:564] [41991] (channel.cc:553): Channel enabled\n[04:10:18:INFO] [000:565] [41991] (channel.cc:863): Changing voice state, recv=0 send=0\n[04:10:18:INFO] [000:565] [41991] (channel.cc:553): Channel enabled\n[04:10:18:INFO] [000:565] [41991] (channel.cc:996): Changing video state, send=0\n[04:10:18:INFO] [000:565] [16139] (peer_connection.cc:4173): Session: 3921930938603699854 Old state: kHaveRemoteOffer New state: kStable\n[04:10:18:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 0\n[04:10:18:INFO] [000:565] [41991] (channel.cc:871): Setting local voice description\n[04:10:18:INFO] [000:565] [41991] (webrtc_voice_engine.cc:1324): 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[04:10:18:INFO] [000:565] [41991] (webrtc_voice_engine.cc:1513): Setting receive voice codecs.\n[04:10:18:INFO] [000:565] [41991] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2982798033, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS}\n[04:10:18:INFO] [000:566] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:18:INFO] [000:566] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:18:INFO] [000:566] [41991] (audio_receive_stream.cc:140): ~AudioReceiveStream: 2982798033\n[04:10:18:INFO] [000:566] [41991] (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[04:10:18:INFO] [000:566] [41991] (audio_coding_module.cc:333): Created\n[04:10:18:INFO] [000:566] [41991] (audio_receive_stream.cc:117): AudioReceiveStream: 2982798033\n[04:10:18:INFO] [000:566] [41991] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2982798033, 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[04:10:18:INFO] [000:567] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:18:INFO] [000:567] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:18:INFO] [000:567] [41991] (webrtc_voice_engine.cc:1784): AddSendStream: {id:ARDAMSa0;ssrcs:[4239671322];ssrc_groups:;cname:uXYWzEVOa1rZjSAF;stream_ids:ARDAMS;}\n[04:10:18:INFO] [000:567] [41991] (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[04:10:18:INFO] [000:567] [41991] (audio_coding_module.cc:333): Created\n[04:10:18:INFO] [000:567] [41991] (channel_send.cc:708): Not setting media_transport_ rate observers.\n[04:10:18:INFO] [000:567] [41991] (audio_send_stream.cc:142): AudioSendStream: 4239671322\n[04:10:19:INFO] [000:568] [41991] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 4239671322, 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: uXYWzEVOa1rZjSAF}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: -1, max_bitrate_bps: -1, 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[04:10:19:INFO] [000:571] [41991] (audio_encoder_opus.cc:758): Set Opus bitrate to 32000 bps.\n[04:10:19:INFO] [000:571] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:19:INFO] [000:571] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:19:INFO] [000:571] [41991] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2982798033, local_ssrc: 4239671322, 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[04:10:19:INFO] [000:571] [41991] (channel.cc:676): Add send stream ssrc: 4239671322\n[04:10:19:INFO] [000:572] [41991] (audio_device_module_ios.mm:543): Playing\n[04:10:19:INFO] [000:572] [41991] (audio_device_module_ios.mm:480): InitPlayout\n[04:10:19:INFO] [000:572] [41991] (audio_device_module_ios.mm:506): PlayoutIsInitialized\n[04:10:19:INFO] [000:572] [41991] (audio_device_ios.mm:185): AudioDeviceIOS::InitPlayout\n[04:10:19:INFO] [000:572] [41991] (audio_device_ios.mm:859): AudioDeviceIOS::InitPlayOrRecord\n[04:10:19:INFO] [000:585] [16403] (basic_port_allocator.cc:1372): Net[en0:192.168.72.0/22:Wifi:id=1]: Allocation Phase=Relay\n[04:10:19:INFO] [000:585] [16403] (basic_port_allocator.cc:1372): Net[en0:192.168.72.0/22:Wifi:id=1]: Allocation Phase=Relay\n[04:10:19:INFO] [000:611] [41991] (RTCLogging.mm:33): (RTCAudioSession.mm:99 -[RTCAudioSession initWithAudioSession:]): RTCAudioSession (0x6000001147f0): init.\n[04:10:19:INFO] [000:611] [41991] (RTCLogging.mm:33): (audio_device_ios.mm:824 ConfigureAudioSession): Configuring audio session.\n[04:10:19:INFO] [000:611] [41991] (RTCLogging.mm:33): (RTCAudioSession.mm:699 -[RTCAudioSession configureWebRTCSession:]): Configuring audio session for WebRTC.\n[04:10:19:INFO] [000:612] [41991] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:64 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set category to: AVAudioSessionCategoryPlayAndRecord\n[04:10:19:INFO] [000:612] [41991] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:75 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set mode to: AVAudioSessionModeVoiceChat\n[04:10:19:INFO] [000:612] [41991] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:103 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred sample rate to: 48000.00\n[04:10:19:INFO] [000:612] [41991] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:116 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred IO buffer duration to: 0.020000\n[04:10:19:INFO] [000:612] [41991] (RTCLogging.mm:33): (RTCAudioSession.mm:624 -[RTCAudioSession incrementActivationCount]): Incrementing activation count.\n[04:10:19:INFO] [000:612] [41991] (RTCLogging.mm:33): (RTCAudioSession.mm:377 -[RTCAudioSession setActive:error:]): Number of current activations: 1\n[04:10:19:INFO] [000:612] [41991] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:145 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set input number of channels to: 1\n[04:10:19:INFO] [000:612] [41991] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:158 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set output number of channels to: 1\n[04:10:19:INFO] [000:612] [41991] (RTCLogging.mm:33): (audio_device_ios.mm:835 ConfigureAudioSession): Configured audio session.\n[04:10:19:INFO] [000:612] [41991] (audio_device_ios.mm:678): AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession\n[04:10:19:INFO] [000:612] [41991] (RTCLogging.mm:33): (audio_device_ios.mm:683 SetupAudioBuffersForActiveAudioSession): RTCAudioSession: {\n[04:10:19:INFO] category: AVAudioSessionCategoryPlayAndRecord\n[04:10:19:INFO] categoryOptions: 4\n[04:10:19:INFO] mode: AVAudioSessionModeVoiceChat\n[04:10:19:INFO] isActive: 1\n[04:10:19:INFO] sampleRate: 48000.00\n[04:10:19:INFO] IOBufferDuration: 0.010667\n[04:10:19:INFO] outputNumberOfChannels: 2\n[04:10:19:INFO] inputNumberOfChannels: 2\n[04:10:19:INFO] outputLatency: 0.010000\n[04:10:19:INFO] inputLatency: 0.010000\n[04:10:19:INFO] outputVolume: 0.600000\n[04:10:19:INFO] }\n[04:10:19:INFO] [000:612] [41991] (audio_device_ios.mm:714): frames per I/O buffer: 512\n[04:10:19:INFO] [000:613] [41991] (audio_device_ios.mm:715): bytes per I/O buffer: 1024\n[04:10:19:INFO] [000:613] [41991] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer\n[04:10:19:INFO] [000:613] [41991] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000)\n[04:10:19:INFO] [000:613] [41991] (audio_device_buffer.cc:206): SetPlayoutChannels(1)\n[04:10:19:INFO] [000:613] [41991] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000)\n[04:10:19:INFO] [000:613] [41991] (audio_device_buffer.cc:200): SetRecordingChannels(1)\n[04:10:19:INFO] [000:613] [41991] (fine_audio_buffer.cc:32): FineAudioBuffer\n[04:10:19:INFO] [000:613] [41991] (fine_audio_buffer.cc:34): playout_samples_per_channel_10ms: 480\n[04:10:19:INFO] [000:613] [41991] (fine_audio_buffer.cc:36): playout_channels: 1\n[04:10:19:INFO] [000:613] [41991] (fine_audio_buffer.cc:39): record_samples_per_channel_10ms: 480\n[04:10:19:INFO] [000:613] [41991] (fine_audio_buffer.cc:41): record_channels: 1\n[04:10:19:INFO] [000:613] [41991] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:196 Initialize): Initializing audio unit with sample rate: 48000.000000\n[04:10:19:INFO] [000:613] [41991] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:43 LogStreamDescription): AudioStreamBasicDescription: {\n[04:10:19:INFO] mSampleRate: 48000.00\n[04:10:19:INFO] formatIDString: lpcm\n[04:10:19:INFO] mFormatFlags: 0xC\n[04:10:19:INFO] mBytesPerPacket: 2\n[04:10:19:INFO] mFramesPerPacket: 1\n[04:10:19:INFO] mBytesPerFrame: 2\n[04:10:19:INFO] mChannelsPerFrame: 1\n[04:10:19:INFO] mBitsPerChannel: 16\n[04:10:19:INFO] mReserved: 28672\n[04:10:19:INFO] }\n[04:10:19:INFO] [000:636] [16403] (basic_port_allocator.cc:1372): Net[en0:192.168.72.0/22:Wifi:id=1]: Allocation Phase=Tcp\n[04:10:19:INFO] [000:636] [16403] (port.cc:320): Port[3583fa00::1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port created with network cost 10\n[04:10:19:INFO] [000:636] [16403] (basic_port_allocator.cc:942): Adding allocated port for 0\n[04:10:19:INFO] [000:636] [16403] (basic_port_allocator.cc:961): Port[3583fa00:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Added port to allocator\n[04:10:19:INFO] [000:636] [16403] (basic_port_allocator.cc:979): Port[3583fa00:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Gathered candidate: Cand[:2392887298:1:tcp:1518280447:192.168.72.27:52344:local::0:qFn3:YzaQUb5kFo04/Vz8vnZxMqt5:1:10:0]\n[04:10:19:INFO] [000:636] [16403] (basic_port_allocator.cc:1007): Port[3583fa00:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port ready.\n[04:10:19:INFO] [000:636] [16403] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported.\n[04:10:19:INFO] [000:637] [16403] (p2p_transport_channel.cc:805): Port[3583fa00:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0\n[04:10:19:INFO] [000:637] [16403] (basic_port_allocator.cc:1084): Port[3583fa00:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port completed gathering candidates.\n[04:10:19:INFO] [000:637] [16403] (basic_port_allocator.cc:1159): All candidates gathered for 0:1:0\n[04:10:19:INFO] [000:638] [16403] (p2p_transport_channel.cc:860): P2PTransportChannel: 0, component 1 gathering complete\n[04:10:19:INFO] [000:638] [16403] (basic_port_allocator.cc:1372): Net[en0:192.168.72.0/22:Wifi:id=1]: Allocation Phase=Tcp\n[04:10:19:INFO] [000:638] [16403] (port.cc:320): Port[370c6a00::1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port created with network cost 10\n[04:10:19:INFO] [000:639] [16403] (basic_port_allocator.cc:942): Adding allocated port for 1\n[04:10:19:INFO] [000:639] [16403] (basic_port_allocator.cc:961): Port[370c6a00:1:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Added port to allocator\n[04:10:19:INFO] [000:639] [16403] (basic_port_allocator.cc:979): Port[370c6a00:1:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Gathered candidate: Cand[:2392887298:1:tcp:1518280447:192.168.72.27:52345:local::0:qFn3:YzaQUb5kFo04/Vz8vnZxMqt5:1:10:0]\n[04:10:19:INFO] [000:639] [16403] (basic_port_allocator.cc:1007): Port[370c6a00:1:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port ready.\n[04:10:19:INFO] [000:639] [16403] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported.\n[04:10:19:INFO] [000:639] [16403] (p2p_transport_channel.cc:805): Port[370c6a00:1:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0\n[04:10:19:INFO] [000:640] [41991] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:250 Initialize): Voice Processing I/O unit is now initialized.\n[04:10:19:INFO] [000:642] [16403] (basic_port_allocator.cc:1084): Port[370c6a00:1:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port completed gathering candidates.\n[04:10:19:INFO] [000:642] [16403] (basic_port_allocator.cc:1159): All candidates gathered for 1:1:0\n[04:10:19:INFO] [000:642] [16403] (p2p_transport_channel.cc:860): P2PTransportChannel: 1, component 1 gathering complete\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 2\n[04:10:19:INFO] [000:762] [41991] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:71 GetAGCState): VPIO unit AGC: 0\n[04:10:19:INFO] [000:762] [41991] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:264 Initialize): Failed to get AGC state (1st attempt). Error=-66631.\n[04:10:19:INFO] [000:762] [41991] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:302 Initialize): WebRTC.Audio.BuiltInAGCWasEnabledByDefault: 0\n[04:10:19:INFO] [000:762] [41991] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:308 Initialize): WebRTC.Audio.BuiltInAGCIsEnabled: 0\n[04:10:19:INFO] [000:762] [41991] (audio_device_module_ios.mm:486): output: 0\n[04:10:19:INFO] [000:762] [41991] (audio_device_module_ios.mm:518): StartPlayout\n[04:10:19:INFO] [000:762] [41991] (audio_device_module_ios.mm:543): Playing\n[04:10:19:INFO] [000:762] [41991] (audio_device_buffer.cc:98): StartPlayout\n[04:10:19:INFO] [000:762] [41991] (audio_device_ios.mm:227): AudioDeviceIOS::StartPlayout\n[04:10:19:INFO] [000:762] [41991] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:316 Start): Starting audio unit.\n[04:10:19:INFO] [000:772] [41991] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:320 Start): Failed to start audio unit. Error=-66637\n[04:10:19:INFO] [000:772] [41991] (RTCLogging.mm:33): (audio_device_ios.mm:237 StartPlayout): StartPlayout failed to start audio unit.\n[04:10:19:INFO] [000:772] [41991] (audio_device_module_ios.mm:525): output: -1\n[04:10:19:INFO] [000:773] [41991] (channel.cc:863): Changing voice state, recv=1 send=0\n[04:10:19:INFO] [000:773] [41991] (message_queue.cc:517): Message took 208ms to dispatch. Posted from: SetLocalContent@../../pc/channel.cc:293\n[04:10:19:INFO] [000:773] [41991] (channel.cc:1009): Setting local video description\n[04:10:19:INFO] [000:773] [41991] (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[04:10:19:INFO] [000:774] [41991] (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[04:10:19:INFO] [000:774] [41991] (webrtc_video_engine.cc:2542): MaybeRecreateWebRtcFlexfecStream (recv) because of SetRecvParameters\n[04:10:19:INFO] [000:774] [41991] (webrtc_video_engine.cc:2547): RecreateWebRtcVideoStream (recv) because of SetRecvParameters\n[04:10:19:INFO] [000:774] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:19:INFO] [000:774] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:19:INFO] [000:774] [41991] (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: 97654628, 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: 3853731424, 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[04:10:19:INFO] [000:775] [41991] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0\n[04:10:19:INFO] [000:776] [41991] (receive_statistics_proxy.cc:487): Frames decoded 0\n[04:10:19:INFO] WebRTC.Video.DroppedFrames.Receiver 0\n[04:10:19:INFO] \n[04:10:19:INFO] [000:777] [41991] (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: 97654628, 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: 3853731424, 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[04:10:19:INFO] [000:777] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:19:INFO] [000:777] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:19:INFO] [000:777] [41991] (webrtc_video_engine.cc:1072): AddSendStream: {id:ARDAMSv0;ssrcs:[281759528,3619077003];ssrc_groups:{semantics:FID;ssrcs:[281759528,3619077003]};cname:uXYWzEVOa1rZjSAF;stream_ids:ARDAMS;}\n[04:10:19:INFO] [000:778] [41991] (webrtc_video_engine.cc:1883): RecreateWebRtcStream (send) because of SetCodec.\n[04:10:19:INFO] [000:779] [41991] (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[04:10:19:INFO] [000:780] [5379] (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[04:10:19:INFO] [000:781] [5379] (video_send_stream_impl.cc:250): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [281759528], 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: [3619077003], payload_type: 97}, c_name: uXYWzEVOa1rZjSAF}, 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[04:10:19:INFO] [000:781] [5379] (video_send_stream_impl.cc:278): ERROR: Initial encoder max bitrate = -1 which is <= 0!\n[04:10:19:INFO] [000:781] [5379] (video_send_stream_impl.cc:280): Using default encoder max bitrate = 10 Mbps\n[04:10:19:INFO] [000:782] [5379] (video_stream_encoder.cc:628): ConfigureEncoder requested.\n[04:10:19:INFO] [000:782] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:19:INFO] [000:782] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:19:INFO] [000:782] [41991] (video_send_stream.cc:160): VideoSendStream::Stop\n[04:10:19:INFO] [000:782] [41991] (webrtc_video_engine.cc:1111): SetLocalSsrc on all the receive streams because we added a send stream.\n[04:10:19:INFO] [000:782] [41991] (webrtc_video_engine.cc:2483): RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc=281759528\n[04:10:19:INFO] [000:782] [5379] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[04:10:19:INFO] [000:782] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:19:INFO] [000:782] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:19:INFO] [000:782] [41991] (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: 97654628, 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: 3853731424, 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[04:10:19:INFO] [000:783] [41991] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0\n[04:10:19:INFO] [000:783] [41991] (receive_statistics_proxy.cc:487): Frames decoded 0\n[04:10:19:INFO] WebRTC.Video.DroppedFrames.Receiver 0\n[04:10:19:INFO] \n[04:10:19:INFO] [000:784] [41991] (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: 97654628, local_ssrc: 281759528, 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: 3853731424, 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[04:10:19:INFO] [000:784] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:19:INFO] [000:784] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:19:INFO] [000:784] [41991] (channel.cc:676): Add send stream ssrc: 281759528\n[04:10:19:INFO] [000:784] [41991] (video_send_stream.cc:160): VideoSendStream::Stop\n[04:10:19:INFO] [000:784] [41991] (channel.cc:996): Changing video state, send=0\n[04:10:19:INFO] [000:784] [5379] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[04:10:19:INFO] [000:785] [16139] (peer_connection.cc:4109): Changing IceConnectionState 0 => 1\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 1\n[04:10:19:INFO] [000:785] [23047] (jsep_transport_controller.cc:318): Not adding candidate because the JsepTransport doesn't exist. Ignore it.\n[04:10:19:INFO] [000:785] [16139] (peer_connection.cc:5518): Non-rejected SCTP m= section is needed to get the SSL Role of the SCTP transport.\n[04:10:19:INFO] [000:785] [41991] (webrtc_voice_engine.cc:1485): Setting voice channel options: AudioOptions {}\n[04:10:19:INFO] [000:785] [41991] (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[04:10:19:INFO] [000:785] [41991] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[04:10:19:INFO] [000:785] [41991] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[04:10:19:INFO] [000:785] [41991] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[04:10:19:INFO] [000:785] [41991] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[04:10:19:INFO] [000:785] [41991] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[04:10:19:INFO] [000:785] [41991] (audio_device_module_ios.mm:600): output: 0\n[04:10:19:INFO] [000:785] [41991] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:19:INFO] [000:785] [41991] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:19:INFO] [000:785] [41991] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:19:INFO] [000:785] [41991] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[04:10:19:INFO] [000:785] [41991] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[04:10:19:INFO] [000:785] [41991] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[04:10:19:INFO] [000:785] [41991] (audio_device_module_ios.mm:616): output: 0\n[04:10:19:INFO] [000:785] [41991] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[04:10:19:INFO] [000:785] [41991] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[04:10:19:INFO] [000:785] [41991] (audio_device_module_ios.mm:632): output: 0\n[04:10:19:INFO] [000:785] [41991] (apm_helpers.cc:62): NS set to 0\n[04:10:19:INFO] [000:785] [41991] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[04:10:19:INFO] [000:785] [41991] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[04:10:19:INFO] [000:785] [41991] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[04:10:19:INFO] [000:786] [41991] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[04:10:19:INFO] [000:786] [41991] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[04:10:19:INFO] [000:786] [41991] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[04:10:19:INFO] [000:786] [41991] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[04:10:19:INFO] [000:786] [41991] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[04:10:19:INFO] [000:786] [41991] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[04:10:19:INFO] [000:786] [41991] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:19:INFO] [000:786] [41991] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:19:INFO] [000:786] [41991] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:19:INFO] [000:786] [41991] (webrtc_voice_engine.cc:1503): 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[04:10:19:INFO] [000:786] [41991] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 281759528, options: VideoOptions {noise reduction: false, is_screencast : false, }, source = (source))\n[04:10:19:INFO] [000:786] [5379] (video_stream_encoder.cc:628): ConfigureEncoder requested.\n[04:10:19:INFO] [000:787] [23047] (basic_port_allocator.cc:405): Start getting ports with prune_turn_ports disabled\n[04:10:19:INFO] [000:787] [23047] (basic_port_allocator.cc:109): Filtered out ignored networks:\n[04:10:19:INFO] [000:787] [23047] (basic_port_allocator.cc:111): Net[lo0:::1/128:Loopback:id=3]\n[04:10:19:INFO] [000:787] [23047] (basic_port_allocator.cc:111): Net[lo0:127.0.0.0/8:Loopback:id=2]\n[04:10:19:INFO] [000:787] [23047] (basic_port_allocator.cc:918): Network manager has started\n[04:10:19:INFO] [000:787] [23047] (basic_port_allocator.cc:109): Filtered out ignored networks:\n[04:10:19:INFO] [000:787] [23047] (basic_port_allocator.cc:111): Net[lo0:::1/128:Loopback:id=3]\n[04:10:19:INFO] [000:787] [23047] (basic_port_allocator.cc:111): Net[lo0:127.0.0.0/8:Loopback:id=2]\n[04:10:19:INFO] [000:787] [23047] (basic_port_allocator.cc:833): Allocate ports on 1 networks\n[04:10:19:INFO] [000:787] [23047] (basic_port_allocator.cc:1372): Net[en0:192.168.72.0/22:Wifi:id=1]: Allocation Phase=Udp\n[04:10:19:INFO] [000:787] [23047] (port.cc:320): Port[3704be00::1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port created with network cost 10\n[04:10:19:INFO] [000:787] [23047] (basic_port_allocator.cc:942): Adding allocated port for 0\n[04:10:19:INFO] [000:787] [16139] (message_queue.cc:517): Message took 224ms to dispatch. Posted from: SetLocalDescription@../../api/peer_connection_proxy.h:100\n[04:10:19:INFO] [000:787] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[04:10:19:INFO] [000:788] [23047] (basic_port_allocator.cc:961): Port[3704be00:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Added port to allocator\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 1\n[04:10:19:INFO] [000:788] [23047] (basic_port_allocator.cc:979): Port[3704be00:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Gathered candidate: Cand[:3223547122:1:udp:2122260223:192.168.72.27:55307:local::0:f8qE:g/tdCaL4+41pCPyYCJeeI/2k:1:10:0]\n[04:10:19:INFO] [000:788] [23047] (basic_port_allocator.cc:1007): Port[3704be00:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port ready.\n[04:10:19:INFO] [000:788] [23047] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported.\n[04:10:19:INFO] [000:788] [23047] (p2p_transport_channel.cc:805): Port[3704be00:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0\n[04:10:19:INFO] [000:788] [23047] (port.cc:1128): Conn[370ca000:0:Net[en0:192.168.72.0/22:Wifi:id=1]:Beeo0oqC:1:0:local:udp:192.168.72.27:55307->lTF9M0+e:1:2122260223:local:udp:192.168.72.27:51820|C--W|-|0|0|9115038255631187454|-]: Connection created\n[04:10:19:INFO] [000:788] [771] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0\n[04:10:19:INFO] [000:789] [23047] (p2p_transport_channel.cc:1280): Channel[0|1|__]: Created connection with origin: 2, total: 1\n[04:10:19:INFO] [000:794] [23047] (p2p_transport_channel.cc:1930): Channel[0|1|__]: Transport channel state changed from 0 to 2\n[04:10:19:INFO] [000:794] [23047] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete.\n[04:10:19:INFO] [000:794] [23047] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete.\n[04:10:19:INFO] [000:794] [23047] (p2p_transport_channel.cc:1492): Channel[0|1|__]: Have a pingable connection for the first time; starting to ping.\n[04:10:19:INFO] [000:794] [23047] (basic_port_allocator.cc:1084): Port[3704be00:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Port completed gathering candidates.\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 1\n[04:10:19:INFO] [000:796] [18711] (webrtc_video_engine.cc:1349): Call stats: 7457681, {send_bw_bps: 0, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1}\n[04:10:19:INFO] [000:798] [23047] (port.cc:1776): Conn[370ca000:0:Net[en0:192.168.72.0/22:Wifi:id=1]:Beeo0oqC:1:0:local:udp:192.168.72.27:55307->lTF9M0+e:1:2122260223:local:udp:192.168.72.27:51820|C--W|-|0|0|9115038255631187454|-]: Sent STUN ping, id=41476b322b374e6d4b65734d, use_candidate=0, nomination=0\n[04:10:19:INFO] [000:798] [16403] (port.cc:530): Received STUN ping id=41476b322b374e6d4b65734d from unknown address 192.168.72.27:55307\n[04:10:19:INFO] [000:798] [16403] (port.cc:1128): Conn[36056400:0:Net[en0:192.168.72.0/22:Wifi:id=1]:s3b9xNFe:1:0:local:udp:192.168.72.27:51820->ZkcgZpI8:1:1853824767:prflx:udp:192.168.72.27:55307|C--W|-|0|0|7962116751024340479|-]: Connection created\n[04:10:19:INFO] [000:799] [16403] (p2p_transport_channel.cc:994): Adding connection from peer reflexive candidate: Cand[:1962882453:1:udp:1853824767:192.168.72.27:55307:prflx::0:f8qE::1:10:0]\n[04:10:19:INFO] [000:799] [16403] (port.cc:855): Port[3704b800:0:1:0:local:Net[en0:192.168.72.0/22:Wifi:id=1]]: Sent STUN ping response, to=192.168.72.27:55307, id=41476b322b374e6d4b65734d\n[04:10:19:INFO] [000:799] [16403] (p2p_transport_channel.cc:1930): Channel[0|1|R_]: Transport channel state changed from 0 to 2\n[04:10:19:INFO] [000:799] [16403] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete.\n[04:10:19:INFO] [000:799] [23047] (port.cc:1724): Conn[370ca000:0:Net[en0:192.168.72.0/22:Wifi:id=1]:Beeo0oqC:1:0:local:udp:192.168.72.27:55307->lTF9M0+e:1:2122260223:local:udp:192.168.72.27:51820|C--I|-|0|0|9115038255631187454|-]: Received STUN ping response, id=41476b322b374e6d4b65734d, code=0, rtt=1, pings_since_last_response=41476b322b374e6d4b65734d\n[04:10:19:INFO] [000:799] [16403] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete.\n[04:10:19:INFO] [000:799] [16403] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0\n[04:10:19:INFO] [000:799] [16403] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 3563c5b0\n[04:10:19:INFO] [000:799] [23047] (p2p_transport_channel.cc:278): Switching selected connection due to: candidate pair state changed\n[04:10:19:INFO] [000:799] [16403] (p2p_transport_channel.cc:449): Received remote ICE parameters: ufrag=f8qE, renomination enabled\n[04:10:19:INFO] [000:799] [23047] (p2p_transport_channel.cc:1868): Channel[0|1|__]: New selected connection: Conn[370ca000:0:Net[en0:192.168.72.0/22:Wifi:id=1]:Beeo0oqC:1:0:local:udp:192.168.72.27:55307->lTF9M0+e:1:2122260223:local:udp:192.168.72.27:51820|CRWS|S|0|0|9115038255631187454|1]\n[04:10:19:INFO] [000:799] [16403] (dtls_transport.cc:370): DtlsTransport[0|1|__]: DTLS setup complete.\n[04:10:19:INFO] [000:799] [23047] (channel.cc:376): Network route was changed.\n[04:10:19:INFO] [000:799] [23047] (channel.cc:376): Network route was changed.\n[04:10:19:INFO] [000:799] [23047] (dtls_transport.cc:811): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 1\n[04:10:19:INFO] [000:799] [23047] (openssl_stream_adapter.cc:777): BeginSSL with peer.\n[04:10:19:INFO] [000:799] [18711] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:19:INFO] [000:800] [18711] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:19:INFO] [000:800] [18711] (channel.cc:553): Channel enabled\n[04:10:19:INFO] [000:800] [16403] (p2p_transport_channel.cc:1492): Channel[0|1|R_]: Have a pingable connection for the first time; starting to ping.\n[04:10:19:INFO] [000:800] [16403] (p2p_transport_channel.cc:2456): Selecting connection for triggered check: Conn[36056400:0:Net[en0:192.168.72.0/22:Wifi:id=1]:s3b9xNFe:1:0:local:udp:192.168.72.27:51820->ZkcgZpI8:1:1853824767:prflx:udp:192.168.72.27:55307|CR-W|-|0|0|7962116751024340479|-]\n[04:10:19:INFO] [000:801] [16403] (port.cc:1776): Conn[36056400:0:Net[en0:192.168.72.0/22:Wifi:id=1]:s3b9xNFe:1:0:local:udp:192.168.72.27:51820->ZkcgZpI8:1:1853824767:prflx:udp:192.168.72.27:55307|CR-W|-|0|0|7962116751024340479|-]: Sent STUN ping, id=3449434b4374374d4e376444, use_candidate=0, nomination=0\n[04:10:19:INFO] [000:801] [18711] (channel.cc:863): Changing voice state, recv=1 send=0\n[04:10:19:INFO] [000:801] [18711] (channel.cc:553): Channel enabled\n[04:10:19:INFO] [000:801] [18711] (channel.cc:996): Changing video state, send=0\n[04:10:19:INFO] [000:801] [39691] (peer_connection.cc:4173): Session: 2477746311721103795 Old state: kHaveLocalOffer New state: kStable\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 0\n[04:10:19:INFO] [000:801] [18711] (channel.cc:923): Setting remote voice description\n[04:10:19:INFO] [000:801] [18711] (webrtc_voice_engine.cc:1279): 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[04:10:19:INFO] [000:801] [18711] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2982798033, extmap-allow-mixed: false, extensions: [], c_name: S0NY9KGLI1aoJqMd}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: -1, max_bitrate_bps: -1, 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[04:10:19:INFO] [000:802] [18711] (audio_encoder_opus.cc:758): Set Opus bitrate to 32000 bps.\n[04:10:19:INFO] [000:802] [18711] (webrtc_voice_engine.cc:1702): Recreate all the receive streams because the send codec has changed.\n[04:10:19:INFO] [000:802] [18711] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2982798033, 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: S0NY9KGLI1aoJqMd}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: -1, max_bitrate_bps: -1, 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[04:10:19:INFO] [000:802] [18711] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2982798033, 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: S0NY9KGLI1aoJqMd}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: -1, max_bitrate_bps: -1, 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[04:10:19:INFO] [000:803] [18711] (webrtc_voice_engine.cc:2163): WebRtcVoiceMediaChannel::SetMaxSendBitrate.\n[04:10:19:INFO] [000:803] [18711] (webrtc_voice_engine.cc:1485): Setting voice channel options: AudioOptions {}\n[04:10:19:INFO] [000:803] [18711] (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[04:10:19:INFO] [000:803] [18711] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[04:10:19:INFO] [000:803] [18711] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[04:10:19:INFO] [000:803] [18711] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[04:10:19:INFO] [000:803] [18711] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[04:10:19:INFO] [000:803] [18711] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[04:10:19:INFO] [000:803] [18711] (audio_device_module_ios.mm:600): output: 0\n[04:10:19:INFO] [000:803] [18711] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:19:INFO] [000:803] [18711] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:19:INFO] [000:803] [18711] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:19:INFO] [000:803] [18711] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[04:10:19:INFO] [000:803] [18711] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[04:10:19:INFO] [000:803] [18711] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[04:10:19:INFO] [000:803] [18711] (audio_device_module_ios.mm:616): output: 0\n[04:10:19:INFO] [000:803] [18711] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[04:10:19:INFO] [000:803] [18711] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[04:10:19:INFO] [000:803] [18711] (audio_device_module_ios.mm:632): output: 0\n[04:10:19:INFO] [000:803] [18711] (apm_helpers.cc:62): NS set to 0\n[04:10:19:INFO] [000:803] [18711] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[04:10:19:INFO] [000:803] [18711] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[04:10:19:INFO] [000:803] [18711] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[04:10:19:INFO] [000:804] [18711] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[04:10:19:INFO] [000:804] [18711] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[04:10:19:INFO] [000:804] [18711] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[04:10:19:INFO] [000:804] [18711] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[04:10:19:INFO] [000:804] [18711] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[04:10:19:INFO] [000:804] [18711] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[04:10:19:INFO] [000:804] [18711] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:19:INFO] [000:804] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client enter_early_data\n[04:10:19:INFO] [000:804] [18711] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:19:INFO] [000:804] [18711] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:19:INFO] [000:804] [18711] (webrtc_voice_engine.cc:1503): 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[04:10:19:INFO] [000:804] [18711] (webrtc_voice_engine.cc:1849): AddRecvStream: {id:ARDAMSa0;ssrcs:[4239671322];ssrc_groups:;cname:uXYWzEVOa1rZjSAF;stream_ids:ARDAMS;}\n[04:10:19:INFO] [000:804] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client read_hello_verify_request\n[04:10:19:INFO] [000:804] [23047] (openssl_adapter.cc:794): SSL_connect:error in TLS client read_hello_verify_request\n[04:10:19:INFO] [000:804] [18711] (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[04:10:19:INFO] [000:804] [23047] (dtls_transport.cc:698): DtlsTransport[0|1|__]: DtlsTransport: Started DTLS handshake\n[04:10:19:INFO] [000:804] [16403] (dtls_transport.cc:546): DtlsTransport[0|1|__]: Packet received before DTLS started.\n[04:10:19:INFO] [000:804] [18711] (audio_coding_module.cc:333): Created\n[04:10:19:INFO] [000:804] [23047] (srtp_transport.cc:364): The params in SRTP transport are reset.\n[04:10:19:INFO] [000:804] [16403] (dtls_transport.cc:555): DtlsTransport[0|1|__]: Caching DTLS ClientHello packet until DTLS is started.\n[04:10:19:INFO] [000:805] [23047] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete.\n[04:10:19:INFO] [000:805] [18711] (audio_receive_stream.cc:117): AudioReceiveStream: 4239671322\n[04:10:19:INFO] [000:805] [23047] (basic_port_allocator.cc:1159): All candidates gathered for 0:1:0\n[04:10:19:INFO] [000:805] [23047] (p2p_transport_channel.cc:860): P2PTransportChannel: 0, component 1 gathering complete\n[04:10:19:INFO] [000:805] [18711] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 4239671322, local_ssrc: 2982798033, 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[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 2\n[04:10:19:INFO] [000:805] [16403] (port.cc:1724): Conn[36056400:0:Net[en0:192.168.72.0/22:Wifi:id=1]:s3b9xNFe:1:0:local:udp:192.168.72.27:51820->ZkcgZpI8:1:1853824767:prflx:udp:192.168.72.27:55307|CR-I|-|0|0|7962116751024340479|-]: Received STUN ping response, id=3449434b4374374d4e376444, code=0, rtt=4, pings_since_last_response=3449434b4374374d4e376444\n[04:10:19:INFO] [000:805] [18711] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:19:INFO] [000:805] [18711] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:19:INFO] [000:805] [16403] (p2p_transport_channel.cc:278): Switching selected connection due to: candidate pair state changed\n[04:10:19:INFO] [000:806] [18711] (audio_device_module_ios.mm:543): Playing\n[04:10:19:INFO] [000:806] [18711] (audio_device_module_ios.mm:480): InitPlayout\n[04:10:19:INFO] [000:806] [18711] (audio_device_module_ios.mm:506): PlayoutIsInitialized\n[04:10:19:INFO] [000:806] [18711] (audio_device_ios.mm:185): AudioDeviceIOS::InitPlayout\n[04:10:19:INFO] [000:806] [16403] (p2p_transport_channel.cc:1868): Channel[0|1|R_]: New selected connection: Conn[36056400:0:Net[en0:192.168.72.0/22:Wifi:id=1]:s3b9xNFe:1:0:local:udp:192.168.72.27:51820->ZkcgZpI8:1:1853824767:prflx:udp:192.168.72.27:55307|CRWS|S|0|0|7962116751024340479|4]\n[04:10:19:INFO] [000:806] [18711] (audio_device_ios.mm:859): AudioDeviceIOS::InitPlayOrRecord\n[04:10:19:INFO] [000:806] [16403] (channel.cc:376): Network route was changed.\n[04:10:19:INFO] [000:806] [16403] (channel.cc:376): Network route was changed.\n[04:10:19:INFO] [000:806] [16403] (dtls_transport.cc:811): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 4\n[04:10:19:INFO] [000:806] [16403] (openssl_stream_adapter.cc:777): BeginSSL with peer.\n[04:10:19:INFO] [000:806] [18711] (RTCLogging.mm:33): (audio_device_ios.mm:824 ConfigureAudioSession): Configuring audio session.\n[04:10:19:INFO] [000:806] [18711] (RTCLogging.mm:33): (RTCAudioSession.mm:699 -[RTCAudioSession configureWebRTCSession:]): Configuring audio session for WebRTC.\n[04:10:19:INFO] [000:806] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_hello\n[04:10:19:INFO] [000:806] [16403] (openssl_adapter.cc:794): SSL_accept:error in TLS server read_client_hello\n[04:10:19:INFO] [000:806] [18711] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:116 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred IO buffer duration to: 0.020000\n[04:10:19:INFO] [000:806] [18711] (RTCLogging.mm:33): (RTCAudioSession.mm:624 -[RTCAudioSession incrementActivationCount]): Incrementing activation count.\n[04:10:19:INFO] [000:806] [16403] (dtls_transport.cc:698): DtlsTransport[0|1|__]: DtlsTransport: Started DTLS handshake\n[04:10:19:INFO] [000:806] [18711] (RTCLogging.mm:33): (RTCAudioSession.mm:377 -[RTCAudioSession setActive:error:]): Number of current activations: 2\n[04:10:19:INFO] [000:806] [16403] (srtp_transport.cc:364): The params in SRTP transport are reset.\n[04:10:19:INFO] [000:807] [16403] (dtls_transport.cc:704): DtlsTransport[0|1|__]: Handling cached DTLS ClientHello packet.\n[04:10:19:INFO] [000:807] [18711] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:145 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set input number of channels to: 1\n[04:10:19:INFO] [000:807] [18711] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:158 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set output number of channels to: 1\n[04:10:19:INFO] [000:807] [18711] (RTCLogging.mm:33): (audio_device_ios.mm:835 ConfigureAudioSession): Configured audio session.\n[04:10:19:INFO] [000:807] [18711] (audio_device_ios.mm:678): AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession\n[04:10:19:INFO] [000:807] [18711] (RTCLogging.mm:33): (audio_device_ios.mm:683 SetupAudioBuffersForActiveAudioSession): RTCAudioSession: {\n[04:10:19:INFO] category: AVAudioSessionCategoryPlayAndRecord\n[04:10:19:INFO] categoryOptions: 4\n[04:10:19:INFO] mode: AVAudioSessionModeVoiceChat\n[04:10:19:INFO] isActive: 1\n[04:10:19:INFO] sampleRate: 48000.00\n[04:10:19:INFO] IOBufferDuration: 0.010667\n[04:10:19:INFO] outputNumberOfChannels: 2\n[04:10:19:INFO] inputNumberOfChannels: 2\n[04:10:19:INFO] outputLatency: 0.010000\n[04:10:19:INFO] inputLatency: 0.010000\n[04:10:19:INFO] outputVolume: 0.600000\n[04:10:19:INFO] }\n[04:10:19:INFO] [000:807] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server select_certificate\n[04:10:19:INFO] [000:807] [18711] (audio_device_ios.mm:714): frames per I/O buffer: 512\n[04:10:19:INFO] [000:807] [18711] (audio_device_ios.mm:715): bytes per I/O buffer: 1024\n[04:10:19:INFO] [000:807] [18711] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer\n[04:10:19:INFO] [000:807] [18711] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000)\n[04:10:19:INFO] [000:807] [18711] (audio_device_buffer.cc:206): SetPlayoutChannels(1)\n[04:10:19:INFO] [000:807] [18711] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000)\n[04:10:19:INFO] [000:807] [18711] (audio_device_buffer.cc:200): SetRecordingChannels(1)\n[04:10:19:INFO] [000:807] [18711] (fine_audio_buffer.cc:32): FineAudioBuffer\n[04:10:19:INFO] [000:807] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server select_parameters\n[04:10:19:INFO] [000:807] [18711] (fine_audio_buffer.cc:34): playout_samples_per_channel_10ms: 480\n[04:10:19:INFO] [000:811] [18711] (fine_audio_buffer.cc:36): playout_channels: 1\n[04:10:19:INFO] [000:811] [18711] (fine_audio_buffer.cc:39): record_samples_per_channel_10ms: 480\n[04:10:19:INFO] [000:811] [18711] (fine_audio_buffer.cc:41): record_channels: 1\n[04:10:19:INFO] [000:811] [18711] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:196 Initialize): Initializing audio unit with sample rate: 48000.000000\n[04:10:19:INFO] [000:811] [18711] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:43 LogStreamDescription): AudioStreamBasicDescription: {\n[04:10:19:INFO] mSampleRate: 48000.00\n[04:10:19:INFO] formatIDString: lpcm\n[04:10:19:INFO] mFormatFlags: 0xC\n[04:10:19:INFO] mBytesPerPacket: 2\n[04:10:19:INFO] mFramesPerPacket: 1\n[04:10:19:INFO] mBytesPerFrame: 2\n[04:10:19:INFO] mChannelsPerFrame: 1\n[04:10:19:INFO] mBitsPerChannel: 16\n[04:10:19:INFO] mReserved: 28672\n[04:10:19:INFO] }\n[04:10:19:INFO] [000:811] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_hello\n[04:10:19:INFO] [000:811] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_certificate\n[04:10:19:INFO] [000:811] [18711] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:250 Initialize): Voice Processing I/O unit is now initialized.\n[04:10:19:INFO] [000:812] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_key_exchange\n[04:10:19:INFO] [000:828] [18711] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:71 GetAGCState): VPIO unit AGC: 0\n[04:10:19:INFO] [000:828] [18711] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:264 Initialize): Failed to get AGC state (1st attempt). Error=-66631.\n[04:10:19:INFO] [000:828] [18711] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:302 Initialize): WebRTC.Audio.BuiltInAGCWasEnabledByDefault: 0\n[04:10:19:INFO] [000:828] [18711] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:308 Initialize): WebRTC.Audio.BuiltInAGCIsEnabled: 0\n[04:10:19:INFO] [000:828] [18711] (audio_device_module_ios.mm:486): output: 0\n[04:10:19:INFO] [000:828] [18711] (audio_device_module_ios.mm:518): StartPlayout\n[04:10:19:INFO] [000:828] [18711] (audio_device_module_ios.mm:543): Playing\n[04:10:19:INFO] [000:828] [18711] (audio_device_buffer.cc:98): StartPlayout\n[04:10:19:INFO] [000:828] [18711] (audio_device_ios.mm:227): AudioDeviceIOS::StartPlayout\n[04:10:19:INFO] [000:828] [18711] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:316 Start): Starting audio unit.\n[04:10:19:INFO] [000:832] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_hello_done\n[04:10:19:INFO] [000:835] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_certificate\n[04:10:19:INFO] [000:835] [16403] (openssl_adapter.cc:794): SSL_accept:error in TLS server read_client_certificate\n[04:10:19:INFO] [000:835] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_hello\n[04:10:19:INFO] [000:835] [16403] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete.\n[04:10:19:INFO] [000:835] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_certificate\n[04:10:19:INFO] [000:835] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client read_certificate_status\n[04:10:19:INFO] [000:835] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client verify_server_certificate\n[04:10:19:INFO] [000:835] [23047] (openssl_stream_adapter.cc:1049): Accepted peer certificate.\n[04:10:19:INFO] [000:835] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_key_exchange\n[04:10:19:INFO] [000:835] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client read_certificate_request\n[04:10:19:INFO] [000:841] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_hello_done\n[04:10:19:INFO] [000:841] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_certificate\n[04:10:19:INFO] [000:841] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_key_exchange\n[04:10:19:INFO] [000:842] [18711] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:320 Start): Failed to start audio unit. Error=-66637\n[04:10:19:INFO] [000:842] [18711] (RTCLogging.mm:33): (audio_device_ios.mm:237 StartPlayout): StartPlayout failed to start audio unit.\n[04:10:19:INFO] [000:842] [18711] (audio_device_module_ios.mm:525): output: -1\n[04:10:19:INFO] [000:842] [18711] (channel.cc:719): Add remote ssrc: 4239671322\n[04:10:19:INFO] [000:842] [18711] (channel.cc:863): Changing voice state, recv=1 send=0\n[04:10:19:INFO] [000:842] [18711] (channel.cc:1061): Setting remote video description\n[04:10:19:INFO] [000:843] [18711] (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[04:10:19:INFO] [000:843] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_certificate_verify\n[04:10:19:INFO] [000:853] [18711] (webrtc_video_engine.cc:690): Using codec: VideoCodec[96:H264]\n[04:10:19:INFO] [000:853] [18711] (webrtc_video_engine.cc:1883): RecreateWebRtcStream (send) because of SetCodec.\n[04:10:19:INFO] [000:853] [18711] (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[04:10:19:INFO] [000:853] [9731] (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[04:10:19:INFO] [000:854] [9731] (video_send_stream_impl.cc:250): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [97654628], 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: [3853731424], payload_type: 97}, c_name: S0NY9KGLI1aoJqMd}, 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[04:10:19:INFO] [000:856] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_finished\n[04:10:19:INFO] [000:888] [16403] (openssl_stream_adapter.cc:940): DTLS timeout expired\n[04:10:19:INFO] [000:890] [9731] (video_send_stream_impl.cc:278): ERROR: Initial encoder max bitrate = -1 which is <= 0!\n[04:10:19:INFO] [000:890] [9731] (video_send_stream_impl.cc:280): Using default encoder max bitrate = 10 Mbps\n[04:10:19:INFO] [000:890] [9731] (video_stream_encoder.cc:628): ConfigureEncoder requested.\n[04:10:19:INFO] [000:890] [18711] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:19:INFO] [000:890] [18711] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:19:INFO] [000:890] [16403] (openssl_adapter.cc:794): SSL_accept:error in TLS server read_client_certificate\n[04:10:19:INFO] [000:890] [18711] (video_send_stream.cc:160): VideoSendStream::Stop\n[04:10:19:INFO] [000:890] [9731] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[04:10:19:INFO] [000:890] [18711] (webrtc_video_engine.cc:758): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed.\n[04:10:19:INFO] [000:890] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client finish_flight\n[04:10:19:INFO] [000:890] [18711] (webrtc_video_engine.cc:1174): AddRecvStream: {id:ARDAMSv0;ssrcs:[281759528,3619077003];ssrc_groups:{semantics:FID;ssrcs:[281759528,3619077003]};cname:uXYWzEVOa1rZjSAF;stream_ids:ARDAMS;}\n[04:10:19:INFO] [000:893] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client read_session_ticket\n[04:10:19:INFO] [000:893] [23047] (openssl_adapter.cc:794): SSL_connect:error in TLS client read_session_ticket\n[04:10:19:INFO] [000:893] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server verify_client_certificate\n[04:10:19:INFO] [000:894] [16403] (openssl_stream_adapter.cc:1049): Accepted peer certificate.\n[04:10:19:INFO] [000:894] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_key_exchange\n[04:10:19:INFO] [000:894] [23047] (openssl_adapter.cc:794): SSL_connect:error in TLS client read_session_ticket\n[04:10:19:INFO] [000:894] [18711] (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: 281759528, local_ssrc: 97654628, 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: 3619077003, 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[04:10:19:INFO] [000:894] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_certificate_verify\n[04:10:19:INFO] [000:903] [18711] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:19:INFO] [000:903] [18711] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:19:INFO] [000:903] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server read_change_cipher_spec\n[04:10:19:INFO] [000:903] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server process_change_cipher_spec\n[04:10:19:INFO] [000:903] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server read_next_proto\n[04:10:19:INFO] [000:903] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server read_channel_id\n[04:10:19:INFO] [000:903] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_finished\n[04:10:19:INFO] [000:903] [18711] (channel.cc:719): Add remote ssrc: 281759528\n[04:10:19:INFO] [000:903] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_finished\n[04:10:19:INFO] [000:904] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server finish_server_handshake\n[04:10:19:INFO] [000:904] [16403] (openssl_adapter.cc:784): SSL_accept:TLS server done\n[04:10:19:INFO] [000:904] [16403] (dtls_transport.cc:630): DtlsTransport[0|1|__]: DTLS handshake complete.\n[04:10:19:INFO] [000:904] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client process_change_cipher_spec\n[04:10:19:INFO] [000:904] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_finished\n[04:10:19:INFO] [000:904] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client finish_client_handshake\n[04:10:19:INFO] [000:904] [16403] (jsep_transport_controller.cc:1213): Transport 0 writability changed to 1.\n[04:10:19:INFO] [000:904] [23047] (openssl_adapter.cc:784): SSL_connect:TLS client done\n[04:10:19:INFO] [000:904] [16403] (dtls_srtp_transport.cc:219): Extracting keys from transport: 0\n[04:10:19:INFO] [000:938] [23047] (dtls_transport.cc:630): DtlsTransport[0|1|__]: DTLS handshake complete.\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 2\n[04:10:19:INFO] [000:938] [23047] (jsep_transport_controller.cc:1213): Transport 0 writability changed to 1.\n[04:10:19:INFO] [000:938] [16139] (peer_connection.cc:6044): Changing to ICE connected state because all transports are writable.\n[04:10:19:INFO] [000:938] [23047] (dtls_srtp_transport.cc:219): Extracting keys from transport: 0\n[04:10:19:INFO] [000:938] [16139] (peer_connection.cc:4109): Changing IceConnectionState 1 => 2\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 2\n[04:10:19:INFO] [000:938] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[04:10:19:INFO] [000:938] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[04:10:19:INFO] [000:938] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[04:10:19:INFO] [000:938] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[04:10:19:INFO] [000:938] [9731] (rtp_transport_controller_send.cc:459): Creating fallback congestion controller\n[04:10:19:INFO] [000:944] [16403] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1\n[04:10:19:INFO] [000:944] [23047] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1\n[04:10:19:INFO] [000:956] [23047] (channel.cc:583): Channel writable (0) for the first time\n[04:10:19:INFO] [000:956] [23047] (channel.cc:583): Channel writable (1) for the first time\n[04:10:19:INFO] [000:956] [16403] (channel.cc:583): Channel writable (0) for the first time\n[04:10:19:INFO] [000:956] [16403] (channel.cc:583): Channel writable (1) for the first time\n[04:10:19:INFO] [000:957] [41991] (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[04:10:19:INFO] [000:957] [9731] (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[04:10:19:INFO] [000:957] [18711] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers\n[04:10:19:INFO] [000:965] [41991] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[04:10:19:INFO] [000:965] [41991] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[04:10:19:INFO] [000:965] [5379] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers\n[04:10:19:INFO] [000:965] [9731] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85\n[04:10:19:INFO] [000:965] [9731] (delay_based_bwe.cc:107): Using Trendline filter for delay change estimation with window size 20\n[04:10:19:INFO] [000:965] [41991] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[04:10:19:INFO] [000:965] [41991] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[04:10:19:INFO] [000:965] [41991] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[04:10:19:INFO] [000:966] [9731] (delay_based_bwe.cc:297): BWE Setting start bitrate to: 300 kbps\n[04:10:19:INFO] [000:966] [5379] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps\n[04:10:19:INFO] [000:975] [41991] (audio_device_module_ios.mm:600): output: 0\n[04:10:19:INFO] [000:975] [41991] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:19:INFO] [000:975] [18711] (channel.cc:996): Changing video state, send=1\n[04:10:19:INFO] [000:975] [18711] (message_queue.cc:517): Message took 133ms to dispatch. Posted from: SetRemoteContent@../../pc/channel.cc:302\n[04:10:19:INFO] [000:975] [39691] (peer_connection.cc:4109): Changing IceConnectionState 0 => 1\n[04:10:19:INFO] [000:975] [18711] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 1\n[04:10:19:INFO] [000:975] [18711] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[04:10:19:INFO] [000:975] [7171] (rtp_transport_controller_send.cc:459): Creating fallback congestion controller\n[04:10:19:INFO] [000:975] [18711] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[04:10:19:INFO] [000:975] [39691] (peer_connection.cc:5518): Non-rejected SCTP m= section is needed to get the SSL Role of the SCTP transport.\n[04:10:19:INFO] [000:975] [9731] (probe_controller.cc:266): Measured bitrate: 300000 Minimum to probe further: 1260000\n[04:10:19:INFO] [000:976] [41991] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:19:INFO] [000:980] [18711] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[04:10:19:INFO] [000:980] [39691] (peer_connection.cc:2745): Processing the MSIDs for MID=0 (streams=[ARDAMS]).\n[04:10:19:INFO] [000:980] [39691] (peer_connection.cc:2756): Processing the addition of a remote track for MID=0.\n[04:10:19:INFO] [000:980] [7171] (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[04:10:19:INFO] [000:980] [39691] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv.\n[04:10:19:INFO] [000:980] [9731] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (900000:1687:5)\n[04:10:19:INFO] [000:980] [7171] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85\n[04:10:19:INFO] [000:980] [41991] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:19:INFO] [000:980] [18711] (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[04:10:19:INFO] [000:989] [18711] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead.\n[04:10:19:INFO] [000:989] [7171] (delay_based_bwe.cc:107): Using Trendline filter for delay change estimation with window size 20\n[04:10:19:INFO] [000:989] [18711] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead.\n[04:10:19:INFO] [000:989] [9731] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5)\n[04:10:19:INFO] [000:989] [18711] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead.\n[04:10:19:INFO] [000:989] [41991] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[04:10:19:INFO] [000:989] [9731] (bitrate_allocator.cc:115): Current BWE 300000\n[04:10:19:INFO] [000:989] [7171] (delay_based_bwe.cc:297): BWE Setting start bitrate to: 300 kbps\n[04:10:19:INFO] [000:996] [18711] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable\n[04:10:19:INFO] [000:997] [7171] (probe_controller.cc:266): Measured bitrate: 300000 Minimum to probe further: 1260000\n[04:10:19:INFO] [000:997] [41991] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[04:10:19:INFO] [000:997] [18711] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform\n[04:10:19:INFO] [000:997] [7171] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (900000:1687:5)\n[04:10:19:INFO] [000:997] [18711] (audio_device_module_ios.mm:600): output: 0\n[04:10:19:INFO] [000:997] [7171] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5)\n[04:10:19:INFO] [000:997] [18711] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:19:INFO] [000:997] [7171] (bitrate_allocator.cc:115): Current BWE 300000\n[04:10:19:INFO] [000:997] [41991] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[04:10:19:INFO] [000:997] [18711] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:19:INFO] [001:002] [41991] (audio_device_module_ios.mm:616): output: 0\n[04:10:19:INFO] [001:002] [18711] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:19:INFO] [001:002] [41991] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[04:10:19:INFO] [001:002] [18711] (apm_helpers.cc:48): Echo control set to 0 with mode 0\n[04:10:19:INFO] [001:002] [41991] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[04:10:19:INFO] [001:002] [18711] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable\n[04:10:19:INFO] [001:002] [41991] (audio_device_module_ios.mm:632): output: 0\n[04:10:19:INFO] [001:002] [18711] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform\n[04:10:19:INFO] [001:002] [41991] (apm_helpers.cc:62): NS set to 0\n[04:10:19:INFO] [001:003] [18711] (audio_device_module_ios.mm:616): output: 0\n[04:10:19:INFO] [001:003] [6403] (video_stream_encoder.cc:1568): Video suspend state changed to: not suspended\n[04:10:19:INFO] [001:009] [41991] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[04:10:19:INFO] [001:009] [18711] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable\n[04:10:19:INFO] [001:009] [41991] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[04:10:19:INFO] [001:009] [41991] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[04:10:19:INFO] [001:009] [18711] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform\n[04:10:19:INFO] [001:009] [41991] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[04:10:19:INFO] [001:009] [18711] (audio_device_module_ios.mm:632): output: 0\n[04:10:19:INFO] [001:009] [41991] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[04:10:19:INFO] [001:009] [18711] (apm_helpers.cc:62): NS set to 0\n[04:10:19:INFO] [001:009] [41991] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[04:10:19:INFO] [001:009] [18711] (webrtc_voice_engine.cc:452): NetEq capacity is 200\n[04:10:19:INFO] [001:009] [41991] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[04:10:19:INFO] [001:009] [18711] (webrtc_voice_engine.cc:458): NetEq fast mode? 0\n[04:10:19:INFO] [001:020] [18711] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0\n[04:10:19:INFO] [001:020] [41991] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[04:10:19:INFO] [001:020] [18711] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0\n[04:10:19:INFO] [001:020] [41991] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[04:10:19:INFO] [001:020] [18711] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0\n[04:10:19:INFO] [001:020] [41991] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:19:INFO] [001:020] [18711] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0\n[04:10:19:INFO] [001:020] [18711] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0\n[04:10:19:INFO] [001:020] [18711] (webrtc_voice_engine.cc:511): Setting AGC to 0\n[04:10:19:INFO] [001:020] [18711] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0\n[04:10:19:INFO] [001:020] [18711] (audio_processing_impl.cc:727): Highpass filter activated: 1\n[04:10:19:INFO] [001:020] [41991] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:19:INFO] [001:021] [18711] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0\n[04:10:19:INFO] [001:027] [18711] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:19:INFO] [001:027] [41991] (audio_processing_impl.cc:747): Pre-amplifier activated: 0\n[04:10:19:INFO] [001:027] [18711] (audio_device_module_ios.mm:512): RecordingIsInitialized\n[04:10:19:INFO] [001:027] [41991] (audio_device_module_ios.mm:512): RecordingIsInitialized\n[04:10:19:INFO] [001:027] [18711] (audio_device_module_ios.mm:574): Recording\n[04:10:19:INFO] [001:027] [41991] (audio_device_module_ios.mm:574): Recording\n[04:10:19:INFO] [001:027] [18711] (audio_device_module_ios.mm:493): InitRecording\n[04:10:19:INFO] [001:027] [41991] (audio_device_module_ios.mm:493): InitRecording\n[04:10:19:INFO] [001:027] [18711] (audio_device_module_ios.mm:512): RecordingIsInitialized\n[04:10:19:INFO] [001:027] [41991] (audio_device_module_ios.mm:512): RecordingIsInitialized\n[04:10:19:INFO] [001:027] [18711] (audio_device_module_ios.mm:549): StartRecording\n[04:10:19:INFO] [001:027] [41991] (audio_device_module_ios.mm:549): StartRecording\n[04:10:19:INFO] [001:027] [18711] (audio_device_module_ios.mm:574): Recording\n[04:10:19:INFO] [001:037] [41991] (audio_device_module_ios.mm:574): Recording\n[04:10:19:INFO] [001:037] [18711] (audio_device_buffer.cc:117): StartRecording\n[04:10:19:INFO] [001:037] [41991] (audio_device_buffer.cc:117): StartRecording\n[04:10:19:INFO] [001:037] [41991] (audio_device_ios.mm:280): AudioDeviceIOS::StartRecording\n[04:10:19:INFO] [001:037] [18711] (audio_device_ios.mm:280): AudioDeviceIOS::StartRecording\n[04:10:19:INFO] [001:037] [41991] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:316 Start): Starting audio unit.\n[04:10:19:INFO] [001:037] [18711] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:316 Start): Starting audio unit.\n[04:10:19:INFO] [001:051] [41991] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:320 Start): Failed to start audio unit. Error=-66637\n[04:10:19:INFO] [001:051] [41991] (RTCLogging.mm:33): (audio_device_ios.mm:290 StartRecording): StartRecording failed to start audio unit.\n[04:10:19:INFO] [001:051] [41991] (audio_device_module_ios.mm:556): output: -1\n[04:10:19:INFO] [001:051] [41991] (channel.cc:863): Changing voice state, recv=1 send=1\n[04:10:19:INFO] [001:051] [41991] (message_queue.cc:517): Message took 95ms to dispatch. Posted from: UpdateMediaSendRecvState@../../pc/channel.cc:830\n[04:10:19:INFO] [001:051] [41991] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers\n[04:10:19:INFO] [001:058] [7171] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers\n[04:10:19:INFO] [001:058] [7171] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps\n[04:10:19:INFO] [001:058] [6403] (video_stream_encoder.cc:1568): Video suspend state changed to: not suspended\n[04:10:19:INFO] [001:058] [41991] (channel.cc:996): Changing video state, send=1\n[04:10:19:INFO] [001:062] [18711] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:320 Start): Failed to start audio unit. Error=-66637\n[04:10:19:INFO] [001:063] [18711] (RTCLogging.mm:33): (audio_device_ios.mm:290 StartRecording): StartRecording failed to start audio unit.\n[04:10:19:INFO] [001:063] [18711] (audio_device_module_ios.mm:556): output: -1\n[04:10:19:INFO] [001:063] [18711] (channel.cc:863): Changing voice state, recv=1 send=1\n[04:10:19:INFO] [001:063] [18711] (message_queue.cc:517): Message took 83ms to dispatch. Posted from: UpdateMediaSendRecvState@../../pc/channel.cc:830\n[04:10:19:INFO] [001:063] [18711] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers\n[04:10:19:INFO] [001:072] [6403] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers\n[04:10:19:INFO] [001:072] [18711] (channel.cc:996): Changing video state, send=1\n[04:10:19:INFO] [001:072] [18711] (webrtc_voice_engine.cc:1962): SetOutputVolume() to 1 for recv stream with ssrc 4239671322\n[04:10:19:INFO] [001:072] [39691] (peer_connection.cc:2745): Processing the MSIDs for MID=1 (streams=[ARDAMS]).\n[04:10:19:INFO] [001:072] [39691] (peer_connection.cc:2756): Processing the addition of a remote track for MID=1.\n[04:10:19:INFO] [001:072] [39691] (rtp_transceiver.cc:169): Changing transceiver (MID=1) current direction from <not set> to kSendRecv.\n[04:10:19:INFO] [001:072] [18711] (webrtc_video_engine.cc:1304): SetSink: ssrc:281759528 (ptr)\n[04:10:19:INFO] [001:073] [39691] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60400000bdf0): created DTMF sender: RTCDtmfSender {\n[04:10:19:INFO] remainingTones:\n[04:10:19:INFO] duration: 0.100000 sec\n[04:10:19:INFO] interToneGap: 0.050000 sec\n[04:10:19:INFO] }\n[04:10:19:INFO] [001:073] [39691] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x604000231200): created sender: RTCRtpSender {\n[04:10:19:INFO] senderId: ARDAMSa0\n[04:10:19:INFO] }\n[04:10:19:INFO] [001:073] [39691] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x604000058000): created receiver: RTCRtpReceiver {\n[04:10:19:INFO] receiverId: 77f96473-0b6f-4fd9-9c66-b60f97444634\n[04:10:19:INFO] }\n[04:10:19:INFO] [001:082] [39691] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x604000057940): created transceiver: RTCRtpTransceiver {\n[04:10:19:INFO] sender: RTCRtpSender {\n[04:10:19:INFO] senderId: ARDAMSa0\n[04:10:19:INFO] }\n[04:10:19:INFO] receiver: RTCRtpReceiver {\n[04:10:19:INFO] receiverId: 77f96473-0b6f-4fd9-9c66-b60f97444634\n[04:10:19:INFO] }\n[04:10:19:INFO] }\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving audio on track 77f96473-0b6f-4fd9-9c66-b60f97444634.\n[04:10:19:INFO] [001:083] [39691] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[04:10:19:INFO] [001:083] [39691] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60000022cd00): created sender: RTCRtpSender {\n[04:10:19:INFO] senderId: ARDAMSv0\n[04:10:19:INFO] }\n[04:10:19:INFO] [001:083] [39691] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x600000059a70): created receiver: RTCRtpReceiver {\n[04:10:19:INFO] receiverId: 3efb1fbd-6f62-4b42-9491-83041d61cca3\n[04:10:19:INFO] }\n[04:10:19:INFO] [001:089] [39691] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x600000059ad0): created transceiver: RTCRtpTransceiver {\n[04:10:19:INFO] sender: RTCRtpSender {\n[04:10:19:INFO] senderId: ARDAMSv0\n[04:10:19:INFO] }\n[04:10:19:INFO] receiver: RTCRtpReceiver {\n[04:10:19:INFO] receiverId: 3efb1fbd-6f62-4b42-9491-83041d61cca3\n[04:10:19:INFO] }\n[04:10:19:INFO] }\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving video on track 3efb1fbd-6f62-4b42-9491-83041d61cca3.\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:381 -[ARDAppClient peerConnection:didAddStream:]): Stream with 1 video tracks and 1 audio tracks was added.\n[04:10:19:INFO] [001:089] [39691] (message_queue.cc:517): Message took 298ms to dispatch. Posted from: SetRemoteDescription@../../api/peer_connection_proxy.h:104\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 1\n[04:10:19:INFO] [001:089] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x600000011d30): created DTMF sender: RTCDtmfSender {\n[04:10:19:INFO] remainingTones:\n[04:10:19:INFO] duration: 0.100000 sec\n[04:10:19:INFO] interToneGap: 0.050000 sec\n[04:10:19:INFO] }\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 2\n[04:10:19:INFO] [001:099] [39691] (peer_connection.cc:6050): Changing to ICE completed state because all transports are complete.\n[04:10:19:INFO] [001:099] [39691] (peer_connection.cc:4109): Changing IceConnectionState 1 => 2\n[04:10:19:INFO] [001:099] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60000022c240): created sender: RTCRtpSender {\n[04:10:19:INFO] senderId: ARDAMSa0\n[04:10:19:INFO] }\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 2\n[04:10:19:INFO] [001:099] [39691] (peer_connection.cc:4109): Changing IceConnectionState 2 => 3\n[04:10:19:INFO] [001:099] [16139] (rtp_sender.cc:572): Tried to get DTMF sender from video sender.\n[04:10:19:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 3\n[04:10:19:INFO] [001:108] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60000022cd00): created sender: RTCRtpSender {\n[04:10:19:INFO] senderId: ARDAMSv0\n[04:10:19:INFO] }\n[04:10:19:INFO] [001:110] [23047] (jsep_transport_controller.cc:318): Not adding candidate because the JsepTransport doesn't exist. Ignore it.\n[04:10:19:INFO] [001:113] [41991] (rtc_event_log_impl.cc:211): Stopping WebRTC event log.\n[04:10:19:INFO] [001:113] [41991] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped.\n[04:10:19:INFO] [001:115] [41991] (webrtc_video_engine.cc:2234): VideoSendStream stats: 7458001, {input_fps: 0, encode_fps: 0, encode_ms: 0, encode_usage_perc: 0, target_bps: 300000, media_bps: 0, suspended: false, bw_adapted: false} {ssrc: 281759528, 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[04:10:20:INFO] [001:116] [41991] (webrtc_video_engine.cc:2731): VideoReceiveStream stats: 7458002, {ssrc: 97654628, 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[04:10:20:INFO] [001:116] [41991] (webrtc_video_engine.cc:1349): Call stats: 7458001, {send_bw_bps: 300000, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1}\n[04:10:20:INFO] [001:117] [16139] (peer_connection.cc:4173): Session: 3921930938603699854 Old state: kStable New state: kClosed\n[04:10:20:INFO] [001:117] [41991] (audio_device_module_ios.mm:563): StopRecording\n[04:10:20:INFO] [001:117] [41991] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording\n[04:10:20:INFO] [001:117] [41991] (audio_device_buffer.cc:153): StopRecording\n[04:10:20:INFO] [001:117] [41991] (audio_device_buffer.cc:176): total recording time: 80\n[04:10:20:INFO] [001:117] [41991] (audio_device_module_ios.mm:567): output: 0\n[04:10:20:INFO] [001:122] [41991] (webrtc_voice_engine.cc:1962): SetOutputVolume() to 0 for recv stream with ssrc 2982798033\n[04:10:20:INFO] [001:122] [41991] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 281759528, options: nullptr, source = nullptr)\n[04:10:20:INFO] [001:122] [41991] (webrtc_video_engine.cc:1304): SetSink: ssrc:97654628 nullptr\n[04:10:20:INFO] [001:123] [41991] (channel.cc:563): Channel disabled\n[04:10:20:INFO] [001:123] [41991] (video_send_stream.cc:160): VideoSendStream::Stop\n[04:10:20:INFO] [001:123] [41991] (channel.cc:996): Changing video state, send=0\n[04:10:20:INFO] [001:123] [6403] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[04:10:20:INFO] [001:123] [6403] (rtp_rtcp_impl.cc:360): Failed to send RTCP BYE\n[04:10:20:INFO] [001:123] [6403] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps\n[04:10:20:INFO] [001:123] [6403] (video_stream_encoder.cc:1568): Video suspend state changed to: suspended\n[04:10:20:INFO] [001:123] [41991] (video_send_stream.cc:160): VideoSendStream::Stop\n[04:10:20:INFO] [001:123] [6403] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[04:10:20:INFO] [001:127] [6403] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[04:10:20:INFO] [001:127] [6403] (video_send_stream_impl.cc:338): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [281759528], 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: [3619077003], payload_type: 97}, c_name: uXYWzEVOa1rZjSAF}, 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[04:10:20:INFO] [001:130] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[04:10:20:INFO] [001:130] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[04:10:20:INFO] [001:131] [41991] (send_statistics_proxy.cc:655): Frames encoded 0\n[04:10:20:INFO] WebRTC.Video.DroppedFrames.Capturer 0\n[04:10:20:INFO] WebRTC.Video.DroppedFrames.EncoderQueue 0\n[04:10:20:INFO] WebRTC.Video.DroppedFrames.Encoder 0\n[04:10:20:INFO] WebRTC.Video.DroppedFrames.Ratelimiter 0\n[04:10:20:INFO] [001:131] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[04:10:20:INFO] [001:131] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[04:10:20:INFO] [001:131] [41991] (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: 97654628, local_ssrc: 281759528, 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: 3853731424, 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[04:10:20:INFO] [001:144] [41991] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0\n[04:10:20:INFO] [001:145] [41991] (receive_statistics_proxy.cc:487): Frames decoded 0\n[04:10:20:INFO] WebRTC.Video.DroppedFrames.Receiver 0\n[04:10:20:INFO] \n[04:10:20:INFO] [001:145] [41991] (channel.cc:162): Destroyed channel: 1\n[04:10:20:INFO] [001:145] [41991] (channel.cc:563): Channel disabled\n[04:10:20:INFO] [001:145] [41991] (audio_device_module_ios.mm:532): StopPlayout\n[04:10:20:INFO] [001:145] [41991] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout\n[04:10:20:INFO] [001:145] [41991] (audio_device_buffer.cc:139): StopPlayout\n[04:10:20:INFO] [001:145] [41991] (audio_device_buffer.cc:145): total playout time: 383\n[04:10:20:INFO] [001:145] [41991] (audio_device_module_ios.mm:536): output: 0\n[04:10:20:INFO] [001:145] [41991] (channel.cc:863): Changing voice state, recv=0 send=0\n[04:10:20:INFO] [001:145] [41991] (webrtc_voice_engine.cc:1823): RemoveSendStream: 4239671322\n[04:10:20:INFO] [001:145] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[04:10:20:INFO] [001:145] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[04:10:20:INFO] [001:145] [41991] (audio_send_stream.cc:179): ~AudioSendStream: 4239671322\n[04:10:20:INFO] [001:146] [41991] (webrtc_voice_engine.cc:1900): RemoveRecvStream: 2982798033\n[04:10:20:INFO] [001:146] [41991] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:20:INFO] [001:146] [41991] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:20:INFO] [001:146] [41991] (audio_receive_stream.cc:140): ~AudioReceiveStream: 2982798033\n[04:10:20:INFO] [001:146] [7171] (paced_sender.cc:106): PacedSender paused.\n[04:10:20:INFO] [001:146] [7171] (control_handler.cc:77): Bitrate estimate state changed, BWE: 300 kbps.\n[04:10:20:INFO] [001:146] [41991] (channel.cc:162): Destroyed channel: 0\n[04:10:20:INFO] [001:146] [23047] (openssl_stream_adapter.cc:896): Cleanup\n[04:10:20:INFO] [001:146] [23047] (openssl_adapter.cc:787): SSL3 alert write:warning:close notify\n[04:10:20:INFO] [001:146] [16403] (openssl_adapter.cc:787): SSL3 alert read:warning:close notify\n[04:10:20:INFO] [001:146] [16403] (openssl_stream_adapter.cc:896): Cleanup\n[04:10:20:INFO] [001:146] [16403] (openssl_adapter.cc:787): SSL3 alert write:warning:close notify\n[04:10:20:INFO] [001:153] [41991] (paced_sender.cc:418): ProcessThreadAttached 0x0\n[04:10:20:INFO] [001:161] [41991] (paced_sender.cc:418): ProcessThreadAttached 0x0\n[04:10:20:INFO] [001:161] [16403] (dtls_transport.cc:651): DtlsTransport[0|1|_W]: DTLS transport closed\n[04:10:20:INFO] [001:161] [16403] (jsep_transport_controller.cc:1213): Transport 0 writability changed to 0.\n[04:10:20:INFO] [001:161] [39691] (peer_connection.cc:4109): Changing IceConnectionState 3 => 5\n[04:10:20:INFO] [001:161] [16403] (channel.cc:596): Channel not writable (0)\n[04:10:20:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 5\n[04:10:20:INFO] [001:161] [16403] (channel.cc:596): Channel not writable (1)\n[04:10:20:INFO] [001:161] [16403] (srtp_transport.cc:364): The params in SRTP transport are reset.\n[04:10:20:INFO] [001:161] [41991] (rtc_event_log_impl.cc:211): Stopping WebRTC event log.\n[04:10:20:INFO] [001:161] [41991] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped.\n[04:10:20:INFO] [001:161] [18711] (channel.cc:863): Changing voice state, recv=1 send=1\n[04:10:20:INFO] [001:161] [16139] (peer_connection.cc:6776): Usage signature is 4088\n[04:10:20:INFO] [001:161] [18711] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers\n[04:10:20:INFO] [001:164] [16139] (message_queue.cc:517): Message took 51ms to dispatch. Posted from: Close@../../api/peer_connection_proxy.h:140\n[04:10:20:INFO] [001:165] [7171] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers\n[04:10:20:INFO] [001:165] [18711] (channel.cc:996): Changing video state, send=1\n[04:10:20:INFO] [001:165] [16139] (peer_connection.cc:915): Session: 3921930938603699854 is destroyed.\n[04:10:20:INFO] [001:165] [16139] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue.\n[04:10:20:INFO] [001:165] [41991] (webrtc_video_engine.cc:474): WebRtcVideoEngine::~WebRtcVideoEngine\n[04:10:20:INFO] [001:166] [41991] (webrtc_voice_engine.cc:205): WebRtcVoiceEngine::~WebRtcVoiceEngine\n[04:10:20:INFO] [001:166] [41991] (audio_device_module_ios.mm:532): StopPlayout\n[04:10:20:INFO] [001:166] [41991] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout\n[04:10:20:INFO] [001:166] [41991] (audio_device_module_ios.mm:536): output: 0\n[04:10:20:INFO] [001:166] [41991] (audio_device_module_ios.mm:563): StopRecording\n[04:10:20:INFO] [001:166] [41991] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording\n[04:10:20:INFO] [001:167] [41991] (audio_device_module_ios.mm:567): output: 0\n[04:10:20:INFO] [001:167] [41991] (audio_device_module_ios.mm:581): RegisterAudioCallback\n[04:10:20:INFO] [001:167] [41991] (audio_device_buffer.cc:81): RegisterAudioCallback\n[04:10:20:INFO] [001:167] [41991] (audio_device_module_ios.mm:92): Terminate\n[04:10:20:INFO] [001:167] [41991] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate\n[04:10:20:INFO] [001:167] [41991] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout\n[04:10:20:INFO] [001:167] [41991] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording\n[04:10:20:INFO] [001:169] [41991] (audio_device_module_ios.mm:55): ~AudioDeviceModuleIOS\n[04:10:20:INFO] [001:169] [41991] (audio_device_buffer.cc:75): AudioDeviceBuffer::~dtor\n[04:10:20:INFO] [001:169] [41991] (audio_device_ios.mm:127): AudioDeviceIOS::~dtor<NSThread: 0x600001272a80>{number = 4, name = (null)}\n[04:10:20:INFO] [001:169] [41991] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate\n[04:10:20:INFO] [001:169] [41991] (fine_audio_buffer.cc:46): ~FineAudioBuffer\n[04:10:20:INFO] [001:169] [41991] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:347 Uninitialize): Unintializing audio unit.\n[04:10:20:INFO] [001:172] [41991] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:354 Uninitialize): Uninitialized audio unit.\n[04:10:20:INFO] [001:172] [41991] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:459 DisposeAudioUnit): Disposing audio unit.\n[04:10:20:INFO] Test Case '-[ARDAppClientTest testSession]' passed (1.184 seconds).\n[04:10:20:INFO] Test Suite 'ARDAppClientTest' passed at 2019-04-19 16:10:19.172.\n[04:10:20:INFO] \t Executed 1 test, with 0 failures (0 unexpected) in 1.184 (1.184) seconds\n[04:10:20:INFO] Test Suite 'ARDFileCaptureControllerTests' started at 2019-04-19 16:10:19.172\n[04:10:20:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStarted]' started.\n[04:10:20:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStarted]' passed (0.001 seconds).\n[04:10:20:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStoped]' started.\n[04:10:20:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStoped]' passed (0.001 seconds).\n[04:10:20:INFO] Test Suite 'ARDFileCaptureControllerTests' passed at 2019-04-19 16:10:19.174.\n[04:10:20:INFO] \t Executed 2 tests, with 0 failures (0 unexpected) in 0.001 (0.002) seconds\n[04:10:20:INFO] Test Suite 'ARDSettingsModelTests' started at 2019-04-19 16:10:19.178\n[04:10:20:INFO] Test Case '-[ARDSettingsModelTests testConstraintComponentIsNilWhenInvalidConstraintString]' started.\n[04:10:20:INFO] Test Case '-[ARDSettingsModelTests testConstraintComponentIsNilWhenInvalidConstraintString]' passed (0.001 seconds).\n[04:10:20:INFO] Test Case '-[ARDSettingsModelTests testHeightConstraintFromStore]' started.\n[04:10:20:INFO] Test Case '-[ARDSettingsModelTests testHeightConstraintFromStore]' passed (0.001 seconds).\n[04:10:20:INFO] Test Case '-[ARDSettingsModelTests testRetrievingSetting]' started.\n[04:10:20:INFO] Test Case '-[ARDSettingsModelTests testRetrievingSetting]' passed (0.001 seconds).\n[04:10:20:INFO] Test Case '-[ARDSettingsModelTests testReturningDefaultCallOption]' started.\n[04:10:20:INFO] Test Case '-[ARDSettingsModelTests testReturningDefaultCallOption]' passed (0.001 seconds).\n[04:10:20:INFO] Test Case '-[ARDSettingsModelTests testStoringAudioSetting]' started.\n[04:10:20:INFO] Test Case '-[ARDSettingsModelTests testStoringAudioSetting]' passed (0.001 seconds).\n[04:10:20:INFO] Test Case '-[ARDSettingsModelTests testStoringInvalidConstraintReturnsNo]' started.\n[04:10:20:INFO] Test Case '-[ARDSettingsModelTests testStoringInvalidConstraintReturnsNo]' passed (0.001 seconds).\n[04:10:20:INFO] Test Case '-[ARDSettingsModelTests testWidthConstraintFromStore]' started.\n[04:10:20:INFO] Test Case '-[ARDSettingsModelTests testWidthConstraintFromStore]' passed (0.001 seconds).\n[04:10:20:INFO] Test Suite 'ARDSettingsModelTests' passed at 2019-04-19 16:10:19.190.\n[04:10:20:INFO] \t Executed 7 tests, with 0 failures (0 unexpected) in 0.007 (0.013) seconds\n[04:10:20:INFO] Test Suite 'apprtcmobile_tests.app' passed at 2019-04-19 16:10:19.191.\n[04:10:20:INFO] \t Executed 10 tests, with 0 failures (0 unexpected) in 1.192 (1.204) seconds\n[04:10:20:INFO] Test Suite 'All tests' passed at 2019-04-19 16:10:19.191.\n[04:10:20:INFO] \t Executed 10 tests, with 0 failures (0 unexpected) in 1.192 (1.206) seconds\n[04:10:20:INFO] [001:198] [18711] (rtc_event_log_impl.cc:211): Stopping WebRTC event log.\n[04:10:20:INFO] [001:198] [18711] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped.\n[04:10:20:INFO] [001:199] [39691] (peer_connection.cc:4173): Session: 2477746311721103795 Old state: kStable New state: kClosed\n[04:10:20:INFO] [001:200] [18711] (audio_device_module_ios.mm:563): StopRecording\n[04:10:20:INFO] [001:200] [18711] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording\n[04:10:20:INFO] [001:200] [18711] (audio_device_buffer.cc:153): StopRecording\n[04:10:20:INFO] [001:200] [18711] (audio_device_buffer.cc:176): total recording time: 163\n[04:10:20:INFO] [001:200] [18711] (audio_device_module_ios.mm:567): output: 0\n[04:10:20:INFO] [001:200] [18711] (webrtc_voice_engine.cc:1962): SetOutputVolume() to 0 for recv stream with ssrc 4239671322\n[04:10:20:INFO] [001:200] [18711] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 97654628, options: nullptr, source = nullptr)\n[04:10:20:INFO] [001:200] [18711] (webrtc_video_engine.cc:1304): SetSink: ssrc:281759528 nullptr\n[04:10:20:INFO] [001:203] [18711] (channel.cc:563): Channel disabled\n[04:10:20:INFO] [001:203] [18711] (video_send_stream.cc:160): VideoSendStream::Stop\n[04:10:20:INFO] [001:203] [18711] (channel.cc:996): Changing video state, send=0\n[04:10:20:INFO] [001:203] [9731] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[04:10:20:INFO] [001:203] [9731] (rtp_rtcp_impl.cc:360): Failed to send RTCP BYE\n[04:10:20:INFO] [001:203] [9731] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps\n[04:10:20:INFO] [001:203] [9731] (video_stream_encoder.cc:1568): Video suspend state changed to: suspended\n[04:10:20:INFO] [001:203] [18711] (video_send_stream.cc:160): VideoSendStream::Stop\n[04:10:20:INFO] [001:203] [9731] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[04:10:20:INFO] [001:203] [6403] (video_send_stream_impl.cc:417): VideoSendStream::Stop\n[04:10:20:INFO] [001:204] [6403] (video_send_stream_impl.cc:338): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [97654628], 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: [3853731424], payload_type: 97}, c_name: S0NY9KGLI1aoJqMd}, 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[04:10:20:INFO] [001:212] [18711] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[04:10:20:INFO] [001:212] [18711] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[04:10:20:INFO] [001:212] [18711] (send_statistics_proxy.cc:655): Frames encoded 0\n[04:10:20:INFO] WebRTC.Video.DroppedFrames.Capturer 0\n[04:10:20:INFO] WebRTC.Video.DroppedFrames.EncoderQueue 0\n[04:10:20:INFO] WebRTC.Video.DroppedFrames.Encoder 0\n[04:10:20:INFO] WebRTC.Video.DroppedFrames.Ratelimiter 0\n[04:10:20:INFO] [001:212] [18711] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[04:10:20:INFO] [001:212] [18711] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[04:10:20:INFO] [001:212] [18711] (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: 281759528, local_ssrc: 97654628, 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: 3619077003, 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[04:10:20:INFO] [001:221] [18711] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0\n[04:10:20:INFO] [001:221] [18711] (receive_statistics_proxy.cc:487): Frames decoded 0\n[04:10:20:INFO] WebRTC.Video.DroppedFrames.Receiver 0\n[04:10:20:INFO] \n[04:10:20:INFO] [001:221] [18711] (channel.cc:162): Destroyed channel: 1\n[04:10:20:INFO] [001:221] [18711] (channel.cc:563): Channel disabled\n[04:10:20:INFO] [001:221] [18711] (audio_device_module_ios.mm:532): StopPlayout\n[04:10:20:INFO] [001:221] [18711] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout\n[04:10:20:INFO] [001:221] [18711] (audio_device_buffer.cc:139): StopPlayout\n[04:10:20:INFO] [001:221] [18711] (audio_device_buffer.cc:145): total playout time: 393\n[04:10:20:INFO] [001:222] [18711] (audio_device_module_ios.mm:536): output: 0\n[04:10:20:INFO] [001:222] [18711] (channel.cc:863): Changing voice state, recv=0 send=0\n[04:10:20:INFO] [001:222] [18711] (webrtc_voice_engine.cc:1823): RemoveSendStream: 2982798033\n[04:10:20:INFO] [001:222] [18711] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up\n[04:10:20:INFO] [001:222] [18711] (rtp_transport_controller_send.cc:262): SignalNetworkState Up\n[04:10:20:INFO] [001:222] [18711] (audio_send_stream.cc:179): ~AudioSendStream: 2982798033\n[04:10:20:INFO] [001:222] [18711] (webrtc_voice_engine.cc:1900): RemoveRecvStream: 4239671322\n[04:10:20:INFO] [001:222] [18711] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down\n[04:10:20:INFO] [001:227] [18711] (rtp_transport_controller_send.cc:262): SignalNetworkState Down\n[04:10:20:INFO] [001:227] [18711] (audio_receive_stream.cc:140): ~AudioReceiveStream: 4239671322\n[04:10:20:INFO] [001:227] [7171] (paced_sender.cc:106): PacedSender paused.\n[04:10:20:INFO] [001:227] [7171] (control_handler.cc:77): Bitrate estimate state changed, BWE: 300 kbps.\n[04:10:20:INFO] [001:227] [18711] (channel.cc:162): Destroyed channel: 0\n[04:10:20:INFO] [001:228] [16403] (openssl_stream_adapter.cc:896): Cleanup\n[04:10:20:INFO] [001:229] [18711] (paced_sender.cc:418): ProcessThreadAttached 0x0\n[04:10:20:INFO] [001:229] [18711] (paced_sender.cc:418): ProcessThreadAttached 0x0\n[04:10:20:INFO] [001:229] [18711] (rtc_event_log_impl.cc:211): Stopping WebRTC event log.\n[04:10:20:INFO] [001:229] [18711] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped.\n[04:10:20:INFO] [001:229] [39691] (peer_connection.cc:6776): Usage signature is 4088\n[04:10:20:INFO] [001:229] [39691] (peer_connection.cc:915): Session: 2477746311721103795 is destroyed.\n[04:10:20:INFO] [001:229] [39691] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue.\n[04:10:20:INFO] [001:231] [18711] (webrtc_video_engine.cc:474): WebRtcVideoEngine::~WebRtcVideoEngine\n[04:10:20:INFO] [001:232] [18711] (webrtc_voice_engine.cc:205): WebRtcVoiceEngine::~WebRtcVoiceEngine\n[04:10:20:INFO] [001:232] [18711] (audio_device_module_ios.mm:532): StopPlayout\n[04:10:20:INFO] [001:232] [18711] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout\n[04:10:20:INFO] [001:232] [18711] (audio_device_module_ios.mm:536): output: 0\n[04:10:20:INFO] [001:232] [18711] (audio_device_module_ios.mm:563): StopRecording\n[04:10:20:INFO] [001:232] [18711] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording\n[04:10:20:INFO] [001:232] [18711] (audio_device_module_ios.mm:567): output: 0\n[04:10:20:INFO] [001:232] [18711] (audio_device_module_ios.mm:581): RegisterAudioCallback\n[04:10:20:INFO] [001:232] [18711] (audio_device_buffer.cc:81): RegisterAudioCallback\n[04:10:20:INFO] [001:232] [18711] (audio_device_module_ios.mm:92): Terminate\n[04:10:20:INFO] [001:232] [18711] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate\n[04:10:20:INFO] [001:232] [18711] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout\n[04:10:20:INFO] [001:232] [18711] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording\n[04:10:20:INFO] [001:233] [18711] (audio_device_module_ios.mm:55): ~AudioDeviceModuleIOS\n[04:10:20:INFO] [001:236] [18711] (audio_device_buffer.cc:75): AudioDeviceBuffer::~dtor\n[04:10:20:INFO] [001:236] [18711] (audio_device_ios.mm:127): AudioDeviceIOS::~dtor<NSThread: 0x604000874c00>{number = 3, name = (null)}\n[04:10:20:INFO] [001:236] [18711] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate\n[04:10:20:INFO] [001:236] [18711] (fine_audio_buffer.cc:46): ~FineAudioBuffer\n[04:10:20:INFO] [001:236] [18711] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:347 Uninitialize): Unintializing audio unit.\n[04:10:20:INFO] [001:236] [18711] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:354 Uninitialize): Uninitialized audio unit.\n[04:10:20:INFO] [001:237] [18711] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:459 DisposeAudioUnit): Disposing audio unit.\n[04:10:20:INFO] \n[04:10:20:INFO] \n[04:10:20:INFO] Test session results and logs:\n[04:10:20:INFO] \t/Users/chrome-bot/Library/Developer/Xcode/DerivedData/temporary-bzpukcbbdwktbsdaclipyolfvwfq/Logs/Test/Test-Transient Testing-2019.04.19_16-09-57--0700.xcresult\n[04:10:20:INFO] \n[04:10:20:INFO] 2019-04-19 16:10:19.651 xcodebuild[7348:72779] [MT] IDETestOperationsObserverDebug: 22.009 elapsed -- Testing started completed.\n[04:10:20:INFO] 2019-04-19 16:10:19.651 xcodebuild[7348:72779] [MT] IDETestOperationsObserverDebug: 0.000 sec, +0.000 sec -- start\n[04:10:20:INFO] 2019-04-19 16:10:19.651 xcodebuild[7348:72779] [MT] IDETestOperationsObserverDebug: 22.009 sec, +22.009 sec -- end\n[04:10:20:INFO] ** TEST EXECUTE SUCCEEDED **\n[04:10:20:INFO] \n[04:10:25:DEBUG] Finished print_process_output.\n[04:10:25:INFO] Waiting for test process to terminate.\n[04:10:25:INFO] Test process terminated.\n[04:10:25:DEBUG] Setting sigterm handler.\n[04:10:25:DEBUG] Stdout flushed after test process.\n[04:10:25:DEBUG] Processing test results.\n[04:10:25:INFO] /b/s/w/ir/src/out/Debug-iphonesimulator/iossim returned 0\n\n[04:10:25:DEBUG] Extracting test data.\n[04:10:25:DEBUG] Retrieving crash reports.\n[04:10:25:DEBUG] Retrieving derived data.\n[04:10:25:DEBUG] Making desktop screenshots.\n[04:10:25:DEBUG] Killing simulators.\n[04:10:31:DEBUG] Wiping simulator.\nDevice wiped.\n[04:10:33:DEBUG] End of tear_down.\n", "outputs_ref": { "isolated": "13b66e8c04bdbc909903178c03f34f650ef36624", "isolatedserver": "https://isolateserver.appspot.com", "namespace": "default-gzip" }, "run_id": "4450ae4fdc791911", "server_versions": [ "4267-53b17d5" ], "started_ts": "2019-04-19T23:09:37.428800", "state": "COMPLETED", "tags": [ "build_is_experimental:false", "buildername:iOS64 Sim Debug (iOS 11)", "buildnumber:850", "data:04a2fc63de47fe5fa2f624c35b91fa7fc8f77784", "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:build142-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:b0a6cb78cd34832b49b843b36f65ec5a7a568a79", "test:apprtcmobile_tests", "user:None" ], "task_id": "4450ae4fdc791910", "try_number": "1" } ] }