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python -u /b/s/w/ir/kitchen-checkout/build/scripts/tools/runit.py --show-path --with-third-party-lib -- python /b/s/w/ir/kitchen-checkout/build/scripts/slave/recipe_modules/chromium_swarming/resources/collect_task.py --verbose -o /b/s/w/ir/tmp/t/tmpginBlC.json --task-output-dir /b/s/w/ir/k/recipe_cleanup/0_BfJvP --merge-script /b/s/w/ir/kitchen-checkout/build/scripts/slave/recipe_modules/chromium_swarming/resources/noop_merge.py --merge-script-stdout-file /b/s/w/ir/tmp/t/tmpGtWVx6merge_script_log --merge-additional-args '[]' --summary-json-file /b/s/w/ir/tmp/t/tmptoSpiZ.json -- swarming collect -server https://chromium-swarm.appspot.com -worker 50 -task-summary-python -task-output-stdout all -verbose -requests-json /b/s/w/ir/tmp/t/tmpxbv3g6.json in dir /b/s/w/ir/k: allow_subannotations: False cmd: ['python', '-u', '/b/s/w/ir/kitchen-checkout/build/scripts/tools/runit.py', '--show-path', '--with-third-party-lib', '--', 'python', '/b/s/w/ir/kitchen-checkout/build/scripts/slave/recipe_modules/chromium_swarming/resources/collect_task.py', '--verbose', '-o', '/b/s/w/ir/tmp/t/tmpginBlC.json', '--task-output-dir', '/b/s/w/ir/k/recipe_cleanup/0_BfJvP', '--merge-script', '/b/s/w/ir/kitchen-checkout/build/scripts/slave/recipe_modules/chromium_swarming/resources/noop_merge.py', '--merge-script-stdout-file', '/b/s/w/ir/tmp/t/tmpGtWVx6merge_script_log', '--merge-additional-args', '[]', '--summary-json-file', '/b/s/w/ir/tmp/t/tmptoSpiZ.json', '--', 'swarming', 'collect', '-server', 'https://chromium-swarm.appspot.com', '-worker', '50', '-task-summary-python', '-task-output-stdout', 'all', '-verbose', '-requests-json', '/b/s/w/ir/tmp/t/tmpxbv3g6.json'] env: {} env_prefixes: {'PATH': '/b/s/w/ir/cache/swarming_client'} env_suffixes: {} infra_step: False name_tokens: ('apprtcmobile_tests (iPhone 6s iOS 12.0)',) ok_ret: ALL_OK step_test_data: gen_default_step_test_data(...) trigger_specs: () full environment: Apple_PubSub_Socket_Render: /private/tmp/com.apple.launchd.ucMaoPpcry/Render BOTO_CONFIG: /b/s/w/ir/tmp/gsutil_task/.boto BUILDBUCKET_EXPERIMENTAL: FALSE CIPD_CACHE_DIR: /b/s/cipd_cache/cache CIPD_PROTOCOL: v2 DEVSHELL_CLIENT_PORT: 51991 DOCKER_CONFIG: /b/s/w/ir/tmp/docker_cfg_task DOCKER_TMPDIR: /b/s/w/ir/tmp/docker_tmp_task GIT_CONFIG_NOSYSTEM: 1 GIT_TERMINAL_PROMPT: 0 HOME: /Users/chrome-bot INFRA_GIT_WRAPPER_HOME: /b/s/w/ir/tmp/git_home_task LOGDOG_COORDINATOR_HOST: logs.chromium.org LOGDOG_STREAM_PREFIX: buildbucket/cr-buildbucket.appspot.com/8915859876694831840 LOGDOG_STREAM_PROJECT: webrtc LOGDOG_STREAM_SERVER_PATH: unix:/b/s/w/ir/tmp/ld.sock LOGNAME: chrome-bot LUCI_CONTEXT: /b/s/w/it8YZ0Q1/luci_context.583377508 MAC_CHROMIUM_TMPDIR: /b/s/w/ir/tmp/t NO_GCE_CHECK: False PATH: /b/s/w/ir/cache/swarming_client:/b/s/w/ir/cipd_bin_packages:/b/s/w/ir/cipd_bin_packages/bin:/b/s/cipd_cache/bin:/opt/local/bin:/opt/local/sbin:/usr/local/sbin:/usr/local/git/bin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin PWD: /b/s/w/ir/k PYTHONIOENCODING: UTF-8 PYTHONUNBUFFERED: 1 SHELL: /bin/bash SHLVL: 1 SSH_AUTH_SOCK: /private/tmp/com.apple.launchd.URJxPFWN2u/Listeners SWARMING_BOT_ID: build203-m9 SWARMING_HEADLESS: 1 SWARMING_SERVER: https://chromium-swarm.appspot.com SWARMING_TASK_ID: 44480b01189d9711 TEMP: /b/s/w/ir/tmp/t TEMPDIR: /b/s/w/ir/tmp/t TMP: /b/s/w/ir/tmp/t TMPDIR: /b/s/w/ir/tmp/t USER: chrome-bot VERSIONER_PYTHON_PREFER_32_BIT: no VERSIONER_PYTHON_VERSION: 2.7 VPYTHON_VIRTUALENV_ROOT: /b/s/w/ir/cache/vpython XPC_FLAGS: 0x0 XPC_SERVICE_NAME: 0 _: /b/s/w/ir/cipd_bin_packages/vpython __CF_USER_TEXT_ENCODING: 0x1F4:0x0:0x0 Set PYTHONPATH: /b/s/w/ir/kitchen-checkout/build/scripts:/b/s/w/ir/kitchen-checkout/build/site_config:/b/s/w/ir/kitchen-checkout/build/third_party:/b/s/w/ir/kitchen-checkout/build/third_party/buildbot_8_4p1:/b/s/w/ir/kitchen-checkout/build/third_party/buildbot_slave_8_4:/b/s/w/ir/kitchen-checkout/build/third_party/coverage-3.7.1:/b/s/w/ir/kitchen-checkout/build/third_party/decorator_3_3_1:/b/s/w/ir/kitchen-checkout/build/third_party/google_api_python_client:/b/s/w/ir/kitchen-checkout/build/third_party/httplib2/python2:/b/s/w/ir/kitchen-checkout/build/third_party/infra_libs:/b/s/w/ir/kitchen-checkout/build/third_party/jinja2:/b/s/w/ir/kitchen-checkout/build/third_party/markupsafe:/b/s/w/ir/kitchen-checkout/build/third_party/oauth2client:/b/s/w/ir/kitchen-checkout/build/third_party/pyasn1:/b/s/w/ir/kitchen-checkout/build/third_party/pyasn1-modules:/b/s/w/ir/kitchen-checkout/build/third_party/python-rsa:/b/s/w/ir/kitchen-checkout/build/third_party/requests_2_10_0:/b/s/w/ir/kitchen-checkout/build/third_party/setuptools-0.6c11:/b/s/w/ir/kitchen-checkout/build/third_party/sqlalchemy_0_7_1:/b/s/w/ir/kitchen-checkout/build/third_party/sqlalchemy_migrate_0_7_1:/b/s/w/ir/kitchen-checkout/build/third_party/tempita_0_5:/b/s/w/ir/kitchen-checkout/build/third_party/twisted_10_2:/b/s/w/ir/kitchen-checkout/build/third_party/uritemplate:/b/s/w/ir/kitchen-checkout/build/third_party/site-packages 2019-04-17 23:57:59,862 - root: [DEBUG] Using task_output_dir: '/b/s/w/ir/k/recipe_cleanup/0_BfJvP' 2019-04-17 23:57:59,863 - root: [WARNING] task_output_dir '/b/s/w/ir/k/recipe_cleanup/0_BfJvP' already exists! 2019-04-17 23:57:59,863 - root: [WARNING] task_output_dir existing content: [] 2019-04-17 23:57:59,863 - root: [INFO] collect_cmd: swarming collect -server https://chromium-swarm.appspot.com -worker 50 -task-summary-python -task-output-stdout all -verbose -requests-json /b/s/w/ir/tmp/t/tmpxbv3g6.json -output-dir /b/s/w/ir/k/recipe_cleanup/0_BfJvP -task-summary-json /b/s/w/ir/tmp/t/tmptoSpiZ.json [D2019-04-17T23:57:59.877357-07:00 90427 0 auth.go:1265] Minting a new token {"key":"luci_ctx/56f4e6e3aa849e347a86ebafef4fa483f925594d7a813a86050f3dc84fb14449", "scopes":"https://www.googleapis.com/auth/userinfo.email"} [D2019-04-17T23:57:59.877427-07:00 90427 0 luci_ctx.go:138] POST http://127.0.0.1:51980/rpc/LuciLocalAuthService.GetOAuthToken {"key":"luci_ctx/56f4e6e3aa849e347a86ebafef4fa483f925594d7a813a86050f3dc84fb14449", "scopes":"https://www.googleapis.com/auth/userinfo.email"} [D2019-04-17T23:57:59.878996-07:00 90427 0 auth.go:1222] Token expires in 5m2.121022s {"key":"luci_ctx/56f4e6e3aa849e347a86ebafef4fa483f925594d7a813a86050f3dc84fb14449", "scopes":"https://www.googleapis.com/auth/userinfo.email"} [D2019-04-17T23:58:01.451613-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:02.845367-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:04.282547-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:05.725774-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:07.354566-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:08.875524-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:10.495175-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:12.144104-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:14.008444-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:15.828454-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:17.667360-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:19.602766-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:21.695614-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:24.188969-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:26.576975-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:28.779260-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:31.084154-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:33.530875-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:35.951395-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:38.679485-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:41.431733-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:44.161589-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:46.859994-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:49.892868-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:52.888565-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:56.131919-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:58:59.431246-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:59:02.630672-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:59:05.989798-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 [D2019-04-17T23:59:09.497098-07:00 90427 0 collect.go:344] Waiting task_id: 44480e293dba6910 44480e293dba6910: exit 0 [I 2019-04-17 23:57:52] About to install Xcode 10l232m in Xcode.app for ios [11:57:53:INFO] Using Xcode version 10.0 build 10L232m at /b/s/w/ir/Xcode.app/Contents/Developer Device wiped. [11:58:01:DEBUG] Setting sigterm handler. [11:58:02:INFO] User defaults from command line: [11:58:02:INFO] IDETestRunSpecificationPath = /var/folders/2j/22s2gz0s7hn48k32d47clxf80000gm/T/F4A5C235-51CA-4857-8151-D91DA70AE64A [11:58:02:INFO] [11:58:03:INFO] Testing started on 'iPhone 6s' [11:58:03:INFO] 2019-04-17 23:58:03.256 xcodebuild[5961:141325] IDETestOperationsObserverDebug: Writing diagnostic log for test session to: [11:58:03:INFO] /Users/chrome-bot/Library/Developer/Xcode/DerivedData/temporary-cfnhutywnctkhyghcnwsgfzltxjy/Logs/Test/Test-Transient Testing-2019.04.17_23-58-03--0700.xcresult/1_Test/Diagnostics/apprtcmobile_tests_module-DE9063C5-9467-42C5-A8B6-B99C247F17BD/apprtcmobile_tests_module-615146EF-5D39-43E2-94AB-94845637382E/Session-apprtcmobile_tests_module-2019-04-17_235803-opBeJa.log [11:58:03:INFO] 2019-04-17 23:58:03.257 xcodebuild[5961:141305] [MT] IDETestOperationsObserverDebug: (77798CA1-4054-45AD-A76B-A82281FAAE10) Beginning test session apprtcmobile_tests_module-77798CA1-4054-45AD-A76B-A82281FAAE10 at 2019-04-17 23:58:03.257 with Xcode 10L232m on target <DVTiPhoneSimulator: 0x7f85457babc0> { [11:58:03:INFO] SimDevice: iPhone 6s (D9D387D4-3D8F-477C-A6A9-4756D9D6CECB, iOS 12.0, Shutdown) [11:58:03:INFO] } (12.0 (16A5354b)) [11:58:19:INFO] objc[6074]: Class RTCDispatcher is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x110fc7a70) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/apprtcmobile_tests (0x10cccf330). One of the two will be used. Which one is undefined. [11:58:19:INFO] objc[6074]: Class RTCCameraPreviewView is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x110fc7a98) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/apprtcmobile_tests (0x10cccf358). One of the two will be used. Which one is undefined. [11:58:19:INFO] objc[6074]: Class RTCEncodedImage is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x110fc6940) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/apprtcmobile_tests (0x10cccf3d0). One of the two will be used. Which one is undefined. [11:58:19:INFO] objc[6074]: Class RTCRtpFragmentationHeader is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x110fc6990) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/apprtcmobile_tests (0x10cccf420). One of the two will be used. Which one is undefined. [11:58:19:INFO] objc[6074]: Class RTCVideoCapturer is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x110fc69b8) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/apprtcmobile_tests (0x10cccf448). One of the two will be used. Which one is undefined. [11:58:19:INFO] objc[6074]: Class RTCVideoCodecInfo is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x110fc6a08) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/apprtcmobile_tests (0x10cccf498). One of the two will be used. Which one is undefined. [11:58:19:INFO] objc[6074]: Class RTCVideoEncoderQpThresholds is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x110fc6a58) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/apprtcmobile_tests (0x10cccf4e8). One of the two will be used. Which one is undefined. [11:58:19:INFO] objc[6074]: Class RTCVideoEncoderSettings is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x110fc6ad0) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/apprtcmobile_tests (0x10cccf560). One of the two will be used. Which one is undefined. [11:58:19:INFO] objc[6074]: Class RTCVideoFrame is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x110fc6af8) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/D9D387D4-3D8F-477C-A6A9-4756D9D6CECB/data/Containers/Bundle/Application/B2EC435F-7812-4414-A1FC-ABA43AEC9269/apprtcmobile_tests.app/apprtcmobile_tests (0x10cccf588). One of the two will be used. Which one is undefined. [11:58:19:INFO] 2019-04-17 23:58:19.427 apprtcmobile_tests[6074:142360] +[CATransaction synchronize] called within transaction [11:58:22:INFO] Test Suite 'All tests' started at 2019-04-17 23:58:22.122 [11:58:22:INFO] Test Suite 'apprtcmobile_tests.app' started at 2019-04-17 23:58:22.123 [11:58:22:INFO] Test Suite 'ARDAppClientTest' started at 2019-04-17 23:58:22.123 [11:58:22:INFO] Test Case '-[ARDAppClientTest testSession]' started. [11:58:22:INFO] [000:000] [6403] (audio_device_module.mm:21): CreateAudioDeviceModule [11:58:22:INFO] [000:001] [6403] (audio_device_module_ios.mm:44): current platform is IOS [11:58:22:INFO] [000:001] [6403] (audio_device_module_ios.mm:45): iPhone Audio APIs will be utilized. [11:58:22:INFO] [000:012] [6403] (audio_processing_impl.cc:438): Capture analyzer activated: 0 [11:58:22:INFO] Capture post processor activated: 0 [11:58:22:INFO] Render pre processor activated: 0 [11:58:22:INFO] [000:015] [6403] (webrtc_video_engine.cc:470): WebRtcVideoEngine::WebRtcVideoEngine() [11:58:22:INFO] [000:017] [6403] (webrtc_voice_engine.cc:196): WebRtcVoiceEngine::WebRtcVoiceEngine [11:58:22:INFO] [000:022] [38691] (webrtc_voice_engine.cc:219): WebRtcVoiceEngine::Init [11:58:22:INFO] [000:023] [38691] (webrtc_voice_engine.cc:227): Supported send codecs in order of preference: [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:230): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:230): ISAC/16000/1 (103) [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:230): ISAC/32000/1 (104) [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:230): G722/8000/1 (9) [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:230): ILBC/8000/1 (102) [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:230): PCMU/8000/1 (0) [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:230): PCMA/8000/1 (8) [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:230): CN/32000/1 (106) [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:230): CN/16000/1 (105) [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:230): CN/8000/1 (13) [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:230): telephone-event/48000/1 (110) [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:230): telephone-event/32000/1 (112) [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:230): telephone-event/16000/1 (113) [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:230): telephone-event/8000/1 (126) [11:58:22:INFO] [000:029] [38691] (webrtc_voice_engine.cc:233): Supported recv codecs in order of preference: [11:58:22:INFO] [000:030] [38691] (webrtc_voice_engine.cc:236): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [11:58:22:INFO] [000:030] [38691] (webrtc_voice_engine.cc:236): ISAC/16000/1 (103) [11:58:22:INFO] [000:030] [38691] (webrtc_voice_engine.cc:236): ISAC/32000/1 (104) [11:58:22:INFO] [000:030] [38691] (webrtc_voice_engine.cc:236): G722/8000/1 (9) [11:58:22:INFO] [000:030] [38691] (webrtc_voice_engine.cc:236): ILBC/8000/1 (102) [11:58:22:INFO] [000:030] [38691] (webrtc_voice_engine.cc:236): PCMU/8000/1 (0) [11:58:22:INFO] [000:030] [38691] (webrtc_voice_engine.cc:236): PCMA/8000/1 (8) [11:58:22:INFO] [000:030] [38691] (webrtc_voice_engine.cc:236): CN/32000/1 (106) [11:58:22:INFO] [000:030] [38691] (webrtc_voice_engine.cc:236): CN/16000/1 (105) [11:58:22:INFO] [000:030] [38691] (webrtc_voice_engine.cc:236): CN/8000/1 (13) [11:58:22:INFO] [000:030] [38691] (webrtc_voice_engine.cc:236): telephone-event/48000/1 (110) [11:58:22:INFO] [000:030] [38691] (webrtc_voice_engine.cc:236): telephone-event/32000/1 (112) [11:58:22:INFO] [000:030] [38691] (webrtc_voice_engine.cc:236): telephone-event/16000/1 (113) [11:58:22:INFO] [000:030] [38691] (webrtc_voice_engine.cc:236): telephone-event/8000/1 (126) [11:58:22:INFO] [000:030] [38691] (audio_device_module_ios.mm:69): Init [11:58:22:INFO] [000:031] [38691] (audio_device_buffer.cc:63): AudioDeviceBuffer::ctor [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:117): AudioDeviceIOS::ctor<NSThread: 0x6000039e4900>{number = 3, name = (null)} [11:58:22:INFO] [000:032] [38691] (audio_device_module_ios.mm:49): AttachAudioBuffer [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:133): AudioDeviceIOS::AttachAudioBuffer [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:140): AudioDeviceIOS::Init [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:85): LogDeviceInfo [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:87): system name: iOS [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:88): system version: 12.0 [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:89): device type: iPhone [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:90): device name: x86_64 [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:91): process name: apprtcmobile_tests [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:92): process ID: 6074 [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:93): OS version: Version 12.0 (Build 16A5354b) [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:94): processing cores: 4 [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:95): low power mode: 0 [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:97): TARGET_IPHONE_SIMULATOR is defined [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:99): DeviceIsSimulator: 1 [11:58:22:INFO] [000:032] [38691] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer [11:58:22:INFO] [000:033] [38691] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000) [11:58:22:INFO] [000:033] [38691] (audio_device_buffer.cc:206): SetPlayoutChannels(1) [11:58:22:INFO] [000:033] [38691] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000) [11:58:22:INFO] [000:033] [38691] (audio_device_buffer.cc:200): SetRecordingChannels(1) [11:58:22:INFO] [000:033] [38691] (audio_device_module_ios.mm:406): SetPlayoutDevice(0) [11:58:22:INFO] [000:033] [38691] (audio_device_ios.mm:1006): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetPlayoutDevice(uint16_t): Not implemented [11:58:22:INFO] [000:033] [38691] (audio_device_module_ios.mm:108): InitSpeaker [11:58:22:INFO] [000:034] [38691] (audio_device_module_ios.mm:310): StereoPlayoutIsAvailable [11:58:22:INFO] [000:034] [38691] (audio_device_module_ios.mm:317): output: 0 [11:58:22:INFO] [000:034] [38691] (audio_device_module_ios.mm:322): SetStereoPlayout(0) [11:58:22:INFO] [000:034] [38691] (audio_device_ios.mm:1059): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetStereoPlayout(bool): Not implemented [11:58:22:INFO] [000:034] [38691] (audio_device_module_ios.mm:330): stereo playout is not supported [11:58:22:INFO] [000:034] [38691] (adm_helpers.cc:57): Failed to set stereo playout mode. [11:58:22:INFO] [000:034] [38691] (audio_device_module_ios.mm:468): SetRecordingDevice(0) [11:58:22:INFO] [000:034] [38691] (audio_device_ios.mm:1108): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetRecordingDevice(uint16_t): Not implemented [11:58:22:INFO] [000:034] [38691] (audio_device_module_ios.mm:114): InitMicrophone [11:58:22:INFO] [000:034] [38691] (audio_device_module_ios.mm:277): StereoRecordingIsAvailable [11:58:22:INFO] [000:034] [38691] (audio_device_module_ios.mm:284): output: 0 [11:58:22:INFO] [000:034] [38691] (audio_device_module_ios.mm:289): SetStereoRecording(0) [11:58:22:INFO] [000:034] [38691] (adm_helpers.cc:77): Failed to set stereo recording mode. [11:58:22:INFO] [000:034] [38691] (apm_helpers.cc:32): Setting AGC mode to 2 [11:58:22:INFO] [000:034] [38691] (audio_processing_impl.cc:727): Highpass filter activated: 0 [11:58:22:INFO] [000:034] [38691] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:22:INFO] [000:034] [38691] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:22:INFO] [000:036] [38691] (audio_device_module_ios.mm:581): RegisterAudioCallback [11:58:22:INFO] [000:036] [38691] (audio_device_buffer.cc:81): RegisterAudioCallback [11:58:22:INFO] [000:036] [38691] (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, } [11:58:22:INFO] [000:036] [38691] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [11:58:22:INFO] [000:036] [38691] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [11:58:22:INFO] [000:036] [38691] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [11:58:22:INFO] [000:036] [38691] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [11:58:22:INFO] [000:036] [38691] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [11:58:22:INFO] [000:036] [38691] (audio_device_module_ios.mm:600): output: 0 [11:58:22:INFO] [000:036] [38691] (audio_processing_impl.cc:727): Highpass filter activated: 0 [11:58:22:INFO] [000:037] [38691] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:22:INFO] [000:037] [38691] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:22:INFO] [000:037] [38691] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [11:58:22:INFO] [000:037] [38691] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [11:58:22:INFO] [000:037] [38691] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [11:58:22:INFO] [000:037] [38691] (audio_device_module_ios.mm:616): output: 0 [11:58:22:INFO] [000:037] [38691] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [11:58:22:INFO] [000:037] [38691] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [11:58:22:INFO] [000:037] [38691] (audio_device_module_ios.mm:632): output: 0 [11:58:22:INFO] [000:037] [38691] (audio_processing_impl.cc:727): Highpass filter activated: 0 [11:58:22:INFO] [000:037] [38691] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:22:INFO] [000:037] [38691] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:22:INFO] [000:037] [38691] (apm_helpers.cc:62): NS set to 0 [11:58:22:INFO] [000:037] [38691] (webrtc_voice_engine.cc:447): Stereo swapping enabled? 0 [11:58:22:INFO] [000:037] [38691] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [11:58:22:INFO] [000:037] [38691] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [11:58:22:INFO] [000:037] [38691] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [11:58:22:INFO] [000:037] [38691] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [11:58:22:INFO] [000:037] [38691] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [11:58:22:INFO] [000:037] [38691] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [11:58:22:INFO] [000:037] [38691] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [11:58:22:INFO] [000:038] [38691] (webrtc_voice_engine.cc:511): Setting AGC to 0 [11:58:22:INFO] [000:038] [38691] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [11:58:22:INFO] [000:038] [38691] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:22:INFO] [000:038] [38691] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:22:INFO] [000:038] [38691] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:22:INFO] (RTCLogging.mm:33): (ARDAppClient.m:270 -[ARDAppClient connectToRoomWithId:settings:isLoopback:]_block_invoke): Joined room:testRoom on room server. [11:58:22:INFO] [000:046] [6403] (openssl_identity.cc:44): Making key pair [11:58:22:INFO] [000:213] [6403] (openssl_identity.cc:92): Returning key pair [11:58:22:INFO] [000:213] [6403] (openssl_certificate.cc:58): Making certificate for WebRTC [11:58:22:INFO] [000:224] [6403] (openssl_certificate.cc:108): Returning certificate [11:58:22:INFO] [000:226] [6403] (RTCCertificate.mm:63): CERT PEM [11:58:22:INFO] [000:226] [6403] (RTCCertificate.mm:64): -----BEGIN CERTIFICATE----- [11:58:22:INFO] MIIBnjCCAQegAwIBAgIJALNISa5PKVSLMA0GCSqGSIb3DQEBCwUAMBExDzANBgNV [11:58:22:INFO] BAMMBldlYlJUQzAeFw0xOTA0MTcwNjU4MjJaFw0xOTA0MTgwNzAwMDJaMBExDzAN [11:58:22:INFO] BgNVBAMMBldlYlJUQzCBnzANBgkqhkiG9w0BAQEFAAOBjQAwgYkCgYEAwoYK0vaa [11:58:22:INFO] gXj8nMq5L1ZCbMc3RBq3UhVi6raeiqp4X7CxK/nWmpArPHBZ7jBFriK4/5aWMJOy [11:58:22:INFO] Lvyjx6OQ6yY9DdxW3sbCpQcZ/layFQccDjbp4idLlQqhpre9rxzjmVtesSp0kasE [11:58:22:INFO] 1ceZ15i9Xh4xvZd47pDhQPA3H2LwrFmDJ6MCAwEAATANBgkqhkiG9w0BAQsFAAOB [11:58:22:INFO] gQCjKNLYjLNb2wbM+WF7Zu3Pxv0pa13A0heq5GJaVgGwAr7Cgg3R/FpvnRFMvfon [11:58:22:INFO] k1c1oltRDXTzwwdX+jOMAj3OK6o2LpCB+OVIgCOeLXbZeHqlGpEJySR1+8crWwae [11:58:22:INFO] yOl6IxAqKhgYFrdR2I88VpB/4FZefiz0/U4QdJ3BqtFWAQ== [11:58:22:INFO] -----END CERTIFICATE----- [11:58:22:INFO] [11:58:22:INFO] [000:227] [6403] (RTCConfiguration.mm:213): Have configured cert - using it. [11:58:22:INFO] [000:229] [6403] (RTCConfiguration.mm:218): Created cert from PEM strings. [11:58:22:INFO] [000:233] [38691] (rtc_event_log_impl.cc:71): Creating legacy encoder for RTC event log. [11:58:22:INFO] [000:235] [38691] (peer_connection_factory.cc:407): Using default network controller factory [11:58:22:INFO] [000:237] [38691] (bitrate_prober.cc:66): Bandwidth probing enabled, set to inactive [11:58:22:INFO] [000:240] [38691] (paced_sender.cc:418): ProcessThreadAttached 0x2273d5e0 [11:58:22:INFO] [000:240] [38691] (cpu_info.cc:49): Available number of cores: 4 [11:58:22:INFO] [000:243] [38691] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85 [11:58:22:INFO] [000:243] [38691] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating. [11:58:22:INFO] [000:255] [79375] (webrtc_session_description_factory.cc:314): Using certificate supplied to the constructor. [11:58:22:INFO] [000:258] [79375] (peer_connection.cc:1405): Adding audio transceiver in response to a call to AddTrack. [11:58:22:INFO] (RTCLogging.mm:33): (ARDAppClient.m:396 -[ARDAppClient peerConnectionShouldNegotiate:]): WARNING: Renegotiation needed but unimplemented. [11:58:22:INFO] [000:262] [6403] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60000215c100): created DTMF sender: RTCDtmfSender { [11:58:22:INFO] remainingTones: [11:58:22:INFO] duration: 0.100000 sec [11:58:22:INFO] interToneGap: 0.050000 sec [11:58:22:INFO] } [11:58:22:INFO] [000:262] [6403] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60000234bc20): created sender: RTCRtpSender { [11:58:22:INFO] senderId: ARDAMSa0 [11:58:22:INFO] } [11:58:22:INFO] [000:264] [6403] (RTCLogging.mm:33): (RTCFileVideoCapturer.m:103 -[RTCFileVideoCapturer stopCapture]): File capturer stopped. [11:58:22:INFO] [000:265] [79375] (peer_connection.cc:1405): Adding video transceiver in response to a call to AddTrack. [11:58:22:INFO] [000:265] [79375] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [11:58:22:INFO] [000:265] [6403] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x6000023707a0): created sender: RTCRtpSender { [11:58:22:INFO] senderId: ARDAMSv0 [11:58:22:INFO] } [11:58:22:INFO] [000:266] [6403] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60000214d970): created DTMF sender: RTCDtmfSender { [11:58:22:INFO] remainingTones: [11:58:22:INFO] duration: 0.100000 sec [11:58:22:INFO] interToneGap: 0.050000 sec [11:58:22:INFO] } [11:58:22:INFO] [000:266] [6403] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60000234b740): created sender: RTCRtpSender { [11:58:22:INFO] senderId: ARDAMSa0 [11:58:22:INFO] } [11:58:22:INFO] [000:267] [6403] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x600002d54fc0): created receiver: RTCRtpReceiver { [11:58:22:INFO] receiverId: 828d5df9-b2e6-42a2-b67c-2d0ddb5b0835 [11:58:22:INFO] } [11:58:22:INFO] [000:267] [6403] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x600002d52820): created transceiver: RTCRtpTransceiver { [11:58:22:INFO] sender: RTCRtpSender { [11:58:22:INFO] senderId: ARDAMSa0 [11:58:22:INFO] } [11:58:22:INFO] receiver: RTCRtpReceiver { [11:58:22:INFO] receiverId: 828d5df9-b2e6-42a2-b67c-2d0ddb5b0835 [11:58:22:INFO] } [11:58:22:INFO] } [11:58:22:INFO] [000:268] [79375] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [11:58:22:INFO] [000:268] [6403] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600002370880): created sender: RTCRtpSender { [11:58:22:INFO] senderId: ARDAMSv0 [11:58:22:INFO] } [11:58:22:INFO] [000:268] [6403] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x600002d528e0): created receiver: RTCRtpReceiver { [11:58:22:INFO] receiverId: 6dc52d42-af21-4e8d-bafa-0766c8fc81e3 [11:58:22:INFO] } [11:58:22:INFO] [000:269] [6403] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x600002d54ff0): created transceiver: RTCRtpTransceiver { [11:58:22:INFO] sender: RTCRtpSender { [11:58:22:INFO] senderId: ARDAMSv0 [11:58:22:INFO] } [11:58:22:INFO] receiver: RTCRtpReceiver { [11:58:22:INFO] receiverId: 6dc52d42-af21-4e8d-bafa-0766c8fc81e3 [11:58:22:INFO] } [11:58:22:INFO] } [11:58:22:INFO] [000:274] [79375] (media_session.cc:335): Duplicate id found. Reassigning from 102 to 127 [11:58:22:INFO] [000:274] [79375] (media_session.cc:335): Duplicate id found. Reassigning from 104 to 125 [11:58:22:INFO] [000:274] [79375] (media_session.cc:335): Duplicate id found. Reassigning from 106 to 124 [11:58:22:INFO] [000:274] [79375] (media_session.cc:335): Duplicate id found. Reassigning from 103 to 123 [11:58:22:INFO] [000:274] [79375] (media_session.cc:335): Duplicate id found. Reassigning from 105 to 122 [11:58:22:INFO] [000:275] [79375] (media_session.cc:335): Duplicate id found. Reassigning from 1 to 14 [11:58:22:INFO] [000:275] [79375] (media_session.cc:335): Duplicate id found. Reassigning from 2 to 13 [11:58:22:INFO] [000:275] [79375] (media_session.cc:335): Duplicate id found. Reassigning from 3 to 12 [11:58:22:INFO] [000:275] [79375] (media_session.cc:335): Duplicate id found. Reassigning from 5 to 11 [11:58:22:INFO] [000:280] [38691] (rtc_event_log_impl.cc:186): Starting WebRTC event log. (Timestamp, UTC) = (3908451440, 1555570702414164). [11:58:22:INFO] [000:282] [6403] (audio_device_module.mm:21): CreateAudioDeviceModule [11:58:22:INFO] [000:282] [6403] (audio_device_module_ios.mm:44): current platform is IOS [11:58:22:INFO] [000:282] [6403] (audio_device_module_ios.mm:45): iPhone Audio APIs will be utilized. [11:58:22:INFO] [000:283] [6403] (audio_processing_impl.cc:438): Capture analyzer activated: 0 [11:58:22:INFO] Capture post processor activated: 0 [11:58:22:INFO] Render pre processor activated: 0 [11:58:22:INFO] [000:284] [6403] (webrtc_video_engine.cc:470): WebRtcVideoEngine::WebRtcVideoEngine() [11:58:22:INFO] [000:284] [6403] (webrtc_voice_engine.cc:196): WebRtcVoiceEngine::WebRtcVoiceEngine [11:58:22:INFO] [000:284] [50183] (webrtc_voice_engine.cc:219): WebRtcVoiceEngine::Init [11:58:22:INFO] [000:284] [50183] (webrtc_voice_engine.cc:227): Supported send codecs in order of preference: [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:230): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:230): ISAC/16000/1 (103) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:230): ISAC/32000/1 (104) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:230): G722/8000/1 (9) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:230): ILBC/8000/1 (102) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:230): PCMU/8000/1 (0) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:230): PCMA/8000/1 (8) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:230): CN/32000/1 (106) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:230): CN/16000/1 (105) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:230): CN/8000/1 (13) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:230): telephone-event/48000/1 (110) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:230): telephone-event/32000/1 (112) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:230): telephone-event/16000/1 (113) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:230): telephone-event/8000/1 (126) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:233): Supported recv codecs in order of preference: [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:236): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:236): ISAC/16000/1 (103) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:236): ISAC/32000/1 (104) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:236): G722/8000/1 (9) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:236): ILBC/8000/1 (102) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:236): PCMU/8000/1 (0) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:236): PCMA/8000/1 (8) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:236): CN/32000/1 (106) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:236): CN/16000/1 (105) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:236): CN/8000/1 (13) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:236): telephone-event/48000/1 (110) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:236): telephone-event/32000/1 (112) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:236): telephone-event/16000/1 (113) [11:58:22:INFO] [000:285] [50183] (webrtc_voice_engine.cc:236): telephone-event/8000/1 (126) [11:58:22:INFO] [000:285] [50183] (audio_device_module_ios.mm:69): Init [11:58:22:INFO] [000:285] [50183] (audio_device_buffer.cc:63): AudioDeviceBuffer::ctor [11:58:22:INFO] [000:285] [50183] (audio_device_ios.mm:117): AudioDeviceIOS::ctor<NSThread: 0x6000039e4700>{number = 4, name = (null)} [11:58:22:INFO] [000:285] [50183] (audio_device_module_ios.mm:49): AttachAudioBuffer [11:58:22:INFO] [000:285] [50183] (audio_device_ios.mm:133): AudioDeviceIOS::AttachAudioBuffer [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:140): AudioDeviceIOS::Init [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:85): LogDeviceInfo [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:87): system name: iOS [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:88): system version: 12.0 [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:89): device type: iPhone [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:90): device name: x86_64 [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:91): process name: apprtcmobile_tests [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:92): process ID: 6074 [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:93): OS version: Version 12.0 (Build 16A5354b) [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:94): processing cores: 4 [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:95): low power mode: 0 [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:97): TARGET_IPHONE_SIMULATOR is defined [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:99): DeviceIsSimulator: 1 [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer [11:58:22:INFO] [000:286] [50183] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000) [11:58:22:INFO] [000:286] [50183] (audio_device_buffer.cc:206): SetPlayoutChannels(1) [11:58:22:INFO] [000:286] [50183] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000) [11:58:22:INFO] [000:286] [50183] (audio_device_buffer.cc:200): SetRecordingChannels(1) [11:58:22:INFO] [000:286] [50183] (audio_device_module_ios.mm:406): SetPlayoutDevice(0) [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:1006): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetPlayoutDevice(uint16_t): Not implemented [11:58:22:INFO] [000:286] [50183] (audio_device_module_ios.mm:108): InitSpeaker [11:58:22:INFO] [000:286] [50183] (audio_device_module_ios.mm:310): StereoPlayoutIsAvailable [11:58:22:INFO] [000:286] [50183] (audio_device_module_ios.mm:317): output: 0 [11:58:22:INFO] [000:286] [50183] (audio_device_module_ios.mm:322): SetStereoPlayout(0) [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:1059): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetStereoPlayout(bool): Not implemented [11:58:22:INFO] [000:286] [50183] (audio_device_module_ios.mm:330): stereo playout is not supported [11:58:22:INFO] [000:286] [50183] (adm_helpers.cc:57): Failed to set stereo playout mode. [11:58:22:INFO] [000:286] [50183] (audio_device_module_ios.mm:468): SetRecordingDevice(0) [11:58:22:INFO] [000:286] [50183] (audio_device_ios.mm:1108): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetRecordingDevice(uint16_t): Not implemented [11:58:22:INFO] [000:286] [50183] (audio_device_module_ios.mm:114): InitMicrophone [11:58:22:INFO] [000:286] [50183] (audio_device_module_ios.mm:277): StereoRecordingIsAvailable [11:58:22:INFO] [000:286] [50183] (audio_device_module_ios.mm:284): output: 0 [11:58:22:INFO] [000:286] [50183] (audio_device_module_ios.mm:289): SetStereoRecording(0) [11:58:22:INFO] [000:286] [50183] (adm_helpers.cc:77): Failed to set stereo recording mode. [11:58:22:INFO] [000:286] [50183] (apm_helpers.cc:32): Setting AGC mode to 2 [11:58:22:INFO] [000:286] [50183] (audio_processing_impl.cc:727): Highpass filter activated: 0 [11:58:22:INFO] [000:286] [50183] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:22:INFO] [000:286] [50183] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:22:INFO] [000:286] [50183] (audio_device_module_ios.mm:581): RegisterAudioCallback [11:58:22:INFO] [000:287] [50183] (audio_device_buffer.cc:81): RegisterAudioCallback [11:58:22:INFO] [000:287] [50183] (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, } [11:58:22:INFO] [000:287] [50183] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [11:58:22:INFO] [000:287] [50183] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [11:58:22:INFO] [000:287] [50183] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [11:58:22:INFO] [000:287] [50183] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [11:58:22:INFO] [000:287] [50183] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [11:58:22:INFO] [000:287] [50183] (audio_device_module_ios.mm:600): output: 0 [11:58:22:INFO] [000:287] [50183] (audio_processing_impl.cc:727): Highpass filter activated: 0 [11:58:22:INFO] [000:287] [50183] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:22:INFO] [000:287] [50183] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:22:INFO] [000:287] [50183] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [11:58:22:INFO] [000:287] [50183] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [11:58:22:INFO] [000:287] [50183] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [11:58:22:INFO] [000:287] [50183] (audio_device_module_ios.mm:616): output: 0 [11:58:22:INFO] [000:287] [50183] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [11:58:22:INFO] [000:287] [50183] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [11:58:22:INFO] [000:287] [50183] (audio_device_module_ios.mm:632): output: 0 [11:58:22:INFO] [000:287] [50183] (audio_processing_impl.cc:727): Highpass filter activated: 0 [11:58:22:INFO] [000:287] [50183] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:22:INFO] [000:287] [50183] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:22:INFO] [000:287] [50183] (apm_helpers.cc:62): NS set to 0 [11:58:22:INFO] [000:287] [50183] (webrtc_voice_engine.cc:447): Stereo swapping enabled? 0 [11:58:22:INFO] [000:287] [50183] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [11:58:22:INFO] [000:287] [50183] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [11:58:22:INFO] [000:287] [50183] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [11:58:22:INFO] [000:287] [50183] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [11:58:22:INFO] [000:287] [50183] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [11:58:22:INFO] [000:287] [50183] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [11:58:22:INFO] [000:287] [50183] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [11:58:22:INFO] [000:287] [50183] (webrtc_voice_engine.cc:511): Setting AGC to 0 [11:58:22:INFO] [000:287] [50183] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [11:58:22:INFO] [000:287] [50183] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:22:INFO] [000:288] [50183] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:22:INFO] [000:288] [50183] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:22:INFO] (RTCLogging.mm:33): (ARDAppClient.m:270 -[ARDAppClient connectToRoomWithId:settings:isLoopback:]_block_invoke): Joined room:testRoom on room server. [11:58:22:INFO] [000:288] [6403] (openssl_identity.cc:44): Making key pair [11:58:23:INFO] [000:968] [6403] (openssl_identity.cc:92): Returning key pair [11:58:23:INFO] [000:968] [6403] (openssl_certificate.cc:58): Making certificate for WebRTC [11:58:23:INFO] [000:971] [6403] (openssl_certificate.cc:108): Returning certificate [11:58:23:INFO] [000:971] [6403] (RTCCertificate.mm:63): CERT PEM [11:58:23:INFO] [000:971] [6403] (RTCCertificate.mm:64): -----BEGIN CERTIFICATE----- [11:58:23:INFO] MIIBnjCCAQegAwIBAgIJAOoxZiTR3TFtMA0GCSqGSIb3DQEBCwUAMBExDzANBgNV [11:58:23:INFO] BAMMBldlYlJUQzAeFw0xOTA0MTcwNjU4MjJaFw0xOTA0MTgwNzAwMDJaMBExDzAN [11:58:23:INFO] BgNVBAMMBldlYlJUQzCBnzANBgkqhkiG9w0BAQEFAAOBjQAwgYkCgYEArv0mq7wR [11:58:23:INFO] 6b/SjbDjLQNktDYAHzUznJcvUCOGXtTBcjwdyIGUqzAFJQOGxjKCJ5tb0I/oL2Vt [11:58:23:INFO] lQOvs9ofu63uw1pa+uFsnp45mwQ1q5iLoSl3kVJHSABJA/AQkiiyUuimqbLpYH66 [11:58:23:INFO] UxHP6tZ1cyq3FFHZvFt6ZKtIa0xWlmgJyScCAwEAATANBgkqhkiG9w0BAQsFAAOB [11:58:23:INFO] gQCSQ0S7jEf0vKyAa85u3rHjsMOrmQjVqCzq/fE0tKnImq0YMFqfM/P9m+bwAPm7 [11:58:23:INFO] t1SzLxGOCX/cimCxRh8/WgrwkI9MYABPDiEVnc4dPZ7UFjueGEvHqwGLDYotOXIg [11:58:23:INFO] F3lVgSNz+aGpf1UR9hIuHjKyE5Pdgm+MuHnqaKGRakRXNg== [11:58:23:INFO] -----END CERTIFICATE----- [11:58:23:INFO] [11:58:23:INFO] [000:971] [6403] (RTCConfiguration.mm:213): Have configured cert - using it. [11:58:23:INFO] [000:973] [6403] (RTCConfiguration.mm:218): Created cert from PEM strings. [11:58:23:INFO] [000:973] [50183] (rtc_event_log_impl.cc:71): Creating legacy encoder for RTC event log. [11:58:23:INFO] [000:978] [50183] (peer_connection_factory.cc:407): Using default network controller factory [11:58:23:INFO] [000:978] [50183] (bitrate_prober.cc:66): Bandwidth probing enabled, set to inactive [11:58:23:INFO] [000:978] [50183] (paced_sender.cc:418): ProcessThreadAttached 0x2240c560 [11:58:23:INFO] [000:978] [50183] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85 [11:58:23:INFO] [000:978] [50183] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating. [11:58:23:INFO] [000:979] [79107] (webrtc_session_description_factory.cc:314): Using certificate supplied to the constructor. [11:58:23:INFO] [000:979] [79107] (peer_connection.cc:1405): Adding audio transceiver in response to a call to AddTrack. [11:58:23:INFO] (RTCLogging.mm:33): (ARDAppClient.m:396 -[ARDAppClient peerConnectionShouldNegotiate:]): WARNING: Renegotiation needed but unimplemented. [11:58:23:INFO] [000:980] [6403] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60000214da60): created DTMF sender: RTCDtmfSender { [11:58:23:INFO] remainingTones: [11:58:23:INFO] duration: 0.100000 sec [11:58:23:INFO] interToneGap: 0.050000 sec [11:58:23:INFO] } [11:58:23:INFO] [000:980] [6403] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x6000023719e0): created sender: RTCRtpSender { [11:58:23:INFO] senderId: ARDAMSa0 [11:58:23:INFO] } [11:58:23:INFO] [000:980] [6403] (RTCLogging.mm:33): (RTCFileVideoCapturer.m:103 -[RTCFileVideoCapturer stopCapture]): File capturer stopped. [11:58:23:INFO] [000:980] [79107] (peer_connection.cc:1405): Adding video transceiver in response to a call to AddTrack. [11:58:23:INFO] [000:980] [79107] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [11:58:23:INFO] [000:983] [6403] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600002375420): created sender: RTCRtpSender { [11:58:23:INFO] senderId: ARDAMSv0 [11:58:23:INFO] } [11:58:23:INFO] [000:983] [6403] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60000215c550): created DTMF sender: RTCDtmfSender { [11:58:23:INFO] remainingTones: [11:58:23:INFO] duration: 0.100000 sec [11:58:23:INFO] interToneGap: 0.050000 sec [11:58:23:INFO] } [11:58:23:INFO] [000:983] [6403] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600002371de0): created sender: RTCRtpSender { [11:58:23:INFO] senderId: ARDAMSa0 [11:58:23:INFO] } [11:58:23:INFO] [000:983] [6403] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x600002d53150): created receiver: RTCRtpReceiver { [11:58:23:INFO] receiverId: f51d84c6-6f26-4e91-bbb8-d9da7e44e00d [11:58:23:INFO] } [11:58:23:INFO] [000:983] [6403] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x600002d561f0): created transceiver: RTCRtpTransceiver { [11:58:23:INFO] sender: RTCRtpSender { [11:58:23:INFO] senderId: ARDAMSa0 [11:58:23:INFO] } [11:58:23:INFO] receiver: RTCRtpReceiver { [11:58:23:INFO] receiverId: f51d84c6-6f26-4e91-bbb8-d9da7e44e00d [11:58:23:INFO] } [11:58:23:INFO] } [11:58:23:INFO] [000:985] [79107] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [11:58:23:INFO] [000:985] [6403] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x6000023753a0): created sender: RTCRtpSender { [11:58:23:INFO] senderId: ARDAMSv0 [11:58:23:INFO] } [11:58:23:INFO] [000:985] [6403] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x600002d531b0): created receiver: RTCRtpReceiver { [11:58:23:INFO] receiverId: caa41d84-778a-4a49-9428-e662419ae72f [11:58:23:INFO] } [11:58:23:INFO] [000:985] [6403] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x600002d53180): created transceiver: RTCRtpTransceiver { [11:58:23:INFO] sender: RTCRtpSender { [11:58:23:INFO] senderId: ARDAMSv0 [11:58:23:INFO] } [11:58:23:INFO] receiver: RTCRtpReceiver { [11:58:23:INFO] receiverId: caa41d84-778a-4a49-9428-e662419ae72f [11:58:23:INFO] } [11:58:23:INFO] } [11:58:23:INFO] [000:986] [50183] (rtc_event_log_impl.cc:186): Starting WebRTC event log. (Timestamp, UTC) = (3909157680, 1555570703120425). [11:58:23:INFO] [000:988] [6403] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [11:58:23:INFO] [000:990] [6403] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [11:58:23:INFO] [000:996] [49427] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [11:58:23:INFO] [000:996] [49427] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [11:58:23:INFO] [000:996] [49427] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [11:58:23:INFO] [000:996] [49427] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [11:58:23:INFO] [000:996] [49427] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [11:58:23:INFO] [000:996] [49427] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [11:58:23:INFO] [000:996] [49427] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [11:58:23:INFO] [000:997] [49427] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0 [11:58:23:INFO] [000:997] [49427] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 2240dde0 [11:58:23:INFO] [000:997] [49427] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [11:58:23:INFO] [000:997] [49427] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [11:58:23:INFO] [000:997] [49427] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [11:58:23:INFO] [000:997] [49427] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [11:58:23:INFO] [000:997] [49427] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [11:58:23:INFO] [000:997] [49427] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [11:58:23:INFO] [000:997] [49427] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [11:58:23:INFO] [000:997] [49427] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 1 transport 0 [11:58:23:INFO] [000:998] [49427] (dtls_srtp_transport.cc:66): Setting RTP Transport on 1 transport 2240e210 [11:58:23:INFO] [000:998] [49427] (p2p_transport_channel.cc:432): Set ICE ufrag: YLj5 pwd: /j0o5yoRFRv8u1m0LFUsi0dh on transport 0 [11:58:23:INFO] [000:998] [49427] (p2p_transport_channel.cc:432): Set ICE ufrag: YLj5 pwd: /j0o5yoRFRv8u1m0LFUsi0dh on transport 1 [11:58:23:INFO] [000:998] [38691] (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, } [11:58:23:INFO] [000:998] [38691] (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, } [11:58:23:INFO] [000:999] [38691] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [11:58:23:INFO] [000:999] [38691] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [11:58:23:INFO] [000:999] [38691] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [11:58:23:INFO] [000:999] [38691] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [11:58:23:INFO] [000:999] [38691] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [11:58:23:INFO] [000:999] [38691] (audio_device_module_ios.mm:600): output: 0 [11:58:23:INFO] [000:999] [38691] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:23:INFO] [000:999] [38691] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:23:INFO] [000:999] [38691] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:23:INFO] [000:999] [38691] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [11:58:23:INFO] [000:999] [38691] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [11:58:23:INFO] [000:999] [38691] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [11:58:23:INFO] [000:999] [38691] (audio_device_module_ios.mm:616): output: 0 [11:58:23:INFO] [000:999] [38691] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [11:58:23:INFO] [000:999] [38691] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [11:58:23:INFO] [000:999] [38691] (audio_device_module_ios.mm:632): output: 0 [11:58:23:INFO] [000:999] [38691] (apm_helpers.cc:62): NS set to 0 [11:58:23:INFO] [000:999] [38691] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [11:58:23:INFO] [000:999] [38691] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [11:58:23:INFO] [000:999] [38691] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [11:58:23:INFO] [000:999] [38691] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [11:58:23:INFO] [000:999] [38691] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [11:58:23:INFO] [000:999] [38691] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [11:58:23:INFO] [000:999] [38691] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [11:58:23:INFO] [000:999] [38691] (webrtc_voice_engine.cc:511): Setting AGC to 0 [11:58:23:INFO] [000:999] [38691] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [11:58:23:INFO] [000:999] [38691] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:23:INFO] [000:999] [38691] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:23:INFO] [000:999] [38691] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:23:INFO] [001:000] [38691] (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, } [11:58:23:INFO] [001:000] [38691] (channel.cc:144): Created channel for 0 [11:58:23:INFO] [001:002] [38691] (channel.cc:212): BaseChannel::Init_w, media_transport=0 [11:58:23:INFO] [001:002] [38691] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:002] [38691] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:003] [38691] (webrtc_video_engine.cc:483): CreateMediaChannel. Options: VideoOptions {} [11:58:23:INFO] [001:004] [38691] (channel.cc:144): Created channel for 1 [11:58:23:INFO] [001:004] [38691] (channel.cc:212): BaseChannel::Init_w, media_transport=0 [11:58:23:INFO] [001:004] [38691] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:004] [38691] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:004] [79375] (peer_connection.cc:4173): Session: 3593490849194371550 Old state: kStable New state: kHaveLocalOffer [11:58:23:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 1 [11:58:23:INFO] [001:004] [38691] (channel.cc:871): Setting local voice description [11:58:23:INFO] [001:005] [38691] (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}]} [11:58:23:INFO] [001:005] [38691] (webrtc_voice_engine.cc:1513): Setting receive voice codecs. [11:58:23:INFO] [001:006] [38691] (webrtc_voice_engine.cc:1784): AddSendStream: {id:ARDAMSa0;ssrcs:[2481983948];ssrc_groups:;cname:evBadRFH64bGTYuc;stream_ids:ARDAMS;} [11:58:23:INFO] [001:010] [38691] (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 [11:58:23:INFO] [001:012] [38691] (audio_coding_module.cc:333): Created [11:58:23:INFO] [001:014] [38691] (channel_send.cc:708): Not setting media_transport_ rate observers. [11:58:23:INFO] [001:015] [38691] (audio_send_stream.cc:142): AudioSendStream: 2481983948 [11:58:23:INFO] [001:015] [38691] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2481983948, extmap-allow-mixed: false, extensions: [], c_name: evBadRFH64bGTYuc}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: <unset>} [11:58:23:INFO] [001:016] [38691] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:016] [38691] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:016] [38691] (channel.cc:676): Add send stream ssrc: 2481983948 [11:58:23:INFO] [001:016] [38691] (channel.cc:863): Changing voice state, recv=0 send=0 [11:58:23:INFO] [001:017] [38691] (channel.cc:1009): Setting local video description [11:58:23:INFO] [001:017] [38691] (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}]} [11:58:23:INFO] [001:018] [38691] (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]} [11:58:23:INFO] [001:019] [38691] (webrtc_video_engine.cc:1072): AddSendStream: {id:ARDAMSv0;ssrcs:[715828121,993092192];ssrc_groups:{semantics:FID;ssrcs:[715828121,993092192]};cname:evBadRFH64bGTYuc;stream_ids:ARDAMS;} [11:58:23:INFO] [001:019] [38691] (webrtc_video_engine.cc:1111): SetLocalSsrc on all the receive streams because we added a send stream. [11:58:23:INFO] [001:019] [38691] (channel.cc:676): Add send stream ssrc: 715828121 [11:58:23:INFO] [001:019] [38691] (channel.cc:996): Changing video state, send=0 [11:58:23:INFO] [001:019] [79375] (peer_connection.cc:5512): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport. [11:58:23:INFO] [001:019] [38691] (webrtc_voice_engine.cc:1485): Setting voice channel options: AudioOptions {} [11:58:23:INFO] [001:020] [38691] (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, } [11:58:23:INFO] [001:020] [38691] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [11:58:23:INFO] [001:020] [38691] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [11:58:23:INFO] [001:020] [38691] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [11:58:23:INFO] [001:027] [38691] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [11:58:23:INFO] [001:027] [38691] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [11:58:23:INFO] [001:027] [38691] (audio_device_module_ios.mm:600): output: 0 [11:58:23:INFO] [001:027] [38691] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:23:INFO] [001:027] [38691] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:23:INFO] [001:027] [38691] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:23:INFO] [001:027] [38691] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [11:58:23:INFO] [001:027] [38691] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [11:58:23:INFO] [001:027] [38691] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [11:58:23:INFO] [001:027] [38691] (audio_device_module_ios.mm:616): output: 0 [11:58:23:INFO] [001:027] [38691] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [11:58:23:INFO] [001:027] [38691] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [11:58:23:INFO] [001:027] [38691] (audio_device_module_ios.mm:632): output: 0 [11:58:23:INFO] [001:027] [38691] (apm_helpers.cc:62): NS set to 0 [11:58:23:INFO] [001:027] [38691] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [11:58:23:INFO] [001:027] [38691] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [11:58:23:INFO] [001:027] [38691] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [11:58:23:INFO] [001:028] [38691] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [11:58:23:INFO] [001:028] [38691] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [11:58:23:INFO] [001:028] [38691] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [11:58:23:INFO] [001:028] [38691] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [11:58:23:INFO] [001:028] [38691] (webrtc_voice_engine.cc:511): Setting AGC to 0 [11:58:23:INFO] [001:028] [38691] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [11:58:23:INFO] [001:028] [38691] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:23:INFO] [001:028] [38691] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:23:INFO] [001:028] [38691] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:23:INFO] [001:028] [38691] (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, } [11:58:23:INFO] [001:029] [38691] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 715828121, options: VideoOptions {noise reduction: false, is_screencast : false, }, source = (source)) [11:58:23:INFO] [001:030] [49427] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [11:58:23:INFO] [001:030] [49427] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [11:58:23:INFO] [001:030] [6403] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [11:58:23:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 1 [11:58:23:INFO] [001:031] [49427] (basic_port_allocator.cc:108): Filtered out ignored networks: [11:58:23:INFO] [001:031] [49427] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [11:58:23:INFO] [001:031] [49427] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [11:58:23:INFO] [001:031] [6403] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [11:58:23:INFO] [001:031] [49427] (basic_port_allocator.cc:851): Network manager has started [11:58:23:INFO] [001:031] [49427] (basic_port_allocator.cc:108): Filtered out ignored networks: [11:58:23:INFO] [001:031] [49427] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [11:58:23:INFO] [001:031] [49427] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [11:58:23:INFO] [001:031] [49427] (basic_port_allocator.cc:851): Network manager has started [11:58:23:INFO] [001:031] [49427] (basic_port_allocator.cc:108): Filtered out ignored networks: [11:58:23:INFO] [001:031] [49427] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [11:58:23:INFO] [001:031] [49427] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [11:58:23:INFO] [001:031] [49427] (basic_port_allocator.cc:766): Allocate ports on 1 networks [11:58:23:INFO] [001:032] [49427] (basic_port_allocator.cc:108): Filtered out ignored networks: [11:58:23:INFO] [001:032] [49427] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [11:58:23:INFO] [001:032] [49427] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [11:58:23:INFO] [001:032] [49427] (basic_port_allocator.cc:766): Allocate ports on 1 networks [11:58:23:INFO] [001:032] [49427] (basic_port_allocator.cc:1318): Net[en0:192.168.148.0/23:Wifi:id=1]: Allocation Phase=Udp [11:58:23:INFO] [001:033] [49427] (port.cc:320): Port[2405b000::1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port created with network cost 10 [11:58:23:INFO] [001:033] [49427] (basic_port_allocator.cc:875): Adding allocated port for 0 [11:58:23:INFO] [001:033] [49427] (basic_port_allocator.cc:894): Port[2405b000:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Added port to allocator [11:58:23:INFO] [001:034] [31499] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [11:58:23:INFO] [001:034] [31499] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [11:58:23:INFO] [001:034] [31499] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [11:58:23:INFO] [001:034] [49427] (basic_port_allocator.cc:912): Port[2405b000:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Gathered candidate: Cand[:1955044956:1:udp:2122260223:192.168.148.41:61495:local::0:YLj5:/j0o5yoRFRv8u1m0LFUsi0dh:1:10:0] [11:58:23:INFO] [001:034] [31499] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [11:58:23:INFO] [001:034] [31499] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [11:58:23:INFO] [001:034] [31499] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [11:58:23:INFO] [001:034] [49427] (basic_port_allocator.cc:940): Port[2405b000:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port ready. [11:58:23:INFO] [001:034] [31499] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [11:58:23:INFO] [001:034] [49427] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [11:58:23:INFO] [001:034] [31499] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0 [11:58:23:INFO] [001:034] [31499] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 225467e0 [11:58:23:INFO] [001:034] [49427] (p2p_transport_channel.cc:799): Port[2405b000:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: SetOption(5, 0) failed: 0 [11:58:23:INFO] [001:034] [31499] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [11:58:23:INFO] [001:034] [31499] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [11:58:23:INFO] [001:034] [31499] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [11:58:23:INFO] [001:034] [31499] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [11:58:23:INFO] [001:035] [49427] (basic_port_allocator.cc:1017): Port[2405b000:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port completed gathering candidates. [11:58:23:INFO] [001:035] [31499] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [11:58:23:INFO] [001:035] [31499] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [11:58:23:INFO] [001:035] [31499] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [11:58:23:INFO] [001:035] [49427] (basic_port_allocator.cc:1318): Net[en0:192.168.148.0/23:Wifi:id=1]: Allocation Phase=Udp [11:58:23:INFO] [001:035] [49427] (port.cc:320): Port[238b0a00::1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port created with network cost 10 [11:58:23:INFO] [001:035] [31499] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 1 transport 0 [11:58:23:INFO] [001:035] [31499] (dtls_srtp_transport.cc:66): Setting RTP Transport on 1 transport 225470a0 [11:58:23:INFO] [001:035] [49427] (basic_port_allocator.cc:875): Adding allocated port for 1 [11:58:23:INFO] [001:035] [49427] (basic_port_allocator.cc:894): Port[238b0a00:1:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Added port to allocator [11:58:23:INFO] [001:035] [31499] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=YLj5, renomination enabled [11:58:23:INFO] [001:036] [49427] (basic_port_allocator.cc:912): Port[238b0a00:1:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Gathered candidate: Cand[:1955044956:1:udp:2122260223:192.168.148.41:50762:local::0:YLj5:/j0o5yoRFRv8u1m0LFUsi0dh:1:10:0] [11:58:23:INFO] [001:036] [49427] (basic_port_allocator.cc:940): Port[238b0a00:1:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port ready. [11:58:23:INFO] [001:036] [49427] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [11:58:23:INFO] [001:036] [49427] (p2p_transport_channel.cc:799): Port[238b0a00:1:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: SetOption(5, 0) failed: 0 [11:58:23:INFO] [001:036] [31499] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=YLj5, renomination enabled [11:58:23:INFO] [001:036] [49427] (basic_port_allocator.cc:1017): Port[238b0a00:1:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port completed gathering candidates. [11:58:23:INFO] [001:036] [50183] (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, } [11:58:23:INFO] [001:037] [50183] (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, } [11:58:23:INFO] [001:037] [50183] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [11:58:23:INFO] [001:037] [50183] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [11:58:23:INFO] [001:037] [50183] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [11:58:23:INFO] [001:037] [50183] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [11:58:23:INFO] [001:037] [50183] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [11:58:23:INFO] [001:037] [50183] (audio_device_module_ios.mm:600): output: 0 [11:58:23:INFO] [001:037] [50183] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:23:INFO] [001:037] [50183] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:23:INFO] [001:037] [50183] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:23:INFO] [001:037] [50183] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [11:58:23:INFO] [001:037] [50183] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [11:58:23:INFO] [001:037] [50183] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [11:58:23:INFO] [001:037] [50183] (audio_device_module_ios.mm:616): output: 0 [11:58:23:INFO] [001:037] [50183] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [11:58:23:INFO] [001:037] [50183] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [11:58:23:INFO] [001:037] [50183] (audio_device_module_ios.mm:632): output: 0 [11:58:23:INFO] [001:037] [50183] (apm_helpers.cc:62): NS set to 0 [11:58:23:INFO] [001:037] [50183] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [11:58:23:INFO] [001:037] [50183] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [11:58:23:INFO] [001:038] [50183] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [11:58:23:INFO] [001:038] [50183] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [11:58:23:INFO] [001:038] [50183] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [11:58:23:INFO] [001:038] [50183] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [11:58:23:INFO] [001:038] [50183] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [11:58:23:INFO] [001:038] [50183] (webrtc_voice_engine.cc:511): Setting AGC to 0 [11:58:23:INFO] [001:038] [50183] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [11:58:23:INFO] [001:038] [50183] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:23:INFO] [001:038] [50183] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:23:INFO] [001:038] [50183] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:23:INFO] [001:038] [50183] (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, } [11:58:23:INFO] [001:038] [50183] (channel.cc:144): Created channel for 0 [11:58:23:INFO] [001:039] [50183] (channel.cc:212): BaseChannel::Init_w, media_transport=0 [11:58:23:INFO] [001:039] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:039] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:039] [50183] (webrtc_video_engine.cc:483): CreateMediaChannel. Options: VideoOptions {} [11:58:23:INFO] [001:039] [50183] (channel.cc:144): Created channel for 1 [11:58:23:INFO] [001:040] [50183] (channel.cc:212): BaseChannel::Init_w, media_transport=0 [11:58:23:INFO] [001:040] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:040] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:040] [79107] (peer_connection.cc:4173): Session: 4729406347323590376 Old state: kStable New state: kHaveRemoteOffer [11:58:23:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 3 [11:58:23:INFO] [001:040] [50183] (channel.cc:923): Setting remote voice description [11:58:23:INFO] [001:040] [50183] (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 {}} [11:58:23:INFO] [001:041] [50183] (webrtc_voice_engine.cc:1702): Recreate all the receive streams because the send codec has changed. [11:58:23:INFO] [001:041] [50183] (webrtc_voice_engine.cc:2163): WebRtcVoiceMediaChannel::SetMaxSendBitrate. [11:58:23:INFO] [001:041] [50183] (webrtc_voice_engine.cc:1485): Setting voice channel options: AudioOptions {} [11:58:23:INFO] [001:041] [50183] (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, } [11:58:23:INFO] [001:041] [50183] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [11:58:23:INFO] [001:041] [50183] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [11:58:23:INFO] [001:041] [50183] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [11:58:23:INFO] [001:041] [50183] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [11:58:23:INFO] [001:041] [50183] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [11:58:23:INFO] [001:041] [50183] (audio_device_module_ios.mm:600): output: 0 [11:58:23:INFO] [001:041] [50183] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:23:INFO] [001:041] [50183] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:23:INFO] [001:041] [50183] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:23:INFO] [001:041] [50183] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [11:58:23:INFO] [001:041] [50183] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [11:58:23:INFO] [001:041] [50183] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [11:58:23:INFO] [001:041] [50183] (audio_device_module_ios.mm:616): output: 0 [11:58:23:INFO] [001:041] [50183] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [11:58:23:INFO] [001:041] [50183] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [11:58:23:INFO] [001:041] [50183] (audio_device_module_ios.mm:632): output: 0 [11:58:23:INFO] [001:042] [50183] (apm_helpers.cc:62): NS set to 0 [11:58:23:INFO] [001:042] [50183] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [11:58:23:INFO] [001:042] [50183] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [11:58:23:INFO] [001:042] [50183] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [11:58:23:INFO] [001:042] [50183] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [11:58:23:INFO] [001:042] [50183] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [11:58:23:INFO] [001:042] [50183] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [11:58:23:INFO] [001:042] [50183] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [11:58:23:INFO] [001:042] [50183] (webrtc_voice_engine.cc:511): Setting AGC to 0 [11:58:23:INFO] [001:042] [50183] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [11:58:23:INFO] [001:042] [50183] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:23:INFO] [001:042] [50183] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:23:INFO] [001:042] [50183] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:23:INFO] [001:042] [50183] (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, } [11:58:23:INFO] [001:042] [50183] (webrtc_voice_engine.cc:1849): AddRecvStream: {id:ARDAMSa0;ssrcs:[2481983948];ssrc_groups:;cname:evBadRFH64bGTYuc;stream_ids:ARDAMS;} [11:58:23:INFO] [001:043] [50183] (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 [11:58:23:INFO] [001:043] [50183] (audio_coding_module.cc:333): Created [11:58:23:INFO] [001:043] [50183] (audio_receive_stream.cc:117): AudioReceiveStream: 2481983948 [11:58:23:INFO] [001:043] [50183] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2481983948, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS} [11:58:23:INFO] [001:044] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:044] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:044] [50183] (channel.cc:719): Add remote ssrc: 2481983948 [11:58:23:INFO] [001:044] [50183] (channel.cc:863): Changing voice state, recv=0 send=0 [11:58:23:INFO] [001:044] [50183] (channel.cc:1061): Setting remote video description [11:58:23:INFO] [001:044] [50183] (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} [11:58:23:INFO] [001:045] [50183] (webrtc_video_engine.cc:690): Using codec: VideoCodec[96:H264] [11:58:23:INFO] [001:045] [50183] (webrtc_video_engine.cc:758): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. [11:58:23:INFO] [001:045] [50183] (webrtc_video_engine.cc:1174): AddRecvStream: {id:ARDAMSv0;ssrcs:[715828121,993092192];ssrc_groups:{semantics:FID;ssrcs:[715828121,993092192]};cname:evBadRFH64bGTYuc;stream_ids:ARDAMS;} [11:58:23:INFO] [001:052] [50183] (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: 715828121, 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: 993092192, 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} [11:58:23:INFO] [001:052] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:052] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:055] [50183] (channel.cc:719): Add remote ssrc: 715828121 [11:58:23:INFO] [001:055] [50183] (channel.cc:996): Changing video state, send=0 [11:58:23:INFO] [001:056] [79107] (peer_connection.cc:5512): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport. [11:58:23:INFO] [001:056] [79107] (peer_connection.cc:2745): Processing the MSIDs for MID=0 (streams=[ARDAMS]). [11:58:23:INFO] [001:056] [79107] (peer_connection.cc:2756): Processing the addition of a remote track for MID=0. [11:58:23:INFO] [001:056] [50183] (webrtc_voice_engine.cc:1962): SetOutputVolume() to 1 for recv stream with ssrc 2481983948 [11:58:23:INFO] [001:056] [79107] (peer_connection.cc:2745): Processing the MSIDs for MID=1 (streams=[ARDAMS]). [11:58:23:INFO] [001:056] [79107] (peer_connection.cc:2756): Processing the addition of a remote track for MID=1. [11:58:23:INFO] [001:057] [50183] (webrtc_video_engine.cc:1304): SetSink: ssrc:715828121 (ptr) [11:58:23:INFO] [001:057] [79107] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60000214df00): created DTMF sender: RTCDtmfSender { [11:58:23:INFO] remainingTones: [11:58:23:INFO] duration: 0.100000 sec [11:58:23:INFO] interToneGap: 0.050000 sec [11:58:23:INFO] } [11:58:23:INFO] [001:057] [79107] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600002375d00): created sender: RTCRtpSender { [11:58:23:INFO] senderId: ARDAMSa0 [11:58:23:INFO] } [11:58:23:INFO] [001:057] [79107] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x600002d6a940): created receiver: RTCRtpReceiver { [11:58:23:INFO] receiverId: f51d84c6-6f26-4e91-bbb8-d9da7e44e00d [11:58:23:INFO] } [11:58:23:INFO] [001:057] [79107] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x600002d69f50): created transceiver: RTCRtpTransceiver { [11:58:23:INFO] sender: RTCRtpSender { [11:58:23:INFO] senderId: ARDAMSa0 [11:58:23:INFO] } [11:58:23:INFO] receiver: RTCRtpReceiver { [11:58:23:INFO] receiverId: f51d84c6-6f26-4e91-bbb8-d9da7e44e00d [11:58:23:INFO] } [11:58:23:INFO] } [11:58:23:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving audio on track f51d84c6-6f26-4e91-bbb8-d9da7e44e00d. [11:58:23:INFO] [001:058] [79107] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [11:58:23:INFO] [001:058] [79107] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600002375d60): created sender: RTCRtpSender { [11:58:23:INFO] senderId: ARDAMSv0 [11:58:23:INFO] } [11:58:23:INFO] [001:058] [79107] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x600002d6aaf0): created receiver: RTCRtpReceiver { [11:58:23:INFO] receiverId: caa41d84-778a-4a49-9428-e662419ae72f [11:58:23:INFO] } [11:58:23:INFO] [001:058] [79107] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x600002d69f50): created transceiver: RTCRtpTransceiver { [11:58:23:INFO] sender: RTCRtpSender { [11:58:23:INFO] senderId: ARDAMSv0 [11:58:23:INFO] } [11:58:23:INFO] receiver: RTCRtpReceiver { [11:58:23:INFO] receiverId: caa41d84-778a-4a49-9428-e662419ae72f [11:58:23:INFO] } [11:58:23:INFO] } [11:58:23:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving video on track caa41d84-778a-4a49-9428-e662419ae72f. [11:58:23:INFO] (RTCLogging.mm:33): (ARDAppClient.m:381 -[ARDAppClient peerConnection:didAddStream:]): Stream with 1 video tracks and 1 audio tracks was added. [11:58:23:INFO] [001:059] [6403] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60000214df40): created DTMF sender: RTCDtmfSender { [11:58:23:INFO] remainingTones: [11:58:23:INFO] duration: 0.100000 sec [11:58:23:INFO] interToneGap: 0.050000 sec [11:58:23:INFO] } [11:58:23:INFO] [001:059] [6403] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600002375d80): created sender: RTCRtpSender { [11:58:23:INFO] senderId: ARDAMSa0 [11:58:23:INFO] } [11:58:23:INFO] [001:059] [79375] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [11:58:23:INFO] [001:059] [6403] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600002372940): created sender: RTCRtpSender { [11:58:23:INFO] senderId: ARDAMSv0 [11:58:23:INFO] } [11:58:23:INFO] [001:061] [79107] (peer_connection.cc:6210): 0 is not ready to use the remote candidate because the local or remote description is not set. [11:58:23:INFO] [001:062] [79107] (peer_connection.cc:6210): 1 is not ready to use the remote candidate because the local or remote description is not set. [11:58:23:INFO] [001:062] [79107] (peer_connection.cc:2110): CreateAnswer: offer_to_receive_audio is not supported with Unified Plan semantics. Use the RtpTransceiver API instead. [11:58:23:INFO] [001:062] [79107] (peer_connection.cc:2115): CreateAnswer: offer_to_receive_video is not supported with Unified Plan semantics. Use the RtpTransceiver API instead. [11:58:23:INFO] [001:062] [79107] (peer_connection.cc:5540): Local and Remote descriptions must be applied to get the SSL Role of the session. [11:58:23:INFO] [001:062] [79107] (peer_connection.cc:5540): Local and Remote descriptions must be applied to get the SSL Role of the session. [11:58:23:INFO] [001:065] [6403] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [11:58:23:INFO] [001:066] [6403] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [11:58:23:INFO] [001:068] [31499] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0 [11:58:23:INFO] [001:068] [31499] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 225467e0 [11:58:23:INFO] [001:068] [31499] (p2p_transport_channel.cc:432): Set ICE ufrag: /vWc pwd: HeeYpJWpsSZFEK6YTE0P3Nmv on transport 0 [11:58:23:INFO] [001:069] [31499] (dtls_transport.cc:370): DtlsTransport[0|1|__]: DTLS setup complete. [11:58:23:INFO] [001:069] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:069] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:069] [79107] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv. [11:58:23:INFO] [001:069] [79107] (rtp_transceiver.cc:169): Changing transceiver (MID=1) current direction from <not set> to kSendRecv. [11:58:23:INFO] [001:069] [50183] (channel.cc:553): Channel enabled [11:58:23:INFO] [001:069] [50183] (channel.cc:863): Changing voice state, recv=0 send=0 [11:58:23:INFO] [001:070] [50183] (channel.cc:553): Channel enabled [11:58:23:INFO] [001:070] [50183] (channel.cc:996): Changing video state, send=0 [11:58:23:INFO] [001:070] [79107] (peer_connection.cc:4173): Session: 4729406347323590376 Old state: kHaveRemoteOffer New state: kStable [11:58:23:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 0 [11:58:23:INFO] [001:070] [50183] (channel.cc:871): Setting local voice description [11:58:23:INFO] [001:070] [50183] (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}]} [11:58:23:INFO] [001:070] [50183] (webrtc_voice_engine.cc:1513): Setting receive voice codecs. [11:58:23:INFO] [001:070] [50183] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2481983948, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS} [11:58:23:INFO] [001:070] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:070] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:070] [50183] (audio_receive_stream.cc:140): ~AudioReceiveStream: 2481983948 [11:58:23:INFO] [001:071] [50183] (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 [11:58:23:INFO] [001:071] [50183] (audio_coding_module.cc:333): Created [11:58:23:INFO] [001:071] [50183] (audio_receive_stream.cc:117): AudioReceiveStream: 2481983948 [11:58:23:INFO] [001:071] [50183] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2481983948, 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} [11:58:23:INFO] [001:071] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:071] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:071] [50183] (webrtc_voice_engine.cc:1784): AddSendStream: {id:ARDAMSa0;ssrcs:[3459412627];ssrc_groups:;cname:QRMy96WNfUpu1Re7;stream_ids:ARDAMS;} [11:58:23:INFO] [001:072] [50183] (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 [11:58:23:INFO] [001:072] [50183] (audio_coding_module.cc:333): Created [11:58:23:INFO] [001:072] [50183] (channel_send.cc:708): Not setting media_transport_ rate observers. [11:58:23:INFO] [001:072] [50183] (audio_send_stream.cc:142): AudioSendStream: 3459412627 [11:58:23:INFO] [001:072] [50183] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 3459412627, 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: QRMy96WNfUpu1Re7}, 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}}}} [11:58:23:INFO] [001:075] [50183] (audio_encoder_opus.cc:758): Set Opus bitrate to 32000 bps. [11:58:23:INFO] [001:075] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:075] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:075] [50183] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2481983948, local_ssrc: 3459412627, 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} [11:58:23:INFO] [001:075] [50183] (channel.cc:676): Add send stream ssrc: 3459412627 [11:58:23:INFO] [001:076] [50183] (audio_device_module_ios.mm:543): Playing [11:58:23:INFO] [001:076] [50183] (audio_device_module_ios.mm:480): InitPlayout [11:58:23:INFO] [001:076] [50183] (audio_device_module_ios.mm:506): PlayoutIsInitialized [11:58:23:INFO] [001:076] [50183] (audio_device_ios.mm:185): AudioDeviceIOS::InitPlayout [11:58:23:INFO] [001:076] [50183] (audio_device_ios.mm:859): AudioDeviceIOS::InitPlayOrRecord [11:58:23:INFO] [001:086] [49427] (basic_port_allocator.cc:1318): Net[en0:192.168.148.0/23:Wifi:id=1]: Allocation Phase=Relay [11:58:23:INFO] [001:086] [49427] (basic_port_allocator.cc:1318): Net[en0:192.168.148.0/23:Wifi:id=1]: Allocation Phase=Relay [11:58:23:INFO] [001:114] [50183] (RTCLogging.mm:33): (RTCAudioSession.mm:99 -[RTCAudioSession initWithAudioSession:]): RTCAudioSession (0x60000115b2a0): init. [11:58:23:INFO] [001:114] [50183] (RTCLogging.mm:33): (audio_device_ios.mm:824 ConfigureAudioSession): Configuring audio session. [11:58:23:INFO] [001:114] [50183] (RTCLogging.mm:33): (RTCAudioSession.mm:699 -[RTCAudioSession configureWebRTCSession:]): Configuring audio session for WebRTC. [11:58:23:INFO] [001:114] [50183] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:64 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set category to: AVAudioSessionCategoryPlayAndRecord [11:58:23:INFO] [001:114] [50183] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:75 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set mode to: AVAudioSessionModeVoiceChat [11:58:23:INFO] [001:114] [50183] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:103 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred sample rate to: 48000.00 [11:58:23:INFO] [001:114] [50183] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:116 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred IO buffer duration to: 0.020000 [11:58:23:INFO] [001:114] [50183] (RTCLogging.mm:33): (RTCAudioSession.mm:624 -[RTCAudioSession incrementActivationCount]): Incrementing activation count. [11:58:23:INFO] [001:115] [50183] (RTCLogging.mm:33): (RTCAudioSession.mm:377 -[RTCAudioSession setActive:error:]): Number of current activations: 1 [11:58:23:INFO] [001:115] [50183] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:145 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set input number of channels to: 1 [11:58:23:INFO] [001:115] [50183] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:158 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set output number of channels to: 1 [11:58:23:INFO] [001:115] [50183] (RTCLogging.mm:33): (audio_device_ios.mm:835 ConfigureAudioSession): Configured audio session. [11:58:23:INFO] [001:115] [50183] (audio_device_ios.mm:678): AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession [11:58:23:INFO] [001:115] [50183] (RTCLogging.mm:33): (audio_device_ios.mm:683 SetupAudioBuffersForActiveAudioSession): RTCAudioSession: { [11:58:23:INFO] category: AVAudioSessionCategoryPlayAndRecord [11:58:23:INFO] categoryOptions: 4 [11:58:23:INFO] mode: AVAudioSessionModeVoiceChat [11:58:23:INFO] isActive: 1 [11:58:23:INFO] sampleRate: 48000.00 [11:58:23:INFO] IOBufferDuration: 0.010667 [11:58:23:INFO] outputNumberOfChannels: 2 [11:58:23:INFO] inputNumberOfChannels: 2 [11:58:23:INFO] outputLatency: 0.010000 [11:58:23:INFO] inputLatency: 0.010000 [11:58:23:INFO] outputVolume: 0.600000 [11:58:23:INFO] } [11:58:23:INFO] [001:115] [50183] (audio_device_ios.mm:714): frames per I/O buffer: 512 [11:58:23:INFO] [001:115] [50183] (audio_device_ios.mm:715): bytes per I/O buffer: 1024 [11:58:23:INFO] [001:115] [50183] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer [11:58:23:INFO] [001:115] [50183] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000) [11:58:23:INFO] [001:115] [50183] (audio_device_buffer.cc:206): SetPlayoutChannels(1) [11:58:23:INFO] [001:115] [50183] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000) [11:58:23:INFO] [001:115] [50183] (audio_device_buffer.cc:200): SetRecordingChannels(1) [11:58:23:INFO] [001:115] [50183] (fine_audio_buffer.cc:32): FineAudioBuffer [11:58:23:INFO] [001:115] [50183] (fine_audio_buffer.cc:34): playout_samples_per_channel_10ms: 480 [11:58:23:INFO] [001:115] [50183] (fine_audio_buffer.cc:36): playout_channels: 1 [11:58:23:INFO] [001:115] [50183] (fine_audio_buffer.cc:39): record_samples_per_channel_10ms: 480 [11:58:23:INFO] [001:115] [50183] (fine_audio_buffer.cc:41): record_channels: 1 [11:58:23:INFO] [001:115] [50183] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:196 Initialize): Initializing audio unit with sample rate: 48000.000000 [11:58:23:INFO] [001:115] [50183] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:43 LogStreamDescription): AudioStreamBasicDescription: { [11:58:23:INFO] mSampleRate: 48000.00 [11:58:23:INFO] formatIDString: lpcm [11:58:23:INFO] mFormatFlags: 0xC [11:58:23:INFO] mBytesPerPacket: 2 [11:58:23:INFO] mFramesPerPacket: 1 [11:58:23:INFO] mBytesPerFrame: 2 [11:58:23:INFO] mChannelsPerFrame: 1 [11:58:23:INFO] mBitsPerChannel: 16 [11:58:23:INFO] mReserved: 28672 [11:58:23:INFO] } [11:58:23:INFO] [001:138] [49427] (basic_port_allocator.cc:1318): Net[en0:192.168.148.0/23:Wifi:id=1]: Allocation Phase=Tcp [11:58:23:INFO] [001:139] [49427] (port.cc:320): Port[240a3c00::1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port created with network cost 10 [11:58:23:INFO] [001:139] [49427] (basic_port_allocator.cc:875): Adding allocated port for 0 [11:58:23:INFO] [001:139] [49427] (basic_port_allocator.cc:894): Port[240a3c00:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Added port to allocator [11:58:23:INFO] [001:139] [49427] (basic_port_allocator.cc:912): Port[240a3c00:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Gathered candidate: Cand[:973365932:1:tcp:1518280447:192.168.148.41:51832:local::0:YLj5:/j0o5yoRFRv8u1m0LFUsi0dh:1:10:0] [11:58:23:INFO] [001:139] [49427] (basic_port_allocator.cc:940): Port[240a3c00:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port ready. [11:58:23:INFO] [001:139] [49427] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [11:58:23:INFO] [001:139] [49427] (p2p_transport_channel.cc:799): Port[240a3c00:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: SetOption(5, 0) failed: 0 [11:58:23:INFO] [001:139] [49427] (basic_port_allocator.cc:1017): Port[240a3c00:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port completed gathering candidates. [11:58:23:INFO] [001:140] [49427] (basic_port_allocator.cc:1119): All candidates gathered for 0:1:0 [11:58:23:INFO] [001:140] [49427] (p2p_transport_channel.cc:854): P2PTransportChannel: 0, component 1 gathering complete [11:58:23:INFO] [001:140] [49427] (basic_port_allocator.cc:1318): Net[en0:192.168.148.0/23:Wifi:id=1]: Allocation Phase=Tcp [11:58:23:INFO] [001:140] [49427] (port.cc:320): Port[23036e00::1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port created with network cost 10 [11:58:23:INFO] [001:140] [49427] (basic_port_allocator.cc:875): Adding allocated port for 1 [11:58:23:INFO] [001:140] [49427] (basic_port_allocator.cc:894): Port[23036e00:1:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Added port to allocator [11:58:23:INFO] [001:140] [49427] (basic_port_allocator.cc:912): Port[23036e00:1:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Gathered candidate: Cand[:973365932:1:tcp:1518280447:192.168.148.41:51833:local::0:YLj5:/j0o5yoRFRv8u1m0LFUsi0dh:1:10:0] [11:58:23:INFO] [001:141] [49427] (basic_port_allocator.cc:940): Port[23036e00:1:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port ready. [11:58:23:INFO] [001:141] [49427] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [11:58:23:INFO] [001:141] [49427] (p2p_transport_channel.cc:799): Port[23036e00:1:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: SetOption(5, 0) failed: 0 [11:58:23:INFO] [001:141] [49427] (basic_port_allocator.cc:1017): Port[23036e00:1:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port completed gathering candidates. [11:58:23:INFO] [001:141] [49427] (basic_port_allocator.cc:1119): All candidates gathered for 1:1:0 [11:58:23:INFO] [001:141] [49427] (p2p_transport_channel.cc:854): P2PTransportChannel: 1, component 1 gathering complete [11:58:23:INFO] [001:141] [50183] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:250 Initialize): Voice Processing I/O unit is now initialized. [11:58:23:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 2 [11:58:23:INFO] [001:264] [50183] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:71 GetAGCState): VPIO unit AGC: 0 [11:58:23:INFO] [001:264] [50183] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:264 Initialize): Failed to get AGC state (1st attempt). Error=-66631. [11:58:23:INFO] [001:264] [50183] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:302 Initialize): WebRTC.Audio.BuiltInAGCWasEnabledByDefault: 0 [11:58:23:INFO] [001:264] [50183] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:308 Initialize): WebRTC.Audio.BuiltInAGCIsEnabled: 0 [11:58:23:INFO] [001:264] [50183] (audio_device_module_ios.mm:486): output: 0 [11:58:23:INFO] [001:264] [50183] (audio_device_module_ios.mm:518): StartPlayout [11:58:23:INFO] [001:264] [50183] (audio_device_module_ios.mm:543): Playing [11:58:23:INFO] [001:264] [50183] (audio_device_buffer.cc:98): StartPlayout [11:58:23:INFO] [001:264] [50183] (audio_device_ios.mm:227): AudioDeviceIOS::StartPlayout [11:58:23:INFO] [001:264] [50183] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:316 Start): Starting audio unit. [11:58:23:INFO] [001:274] [50183] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:320 Start): Failed to start audio unit. Error=-66637 [11:58:23:INFO] [001:274] [50183] (RTCLogging.mm:33): (audio_device_ios.mm:237 StartPlayout): StartPlayout failed to start audio unit. [11:58:23:INFO] [001:274] [50183] (audio_device_module_ios.mm:525): output: -1 [11:58:23:INFO] [001:274] [50183] (channel.cc:863): Changing voice state, recv=1 send=0 [11:58:23:INFO] [001:274] [50183] (message_queue.cc:517): Message took 204ms to dispatch. Posted from: SetLocalContent@../../pc/channel.cc:293 [11:58:23:INFO] [001:274] [50183] (channel.cc:1009): Setting local video description [11:58:23:INFO] [001:275] [50183] (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}]} [11:58:23:INFO] [001:275] [50183] (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]} [11:58:23:INFO] [001:276] [50183] (webrtc_video_engine.cc:2542): MaybeRecreateWebRtcFlexfecStream (recv) because of SetRecvParameters [11:58:23:INFO] [001:276] [50183] (webrtc_video_engine.cc:2547): RecreateWebRtcVideoStream (recv) because of SetRecvParameters [11:58:23:INFO] [001:276] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:276] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:276] [50183] (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: 715828121, 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: 993092192, 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} [11:58:23:INFO] [001:276] [50183] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [11:58:23:INFO] [001:277] [50183] (receive_statistics_proxy.cc:487): Frames decoded 0 [11:58:23:INFO] WebRTC.Video.DroppedFrames.Receiver 0 [11:58:23:INFO] [11:58:23:INFO] [001:278] [50183] (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: 715828121, 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: 993092192, 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} [11:58:23:INFO] [001:278] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:278] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:278] [50183] (webrtc_video_engine.cc:1072): AddSendStream: {id:ARDAMSv0;ssrcs:[2757358605,2295916538];ssrc_groups:{semantics:FID;ssrcs:[2757358605,2295916538]};cname:QRMy96WNfUpu1Re7;stream_ids:ARDAMS;} [11:58:23:INFO] [001:279] [50183] (webrtc_video_engine.cc:1883): RecreateWebRtcStream (send) because of SetCodec. [11:58:23:INFO] [001:280] [50183] (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 [11:58:23:INFO] [001:281] [11011] (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. [11:58:23:INFO] [001:281] [11011] (video_send_stream_impl.cc:250): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [2757358605], 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: [2295916538], payload_type: 97}, c_name: QRMy96WNfUpu1Re7}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [11:58:23:INFO] [001:281] [11011] (video_send_stream_impl.cc:278): ERROR: Initial encoder max bitrate = -1 which is <= 0! [11:58:23:INFO] [001:281] [11011] (video_send_stream_impl.cc:280): Using default encoder max bitrate = 10 Mbps [11:58:23:INFO] [001:282] [11011] (video_stream_encoder.cc:628): ConfigureEncoder requested. [11:58:23:INFO] [001:282] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:282] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:282] [50183] (video_send_stream.cc:160): VideoSendStream::Stop [11:58:23:INFO] [001:282] [50183] (webrtc_video_engine.cc:1111): SetLocalSsrc on all the receive streams because we added a send stream. [11:58:23:INFO] [001:282] [50183] (webrtc_video_engine.cc:2483): RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc=2757358605 [11:58:23:INFO] [001:282] [11011] (video_send_stream_impl.cc:417): VideoSendStream::Stop [11:58:23:INFO] [001:282] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:282] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:282] [50183] (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: 715828121, 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: 993092192, 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} [11:58:23:INFO] [001:282] [50183] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [11:58:23:INFO] [001:283] [50183] (receive_statistics_proxy.cc:487): Frames decoded 0 [11:58:23:INFO] WebRTC.Video.DroppedFrames.Receiver 0 [11:58:23:INFO] [11:58:23:INFO] [001:283] [50183] (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: 715828121, local_ssrc: 2757358605, 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: 993092192, 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} [11:58:23:INFO] [001:283] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:23:INFO] [001:283] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:23:INFO] [001:284] [50183] (channel.cc:676): Add send stream ssrc: 2757358605 [11:58:23:INFO] [001:284] [50183] (video_send_stream.cc:160): VideoSendStream::Stop [11:58:23:INFO] [001:284] [50183] (channel.cc:996): Changing video state, send=0 [11:58:23:INFO] [001:284] [11011] (video_send_stream_impl.cc:417): VideoSendStream::Stop [11:58:23:INFO] [001:284] [79107] (peer_connection.cc:4109): Changing IceConnectionState 0 => 1 [11:58:23:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 1 [11:58:23:INFO] [001:284] [31499] (jsep_transport_controller.cc:318): Not adding candidate because the JsepTransport doesn't exist. Ignore it. [11:58:23:INFO] [001:284] [79107] (peer_connection.cc:5518): Non-rejected SCTP m= section is needed to get the SSL Role of the SCTP transport. [11:58:23:INFO] [001:284] [50183] (webrtc_voice_engine.cc:1485): Setting voice channel options: AudioOptions {} [11:58:23:INFO] [001:284] [50183] (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, } [11:58:23:INFO] [001:284] [50183] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [11:58:23:INFO] [001:284] [50183] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [11:58:23:INFO] [001:284] [50183] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [11:58:23:INFO] [001:284] [50183] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [11:58:23:INFO] [001:284] [50183] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [11:58:23:INFO] [001:284] [50183] (audio_device_module_ios.mm:600): output: 0 [11:58:23:INFO] [001:284] [50183] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:23:INFO] [001:285] [50183] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:23:INFO] [001:285] [50183] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:23:INFO] [001:285] [50183] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [11:58:23:INFO] [001:285] [50183] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [11:58:23:INFO] [001:285] [50183] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [11:58:23:INFO] [001:285] [50183] (audio_device_module_ios.mm:616): output: 0 [11:58:23:INFO] [001:285] [50183] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [11:58:23:INFO] [001:285] [50183] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [11:58:23:INFO] [001:285] [50183] (audio_device_module_ios.mm:632): output: 0 [11:58:23:INFO] [001:285] [50183] (apm_helpers.cc:62): NS set to 0 [11:58:23:INFO] [001:285] [50183] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [11:58:23:INFO] [001:285] [50183] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [11:58:23:INFO] [001:285] [50183] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [11:58:23:INFO] [001:285] [50183] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [11:58:23:INFO] [001:285] [50183] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [11:58:23:INFO] [001:285] [50183] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [11:58:23:INFO] [001:285] [50183] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [11:58:23:INFO] [001:285] [50183] (webrtc_voice_engine.cc:511): Setting AGC to 0 [11:58:23:INFO] [001:285] [50183] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [11:58:23:INFO] [001:285] [50183] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:23:INFO] [001:285] [50183] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:23:INFO] [001:285] [50183] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:23:INFO] [001:285] [50183] (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, } [11:58:23:INFO] [001:285] [50183] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 2757358605, options: VideoOptions {noise reduction: false, is_screencast : false, }, source = (source)) [11:58:23:INFO] [001:286] [11011] (video_stream_encoder.cc:628): ConfigureEncoder requested. [11:58:23:INFO] [001:286] [31499] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [11:58:23:INFO] [001:286] [31499] (basic_port_allocator.cc:108): Filtered out ignored networks: [11:58:23:INFO] [001:286] [31499] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [11:58:23:INFO] [001:286] [31499] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [11:58:23:INFO] [001:286] [31499] (basic_port_allocator.cc:851): Network manager has started [11:58:23:INFO] [001:286] [31499] (basic_port_allocator.cc:108): Filtered out ignored networks: [11:58:23:INFO] [001:286] [31499] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [11:58:23:INFO] [001:286] [31499] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [11:58:23:INFO] [001:286] [31499] (basic_port_allocator.cc:766): Allocate ports on 1 networks [11:58:23:INFO] [001:287] [31499] (basic_port_allocator.cc:1318): Net[en0:192.168.148.0/23:Wifi:id=1]: Allocation Phase=Udp [11:58:23:INFO] [001:287] [31499] (port.cc:320): Port[240a3200::1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port created with network cost 10 [11:58:23:INFO] [001:287] [31499] (basic_port_allocator.cc:875): Adding allocated port for 0 [11:58:23:INFO] [001:287] [31499] (basic_port_allocator.cc:894): Port[240a3200:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Added port to allocator [11:58:23:INFO] [001:287] [79107] (message_queue.cc:517): Message took 220ms to dispatch. Posted from: SetLocalDescription@../../api/peer_connection_proxy.h:100 [11:58:23:INFO] [001:287] [31499] (basic_port_allocator.cc:912): Port[240a3200:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Gathered candidate: Cand[:1955044956:1:udp:2122260223:192.168.148.41:64025:local::0:/vWc:HeeYpJWpsSZFEK6YTE0P3Nmv:1:10:0] [11:58:23:INFO] [001:287] [31499] (basic_port_allocator.cc:940): Port[240a3200:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port ready. [11:58:23:INFO] [001:287] [6403] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [11:58:23:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 1 [11:58:23:INFO] [001:287] [31499] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [11:58:23:INFO] [001:287] [31499] (p2p_transport_channel.cc:799): Port[240a3200:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: SetOption(5, 0) failed: 0 [11:58:23:INFO] [001:287] [31499] (port.cc:1128): Conn[228df600:0:Net[en0:192.168.148.0/23:Wifi:id=1]:CTWoneSU:1:0:local:udp:192.168.148.41:64025->3IZx2iNm:1:2122260223:local:udp:192.168.148.41:61495|C--W|-|0|0|9115038255631187454|-]: Connection created [11:58:23:INFO] [001:288] [31499] (p2p_transport_channel.cc:1264): Channel[0|1|__]: Created connection with origin: 2, total: 1 [11:58:23:INFO] [001:288] [6403] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [11:58:23:INFO] [001:288] [31499] (p2p_transport_channel.cc:1914): Channel[0|1|__]: Transport channel state changed from 0 to 2 [11:58:23:INFO] [001:288] [31499] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete. [11:58:23:INFO] [001:288] [31499] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete. [11:58:23:INFO] [001:288] [31499] (p2p_transport_channel.cc:1476): Channel[0|1|__]: Have a pingable connection for the first time; starting to ping. [11:58:23:INFO] [001:288] [31499] (basic_port_allocator.cc:1017): Port[240a3200:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Port completed gathering candidates. [11:58:24:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 1 [11:58:24:INFO] [001:289] [31499] (port.cc:1776): Conn[228df600:0:Net[en0:192.168.148.0/23:Wifi:id=1]:CTWoneSU:1:0:local:udp:192.168.148.41:64025->3IZx2iNm:1:2122260223:local:udp:192.168.148.41:61495|C--W|-|0|0|9115038255631187454|-]: Sent STUN ping, id=6d612f622f426b306e716979, use_candidate=0, nomination=0 [11:58:24:INFO] [001:290] [49427] (port.cc:530): Received STUN ping id=6d612f622f426b306e716979 from unknown address 192.168.148.41:64025 [11:58:24:INFO] [001:290] [49427] (port.cc:1128): Conn[22853800:0:Net[en0:192.168.148.0/23:Wifi:id=1]:hxLFDt8t:1:0:local:udp:192.168.148.41:61495->eYu8kLSh:1:1853824767:prflx:udp:192.168.148.41:64025|C--W|-|0|0|7962116751024340479|-]: Connection created [11:58:24:INFO] [001:290] [49427] (p2p_transport_channel.cc:988): Adding connection from peer reflexive candidate: Cand[:3330429372:1:udp:1853824767:192.168.148.41:64025:prflx::0:/vWc::1:10:0] [11:58:24:INFO] [001:290] [49427] (port.cc:855): Port[2405b000:0:1:0:local:Net[en0:192.168.148.0/23:Wifi:id=1]]: Sent STUN ping response, to=192.168.148.41:64025, id=6d612f622f426b306e716979 [11:58:24:INFO] [001:290] [49427] (p2p_transport_channel.cc:1914): Channel[0|1|R_]: Transport channel state changed from 0 to 2 [11:58:24:INFO] [001:290] [49427] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete. [11:58:24:INFO] [001:290] [31499] (port.cc:1724): Conn[228df600:0:Net[en0:192.168.148.0/23:Wifi:id=1]:CTWoneSU:1:0:local:udp:192.168.148.41:64025->3IZx2iNm:1:2122260223:local:udp:192.168.148.41:61495|C--I|-|0|0|9115038255631187454|-]: Received STUN ping response, id=6d612f622f426b306e716979, code=0, rtt=1, pings_since_last_response=6d612f622f426b306e716979 [11:58:24:INFO] [001:290] [49427] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete. [11:58:24:INFO] [001:291] [31499] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed [11:58:24:INFO] [001:291] [31499] (p2p_transport_channel.cc:1852): Channel[0|1|__]: New selected connection: Conn[228df600:0:Net[en0:192.168.148.0/23:Wifi:id=1]:CTWoneSU:1:0:local:udp:192.168.148.41:64025->3IZx2iNm:1:2122260223:local:udp:192.168.148.41:61495|CRWS|S|0|0|9115038255631187454|1] [11:58:24:INFO] [001:291] [31499] (channel.cc:376): Network route was changed. [11:58:24:INFO] [001:291] [31499] (channel.cc:376): Network route was changed. [11:58:24:INFO] [001:291] [31499] (dtls_transport.cc:811): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 1 [11:58:24:INFO] [001:291] [31499] (openssl_stream_adapter.cc:777): BeginSSL with peer. [11:58:24:INFO] [001:293] [38691] (webrtc_video_engine.cc:1349): Call stats: 3909464, {send_bw_bps: 0, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1} [11:58:24:INFO] [001:294] [49427] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0 [11:58:24:INFO] [001:294] [49427] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 2240dde0 [11:58:24:INFO] [001:295] [49427] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=/vWc, renomination enabled [11:58:24:INFO] [001:295] [49427] (dtls_transport.cc:370): DtlsTransport[0|1|__]: DTLS setup complete. [11:58:24:INFO] [001:295] [38691] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:24:INFO] [001:295] [38691] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:24:INFO] [001:295] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client enter_early_data [11:58:24:INFO] [001:295] [49427] (p2p_transport_channel.cc:1476): Channel[0|1|R_]: Have a pingable connection for the first time; starting to ping. [11:58:24:INFO] [001:295] [38691] (channel.cc:553): Channel enabled [11:58:24:INFO] [001:295] [49427] (p2p_transport_channel.cc:2440): Selecting connection for triggered check: Conn[22853800:0:Net[en0:192.168.148.0/23:Wifi:id=1]:hxLFDt8t:1:0:local:udp:192.168.148.41:61495->eYu8kLSh:1:1853824767:prflx:udp:192.168.148.41:64025|CR-W|-|0|0|7962116751024340479|-] [11:58:24:INFO] [001:295] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client read_hello_verify_request [11:58:24:INFO] [001:295] [31499] (openssl_adapter.cc:794): SSL_connect:error in TLS client read_hello_verify_request [11:58:24:INFO] [001:295] [49427] (port.cc:1776): Conn[22853800:0:Net[en0:192.168.148.0/23:Wifi:id=1]:hxLFDt8t:1:0:local:udp:192.168.148.41:61495->eYu8kLSh:1:1853824767:prflx:udp:192.168.148.41:64025|CR-W|-|0|0|7962116751024340479|-]: Sent STUN ping, id=46536545476a50662b52394e, use_candidate=0, nomination=0 [11:58:24:INFO] [001:295] [31499] (dtls_transport.cc:698): DtlsTransport[0|1|__]: DtlsTransport: Started DTLS handshake [11:58:24:INFO] [001:295] [38691] (channel.cc:863): Changing voice state, recv=1 send=0 [11:58:24:INFO] [001:295] [49427] (dtls_transport.cc:546): DtlsTransport[0|1|__]: Packet received before DTLS started. [11:58:24:INFO] [001:296] [31499] (srtp_transport.cc:364): The params in SRTP transport are reset. [11:58:24:INFO] [001:296] [49427] (dtls_transport.cc:555): DtlsTransport[0|1|__]: Caching DTLS ClientHello packet until DTLS is started. [11:58:24:INFO] [001:296] [38691] (channel.cc:553): Channel enabled [11:58:24:INFO] [001:296] [31499] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete. [11:58:24:INFO] [001:296] [31499] (basic_port_allocator.cc:1119): All candidates gathered for 0:1:0 [11:58:24:INFO] [001:296] [31499] (p2p_transport_channel.cc:854): P2PTransportChannel: 0, component 1 gathering complete [11:58:24:INFO] [001:296] [38691] (channel.cc:996): Changing video state, send=0 [11:58:24:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 2 [11:58:24:INFO] [001:296] [79375] (peer_connection.cc:4173): Session: 3593490849194371550 Old state: kHaveLocalOffer New state: kStable [11:58:24:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 0 [11:58:24:INFO] [001:296] [38691] (channel.cc:923): Setting remote voice description [11:58:24:INFO] [001:296] [49427] (port.cc:1724): Conn[22853800:0:Net[en0:192.168.148.0/23:Wifi:id=1]:hxLFDt8t:1:0:local:udp:192.168.148.41:61495->eYu8kLSh:1:1853824767:prflx:udp:192.168.148.41:64025|CR-I|-|0|0|7962116751024340479|-]: Received STUN ping response, id=46536545476a50662b52394e, code=0, rtt=1, pings_since_last_response=46536545476a50662b52394e [11:58:24:INFO] [001:296] [38691] (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 {}} [11:58:24:INFO] [001:296] [49427] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed [11:58:24:INFO] [001:296] [49427] (p2p_transport_channel.cc:1852): Channel[0|1|R_]: New selected connection: Conn[22853800:0:Net[en0:192.168.148.0/23:Wifi:id=1]:hxLFDt8t:1:0:local:udp:192.168.148.41:61495->eYu8kLSh:1:1853824767:prflx:udp:192.168.148.41:64025|CRWS|S|0|0|7962116751024340479|1] [11:58:24:INFO] [001:296] [49427] (channel.cc:376): Network route was changed. [11:58:24:INFO] [001:296] [38691] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2481983948, extmap-allow-mixed: false, extensions: [], c_name: evBadRFH64bGTYuc}, 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}}}} [11:58:24:INFO] [001:297] [49427] (channel.cc:376): Network route was changed. [11:58:24:INFO] [001:297] [49427] (dtls_transport.cc:811): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 1 [11:58:24:INFO] [001:297] [38691] (audio_encoder_opus.cc:758): Set Opus bitrate to 32000 bps. [11:58:24:INFO] [001:297] [49427] (openssl_stream_adapter.cc:777): BeginSSL with peer. [11:58:24:INFO] [001:297] [38691] (webrtc_voice_engine.cc:1702): Recreate all the receive streams because the send codec has changed. [11:58:24:INFO] [001:297] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_hello [11:58:24:INFO] [001:297] [38691] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2481983948, 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: evBadRFH64bGTYuc}, 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}}}} [11:58:24:INFO] [001:297] [49427] (openssl_adapter.cc:794): SSL_accept:error in TLS server read_client_hello [11:58:24:INFO] [001:297] [49427] (dtls_transport.cc:698): DtlsTransport[0|1|__]: DtlsTransport: Started DTLS handshake [11:58:24:INFO] [001:297] [38691] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2481983948, 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: evBadRFH64bGTYuc}, 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}}}} [11:58:24:INFO] [001:298] [38691] (webrtc_voice_engine.cc:2163): WebRtcVoiceMediaChannel::SetMaxSendBitrate. [11:58:24:INFO] [001:298] [38691] (webrtc_voice_engine.cc:1485): Setting voice channel options: AudioOptions {} [11:58:24:INFO] [001:298] [49427] (srtp_transport.cc:364): The params in SRTP transport are reset. [11:58:24:INFO] [001:298] [38691] (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, } [11:58:24:INFO] [001:298] [49427] (dtls_transport.cc:704): DtlsTransport[0|1|__]: Handling cached DTLS ClientHello packet. [11:58:24:INFO] [001:298] [38691] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [11:58:24:INFO] [001:298] [38691] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [11:58:24:INFO] [001:298] [38691] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [11:58:24:INFO] [001:298] [38691] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [11:58:24:INFO] [001:298] [38691] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [11:58:24:INFO] [001:298] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server select_certificate [11:58:24:INFO] [001:298] [38691] (audio_device_module_ios.mm:600): output: 0 [11:58:24:INFO] [001:299] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server select_parameters [11:58:24:INFO] [001:299] [38691] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:24:INFO] [001:299] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_hello [11:58:24:INFO] [001:299] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_certificate [11:58:24:INFO] [001:299] [38691] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:24:INFO] [001:299] [38691] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:24:INFO] [001:299] [38691] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [11:58:24:INFO] [001:299] [38691] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [11:58:24:INFO] [001:299] [38691] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [11:58:24:INFO] [001:299] [38691] (audio_device_module_ios.mm:616): output: 0 [11:58:24:INFO] [001:299] [38691] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [11:58:24:INFO] [001:299] [38691] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [11:58:24:INFO] [001:299] [38691] (audio_device_module_ios.mm:632): output: 0 [11:58:24:INFO] [001:299] [38691] (apm_helpers.cc:62): NS set to 0 [11:58:24:INFO] [001:299] [38691] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [11:58:24:INFO] [001:299] [38691] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [11:58:24:INFO] [001:299] [38691] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [11:58:24:INFO] [001:299] [38691] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [11:58:24:INFO] [001:299] [38691] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [11:58:24:INFO] [001:299] [38691] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [11:58:24:INFO] [001:299] [38691] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [11:58:24:INFO] [001:299] [38691] (webrtc_voice_engine.cc:511): Setting AGC to 0 [11:58:24:INFO] [001:299] [38691] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [11:58:24:INFO] [001:299] [38691] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:24:INFO] [001:299] [38691] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:24:INFO] [001:299] [38691] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:24:INFO] [001:299] [38691] (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, } [11:58:24:INFO] [001:299] [38691] (webrtc_voice_engine.cc:1849): AddRecvStream: {id:ARDAMSa0;ssrcs:[3459412627];ssrc_groups:;cname:QRMy96WNfUpu1Re7;stream_ids:ARDAMS;} [11:58:24:INFO] [001:300] [38691] (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 [11:58:24:INFO] [001:300] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_key_exchange [11:58:24:INFO] [001:300] [38691] (audio_coding_module.cc:333): Created [11:58:24:INFO] [001:300] [38691] (audio_receive_stream.cc:117): AudioReceiveStream: 3459412627 [11:58:24:INFO] [001:300] [38691] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3459412627, local_ssrc: 2481983948, 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} [11:58:24:INFO] [001:303] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_hello_done [11:58:24:INFO] [001:304] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_certificate [11:58:24:INFO] [001:305] [38691] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:24:INFO] [001:305] [49427] (openssl_adapter.cc:794): SSL_accept:error in TLS server read_client_certificate [11:58:24:INFO] [001:305] [38691] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:24:INFO] [001:305] [38691] (audio_device_module_ios.mm:543): Playing [11:58:24:INFO] [001:305] [38691] (audio_device_module_ios.mm:480): InitPlayout [11:58:24:INFO] [001:305] [49427] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete. [11:58:24:INFO] [001:305] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_hello [11:58:24:INFO] [001:305] [38691] (audio_device_module_ios.mm:506): PlayoutIsInitialized [11:58:24:INFO] [001:305] [38691] (audio_device_ios.mm:185): AudioDeviceIOS::InitPlayout [11:58:24:INFO] [001:305] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_certificate [11:58:24:INFO] [001:308] [38691] (audio_device_ios.mm:859): AudioDeviceIOS::InitPlayOrRecord [11:58:24:INFO] [001:308] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client read_certificate_status [11:58:24:INFO] [001:309] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client verify_server_certificate [11:58:24:INFO] [001:309] [31499] (openssl_stream_adapter.cc:1049): Accepted peer certificate. [11:58:24:INFO] [001:309] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_key_exchange [11:58:24:INFO] [001:309] [38691] (RTCLogging.mm:33): (audio_device_ios.mm:824 ConfigureAudioSession): Configuring audio session. [11:58:24:INFO] [001:309] [38691] (RTCLogging.mm:33): (RTCAudioSession.mm:699 -[RTCAudioSession configureWebRTCSession:]): Configuring audio session for WebRTC. [11:58:24:INFO] [001:309] [38691] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:116 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred IO buffer duration to: 0.020000 [11:58:24:INFO] [001:309] [38691] (RTCLogging.mm:33): (RTCAudioSession.mm:624 -[RTCAudioSession incrementActivationCount]): Incrementing activation count. [11:58:24:INFO] [001:309] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client read_certificate_request [11:58:24:INFO] [001:324] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_hello_done [11:58:24:INFO] [001:324] [38691] (RTCLogging.mm:33): (RTCAudioSession.mm:377 -[RTCAudioSession setActive:error:]): Number of current activations: 2 [11:58:24:INFO] [001:324] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_certificate [11:58:24:INFO] [001:324] [38691] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:145 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set input number of channels to: 1 [11:58:24:INFO] [001:324] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_key_exchange [11:58:24:INFO] [001:324] [38691] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:158 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set output number of channels to: 1 [11:58:24:INFO] [001:324] [38691] (RTCLogging.mm:33): (audio_device_ios.mm:835 ConfigureAudioSession): Configured audio session. [11:58:24:INFO] [001:325] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_certificate_verify [11:58:24:INFO] [001:331] [38691] (audio_device_ios.mm:678): AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession [11:58:24:INFO] [001:331] [38691] (RTCLogging.mm:33): (audio_device_ios.mm:683 SetupAudioBuffersForActiveAudioSession): RTCAudioSession: { [11:58:24:INFO] category: AVAudioSessionCategoryPlayAndRecord [11:58:24:INFO] categoryOptions: 4 [11:58:24:INFO] mode: AVAudioSessionModeVoiceChat [11:58:24:INFO] isActive: 1 [11:58:24:INFO] sampleRate: 48000.00 [11:58:24:INFO] IOBufferDuration: 0.010667 [11:58:24:INFO] outputNumberOfChannels: 2 [11:58:24:INFO] inputNumberOfChannels: 2 [11:58:24:INFO] outputLatency: 0.010000 [11:58:24:INFO] inputLatency: 0.010000 [11:58:24:INFO] outputVolume: 0.600000 [11:58:24:INFO] } [11:58:24:INFO] [001:331] [38691] (audio_device_ios.mm:714): frames per I/O buffer: 512 [11:58:24:INFO] [001:331] [38691] (audio_device_ios.mm:715): bytes per I/O buffer: 1024 [11:58:24:INFO] [001:331] [38691] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer [11:58:24:INFO] [001:331] [38691] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000) [11:58:24:INFO] [001:334] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_finished [11:58:24:INFO] [001:343] [38691] (audio_device_buffer.cc:206): SetPlayoutChannels(1) [11:58:24:INFO] [001:343] [38691] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000) [11:58:24:INFO] [001:343] [38691] (audio_device_buffer.cc:200): SetRecordingChannels(1) [11:58:24:INFO] [001:343] [38691] (fine_audio_buffer.cc:32): FineAudioBuffer [11:58:24:INFO] [001:343] [38691] (fine_audio_buffer.cc:34): playout_samples_per_channel_10ms: 480 [11:58:24:INFO] [001:343] [38691] (fine_audio_buffer.cc:36): playout_channels: 1 [11:58:24:INFO] [001:343] [38691] (fine_audio_buffer.cc:39): record_samples_per_channel_10ms: 480 [11:58:24:INFO] [001:343] [38691] (fine_audio_buffer.cc:41): record_channels: 1 [11:58:24:INFO] [001:343] [38691] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:196 Initialize): Initializing audio unit with sample rate: 48000.000000 [11:58:24:INFO] [001:343] [38691] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:43 LogStreamDescription): AudioStreamBasicDescription: { [11:58:24:INFO] mSampleRate: 48000.00 [11:58:24:INFO] formatIDString: lpcm [11:58:24:INFO] mFormatFlags: 0xC [11:58:24:INFO] mBytesPerPacket: 2 [11:58:24:INFO] mFramesPerPacket: 1 [11:58:24:INFO] mBytesPerFrame: 2 [11:58:24:INFO] mChannelsPerFrame: 1 [11:58:24:INFO] mBitsPerChannel: 16 [11:58:24:INFO] mReserved: 28672 [11:58:24:INFO] } [11:58:24:INFO] [001:343] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client finish_flight [11:58:24:INFO] [001:345] [38691] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:250 Initialize): Voice Processing I/O unit is now initialized. [11:58:24:INFO] [001:345] [38691] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:71 GetAGCState): VPIO unit AGC: 0 [11:58:24:INFO] [001:345] [38691] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:264 Initialize): Failed to get AGC state (1st attempt). Error=-66631. [11:58:24:INFO] [001:345] [38691] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:302 Initialize): WebRTC.Audio.BuiltInAGCWasEnabledByDefault: 0 [11:58:24:INFO] [001:345] [38691] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:308 Initialize): WebRTC.Audio.BuiltInAGCIsEnabled: 0 [11:58:24:INFO] [001:345] [38691] (audio_device_module_ios.mm:486): output: 0 [11:58:24:INFO] [001:345] [38691] (audio_device_module_ios.mm:518): StartPlayout [11:58:24:INFO] [001:345] [38691] (audio_device_module_ios.mm:543): Playing [11:58:24:INFO] [001:346] [38691] (audio_device_buffer.cc:98): StartPlayout [11:58:24:INFO] [001:346] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client read_session_ticket [11:58:24:INFO] [001:346] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server verify_client_certificate [11:58:24:INFO] [001:360] [38691] (audio_device_ios.mm:227): AudioDeviceIOS::StartPlayout [11:58:24:INFO] [001:360] [31499] (openssl_adapter.cc:794): SSL_connect:error in TLS client read_session_ticket [11:58:24:INFO] [001:360] [38691] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:316 Start): Starting audio unit. [11:58:24:INFO] [001:360] [49427] (openssl_stream_adapter.cc:1049): Accepted peer certificate. [11:58:24:INFO] [001:360] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_key_exchange [11:58:24:INFO] [001:360] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_certificate_verify [11:58:24:INFO] [001:361] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server read_change_cipher_spec [11:58:24:INFO] [001:361] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server process_change_cipher_spec [11:58:24:INFO] [001:361] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server read_next_proto [11:58:24:INFO] [001:367] [38691] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:320 Start): Failed to start audio unit. Error=-66637 [11:58:24:INFO] [001:372] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server read_channel_id [11:58:24:INFO] [001:372] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_finished [11:58:24:INFO] [001:372] [38691] (RTCLogging.mm:33): (audio_device_ios.mm:237 StartPlayout): StartPlayout failed to start audio unit. [11:58:24:INFO] [001:372] [38691] (audio_device_module_ios.mm:525): output: -1 [11:58:24:INFO] [001:372] [38691] (channel.cc:719): Add remote ssrc: 3459412627 [11:58:24:INFO] [001:372] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_finished [11:58:24:INFO] [001:372] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server finish_server_handshake [11:58:24:INFO] [001:372] [49427] (openssl_adapter.cc:784): SSL_accept:TLS server done [11:58:24:INFO] [001:372] [49427] (dtls_transport.cc:630): DtlsTransport[0|1|__]: DTLS handshake complete. [11:58:24:INFO] [001:372] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client process_change_cipher_spec [11:58:24:INFO] [001:373] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_finished [11:58:24:INFO] [001:373] [49427] (jsep_transport_controller.cc:1213): Transport 0 writability changed to 1. [11:58:24:INFO] [001:379] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client finish_client_handshake [11:58:24:INFO] [001:379] [49427] (dtls_srtp_transport.cc:219): Extracting keys from transport: 0 [11:58:24:INFO] [001:379] [31499] (openssl_adapter.cc:784): SSL_connect:TLS client done [11:58:24:INFO] [001:379] [31499] (dtls_transport.cc:630): DtlsTransport[0|1|__]: DTLS handshake complete. [11:58:24:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 2 [11:58:24:INFO] [001:379] [31499] (jsep_transport_controller.cc:1213): Transport 0 writability changed to 1. [11:58:24:INFO] [001:379] [31499] (dtls_srtp_transport.cc:219): Extracting keys from transport: 0 [11:58:24:INFO] [001:379] [79107] (peer_connection.cc:6044): Changing to ICE connected state because all transports are writable. [11:58:24:INFO] [001:379] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [11:58:24:INFO] [001:379] [79107] (peer_connection.cc:4109): Changing IceConnectionState 1 => 2 [11:58:24:INFO] [001:385] [49427] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [11:58:24:INFO] [001:385] [31499] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [11:58:24:INFO] [001:387] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [11:58:24:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 2 [11:58:24:INFO] [001:387] [49427] (channel.cc:583): Channel writable (0) for the first time [11:58:24:INFO] [001:387] [31499] (channel.cc:583): Channel writable (0) for the first time [11:58:24:INFO] [001:387] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [11:58:24:INFO] [001:388] [49427] (channel.cc:583): Channel writable (1) for the first time [11:58:24:INFO] [001:388] [31499] (channel.cc:583): Channel writable (1) for the first time [11:58:24:INFO] [001:388] [36103] (rtp_transport_controller_send.cc:459): Creating fallback congestion controller [11:58:24:INFO] [001:388] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [11:58:24:INFO] [001:388] [38691] (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, } [11:58:24:INFO] [001:393] [38691] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [11:58:24:INFO] [001:393] [38691] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [11:58:24:INFO] [001:393] [38691] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [11:58:24:INFO] [001:393] [38691] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [11:58:24:INFO] [001:393] [38691] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [11:58:24:INFO] [001:393] [38691] (audio_device_module_ios.mm:600): output: 0 [11:58:24:INFO] [001:393] [38691] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:24:INFO] [001:394] [38691] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:24:INFO] [001:394] [38691] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:24:INFO] [001:394] [50183] (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, } [11:58:24:INFO] [001:394] [36103] (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 [11:58:24:INFO] [001:397] [38691] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [11:58:24:INFO] [001:397] [50183] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [11:58:24:INFO] [001:397] [38691] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [11:58:24:INFO] [001:397] [50183] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [11:58:24:INFO] [001:397] [38691] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [11:58:24:INFO] [001:397] [36103] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85 [11:58:24:INFO] [001:407] [50183] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [11:58:24:INFO] [001:407] [50183] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [11:58:24:INFO] [001:407] [36103] (delay_based_bwe.cc:107): Using Trendline filter for delay change estimation with window size 20 [11:58:24:INFO] [001:407] [50183] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [11:58:24:INFO] [001:407] [50183] (audio_device_module_ios.mm:600): output: 0 [11:58:24:INFO] [001:407] [50183] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:24:INFO] [001:407] [38691] (audio_device_module_ios.mm:616): output: 0 [11:58:24:INFO] [001:407] [38691] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [11:58:24:INFO] [001:407] [38691] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [11:58:24:INFO] [001:407] [38691] (audio_device_module_ios.mm:632): output: 0 [11:58:24:INFO] [001:407] [50183] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:24:INFO] [001:408] [36103] (delay_based_bwe.cc:297): BWE Setting start bitrate to: 300 kbps [11:58:24:INFO] [001:429] [38691] (apm_helpers.cc:62): NS set to 0 [11:58:24:INFO] [001:429] [38691] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [11:58:24:INFO] [001:429] [38691] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [11:58:24:INFO] [001:429] [38691] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [11:58:24:INFO] [001:429] [38691] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [11:58:24:INFO] [001:429] [38691] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [11:58:24:INFO] [001:429] [38691] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [11:58:24:INFO] [001:429] [38691] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [11:58:24:INFO] [001:429] [38691] (webrtc_voice_engine.cc:511): Setting AGC to 0 [11:58:24:INFO] [001:429] [38691] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [11:58:24:INFO] [001:429] [38691] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:24:INFO] [001:429] [50183] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:24:INFO] [001:430] [36103] (probe_controller.cc:266): Measured bitrate: 300000 Minimum to probe further: 1260000 [11:58:24:INFO] [001:430] [38691] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:24:INFO] [001:430] [38691] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:24:INFO] [001:430] [50183] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [11:58:24:INFO] [001:430] [38691] (audio_device_module_ios.mm:512): RecordingIsInitialized [11:58:24:INFO] [001:430] [50183] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [11:58:24:INFO] [001:430] [50183] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [11:58:24:INFO] [001:430] [50183] (audio_device_module_ios.mm:616): output: 0 [11:58:24:INFO] [001:430] [36103] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (900000:1687:5) [11:58:24:INFO] [001:430] [36103] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5) [11:58:24:INFO] [001:430] [50183] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [11:58:24:INFO] [001:430] [36103] (bitrate_allocator.cc:115): Current BWE 300000 [11:58:24:INFO] [001:430] [38691] (audio_device_module_ios.mm:574): Recording [11:58:24:INFO] [001:506] [38691] (audio_device_module_ios.mm:493): InitRecording [11:58:24:INFO] [001:506] [50183] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [11:58:24:INFO] [001:506] [38691] (audio_device_module_ios.mm:512): RecordingIsInitialized [11:58:24:INFO] [001:506] [38691] (audio_device_module_ios.mm:549): StartRecording [11:58:24:INFO] [001:506] [50183] (audio_device_module_ios.mm:632): output: 0 [11:58:24:INFO] [001:506] [38691] (audio_device_module_ios.mm:574): Recording [11:58:24:INFO] [001:506] [50183] (apm_helpers.cc:62): NS set to 0 [11:58:24:INFO] [001:506] [38691] (audio_device_buffer.cc:117): StartRecording [11:58:24:INFO] [001:506] [50183] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [11:58:24:INFO] [001:506] [38691] (audio_device_ios.mm:280): AudioDeviceIOS::StartRecording [11:58:24:INFO] [001:506] [50183] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [11:58:24:INFO] [001:506] [50183] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [11:58:24:INFO] [001:506] [38691] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:316 Start): Starting audio unit. [11:58:24:INFO] [001:529] [50183] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [11:58:24:INFO] [001:529] [50183] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [11:58:24:INFO] [001:529] [50183] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [11:58:24:INFO] [001:529] [50183] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [11:58:24:INFO] [001:529] [50183] (webrtc_voice_engine.cc:511): Setting AGC to 0 [11:58:24:INFO] [001:529] [50183] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [11:58:24:INFO] [001:529] [50183] (audio_processing_impl.cc:727): Highpass filter activated: 1 [11:58:24:INFO] [001:529] [50183] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [11:58:24:INFO] [001:530] [50183] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [11:58:24:INFO] [001:530] [50183] (audio_device_module_ios.mm:512): RecordingIsInitialized [11:58:24:INFO] [001:530] [50183] (audio_device_module_ios.mm:574): Recording [11:58:24:INFO] [001:530] [50183] (audio_device_module_ios.mm:493): InitRecording [11:58:24:INFO] [001:541] [38691] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:320 Start): Failed to start audio unit. Error=-66637 [11:58:24:INFO] [001:544] [50183] (audio_device_module_ios.mm:512): RecordingIsInitialized [11:58:24:INFO] [001:544] [50183] (audio_device_module_ios.mm:549): StartRecording [11:58:24:INFO] [001:544] [38691] (RTCLogging.mm:33): (audio_device_ios.mm:290 StartRecording): StartRecording failed to start audio unit. [11:58:24:INFO] [001:544] [50183] (audio_device_module_ios.mm:574): Recording [11:58:24:INFO] [001:544] [38691] (audio_device_module_ios.mm:556): output: -1 [11:58:24:INFO] [001:544] [50183] (audio_device_buffer.cc:117): StartRecording [11:58:24:INFO] [001:544] [38691] (channel.cc:863): Changing voice state, recv=1 send=1 [11:58:24:INFO] [001:544] [50183] (audio_device_ios.mm:280): AudioDeviceIOS::StartRecording [11:58:24:INFO] [001:544] [38691] (message_queue.cc:517): Message took 248ms to dispatch. Posted from: SetRemoteContent@../../pc/channel.cc:302 [11:58:24:INFO] [001:544] [50183] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:316 Start): Starting audio unit. [11:58:24:INFO] [001:544] [38691] (channel.cc:1061): Setting remote video description [11:58:24:INFO] [001:553] [50183] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:320 Start): Failed to start audio unit. Error=-66637 [11:58:24:INFO] [001:559] [50183] (RTCLogging.mm:33): (audio_device_ios.mm:290 StartRecording): StartRecording failed to start audio unit. [11:58:24:INFO] [001:559] [50183] (audio_device_module_ios.mm:556): output: -1 [11:58:24:INFO] [001:559] [50183] (channel.cc:863): Changing voice state, recv=1 send=1 [11:58:24:INFO] [001:559] [50183] (message_queue.cc:517): Message took 166ms to dispatch. Posted from: UpdateMediaSendRecvState@../../pc/channel.cc:830 [11:58:24:INFO] [001:559] [38691] (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} [11:58:24:INFO] [001:560] [50183] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [11:58:24:INFO] [001:565] [36103] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers [11:58:24:INFO] [001:565] [36103] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps [11:58:24:INFO] [001:565] [38691] (webrtc_video_engine.cc:690): Using codec: VideoCodec[96:H264] [11:58:24:INFO] [001:566] [11011] (video_stream_encoder.cc:1568): Video suspend state changed to: not suspended [11:58:24:INFO] [001:573] [50183] (channel.cc:996): Changing video state, send=1 [11:58:24:INFO] [001:573] [38691] (webrtc_video_engine.cc:1883): RecreateWebRtcStream (send) because of SetCodec. [11:58:24:INFO] [001:573] [38691] (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 [11:58:24:INFO] [001:574] [11011] (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. [11:58:24:INFO] [001:574] [11011] (video_send_stream_impl.cc:250): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [715828121], 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: [993092192], payload_type: 97}, c_name: evBadRFH64bGTYuc}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [11:58:24:INFO] [001:594] [11011] (video_send_stream_impl.cc:278): ERROR: Initial encoder max bitrate = -1 which is <= 0! [11:58:24:INFO] [001:594] [11011] (video_send_stream_impl.cc:280): Using default encoder max bitrate = 10 Mbps [11:58:24:INFO] [001:594] [11011] (video_stream_encoder.cc:628): ConfigureEncoder requested. [11:58:24:INFO] [001:594] [38691] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:24:INFO] [001:594] [38691] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:24:INFO] [001:594] [38691] (video_send_stream.cc:160): VideoSendStream::Stop [11:58:24:INFO] [001:594] [38691] (webrtc_video_engine.cc:758): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. [11:58:24:INFO] [001:594] [11011] (video_send_stream_impl.cc:417): VideoSendStream::Stop [11:58:24:INFO] [001:594] [38691] (webrtc_video_engine.cc:1174): AddRecvStream: {id:ARDAMSv0;ssrcs:[2757358605,2295916538];ssrc_groups:{semantics:FID;ssrcs:[2757358605,2295916538]};cname:QRMy96WNfUpu1Re7;stream_ids:ARDAMS;} [11:58:24:INFO] [001:595] [38691] (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: 2757358605, local_ssrc: 715828121, 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: 2295916538, 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} [11:58:24:INFO] [001:620] [38691] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:24:INFO] [001:620] [38691] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:24:INFO] [001:621] [38691] (channel.cc:719): Add remote ssrc: 2757358605 [11:58:24:INFO] [001:621] [38691] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [11:58:24:INFO] [001:621] [11011] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers [11:58:24:INFO] [001:621] [11011] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps [11:58:24:INFO] [001:621] [38691] (channel.cc:996): Changing video state, send=1 [11:58:24:INFO] [001:621] [38691] (message_queue.cc:517): Message took 77ms to dispatch. Posted from: SetRemoteContent@../../pc/channel.cc:302 [11:58:24:INFO] [001:621] [79375] (peer_connection.cc:4109): Changing IceConnectionState 0 => 1 [11:58:24:INFO] [001:621] [38691] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [11:58:24:INFO] [001:621] [38691] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [11:58:24:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 1 [11:58:24:INFO] [001:621] [79375] (peer_connection.cc:5518): Non-rejected SCTP m= section is needed to get the SSL Role of the SCTP transport. [11:58:24:INFO] [001:621] [36103] (rtp_transport_controller_send.cc:459): Creating fallback congestion controller [11:58:24:INFO] [001:621] [38691] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [11:58:24:INFO] [001:637] [79375] (peer_connection.cc:2745): Processing the MSIDs for MID=0 (streams=[ARDAMS]). [11:58:24:INFO] [001:637] [38691] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [11:58:24:INFO] [001:637] [79375] (peer_connection.cc:2756): Processing the addition of a remote track for MID=0. [11:58:24:INFO] [001:637] [79375] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv. [11:58:24:INFO] [001:637] [36103] (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 [11:58:24:INFO] [001:637] [36103] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85 [11:58:24:INFO] [001:637] [36103] (delay_based_bwe.cc:107): Using Trendline filter for delay change estimation with window size 20 [11:58:24:INFO] [001:637] [38691] (channel.cc:863): Changing voice state, recv=1 send=1 [11:58:24:INFO] [001:646] [36103] (delay_based_bwe.cc:297): BWE Setting start bitrate to: 300 kbps [11:58:24:INFO] [001:646] [36103] (probe_controller.cc:266): Measured bitrate: 300000 Minimum to probe further: 1260000 [11:58:24:INFO] [001:647] [36103] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (900000:1687:5) [11:58:24:INFO] [001:647] [36103] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5) [11:58:24:INFO] [001:647] [36103] (bitrate_allocator.cc:115): Current BWE 300000 [11:58:24:INFO] [001:647] [38691] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [11:58:24:INFO] [001:647] [36103] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers [11:58:24:INFO] [001:647] [38691] (channel.cc:996): Changing video state, send=1 [11:58:24:INFO] [001:647] [38691] (webrtc_voice_engine.cc:1962): SetOutputVolume() to 1 for recv stream with ssrc 3459412627 [11:58:24:INFO] [001:647] [11011] (video_stream_encoder.cc:1568): Video suspend state changed to: not suspended [11:58:24:INFO] [001:647] [79375] (peer_connection.cc:2745): Processing the MSIDs for MID=1 (streams=[ARDAMS]). [11:58:24:INFO] [001:655] [79375] (peer_connection.cc:2756): Processing the addition of a remote track for MID=1. [11:58:24:INFO] [001:655] [79375] (rtp_transceiver.cc:169): Changing transceiver (MID=1) current direction from <not set> to kSendRecv. [11:58:24:INFO] [001:655] [38691] (webrtc_video_engine.cc:1304): SetSink: ssrc:2757358605 (ptr) [11:58:24:INFO] [001:655] [79375] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60000215d650): created DTMF sender: RTCDtmfSender { [11:58:24:INFO] remainingTones: [11:58:24:INFO] duration: 0.100000 sec [11:58:24:INFO] interToneGap: 0.050000 sec [11:58:24:INFO] } [11:58:24:INFO] [001:655] [79375] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60000230a400): created sender: RTCRtpSender { [11:58:24:INFO] senderId: ARDAMSa0 [11:58:24:INFO] } [11:58:24:INFO] [001:656] [79375] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x600002d68300): created receiver: RTCRtpReceiver { [11:58:24:INFO] receiverId: 828d5df9-b2e6-42a2-b67c-2d0ddb5b0835 [11:58:24:INFO] } [11:58:24:INFO] [001:670] [79375] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x600002d57360): created transceiver: RTCRtpTransceiver { [11:58:24:INFO] sender: RTCRtpSender { [11:58:24:INFO] senderId: ARDAMSa0 [11:58:24:INFO] } [11:58:24:INFO] receiver: RTCRtpReceiver { [11:58:24:INFO] receiverId: 828d5df9-b2e6-42a2-b67c-2d0ddb5b0835 [11:58:24:INFO] } [11:58:24:INFO] } [11:58:24:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving audio on track 828d5df9-b2e6-42a2-b67c-2d0ddb5b0835. [11:58:24:INFO] [001:670] [79375] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [11:58:24:INFO] [001:670] [79375] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60000230c060): created sender: RTCRtpSender { [11:58:24:INFO] senderId: ARDAMSv0 [11:58:24:INFO] } [11:58:24:INFO] [001:670] [79375] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x600002d79140): created receiver: RTCRtpReceiver { [11:58:24:INFO] receiverId: 6dc52d42-af21-4e8d-bafa-0766c8fc81e3 [11:58:24:INFO] } [11:58:24:INFO] [001:670] [79375] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x600002d79560): created transceiver: RTCRtpTransceiver { [11:58:24:INFO] sender: RTCRtpSender { [11:58:24:INFO] senderId: ARDAMSv0 [11:58:24:INFO] } [11:58:24:INFO] receiver: RTCRtpReceiver { [11:58:24:INFO] receiverId: 6dc52d42-af21-4e8d-bafa-0766c8fc81e3 [11:58:24:INFO] } [11:58:24:INFO] } [11:58:24:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving video on track 6dc52d42-af21-4e8d-bafa-0766c8fc81e3. [11:58:24:INFO] (RTCLogging.mm:33): (ARDAppClient.m:381 -[ARDAppClient peerConnection:didAddStream:]): Stream with 1 video tracks and 1 audio tracks was added. [11:58:24:INFO] [001:684] [79375] (message_queue.cc:517): Message took 394ms to dispatch. Posted from: SetRemoteDescription@../../api/peer_connection_proxy.h:104 [11:58:24:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 1 [11:58:24:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 2 [11:58:24:INFO] [001:684] [79375] (peer_connection.cc:6050): Changing to ICE completed state because all transports are complete. [11:58:24:INFO] [001:684] [79375] (peer_connection.cc:4109): Changing IceConnectionState 1 => 2 [11:58:24:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 2 [11:58:24:INFO] [001:684] [79375] (peer_connection.cc:4109): Changing IceConnectionState 2 => 3 [11:58:24:INFO] [001:684] [6403] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x60000215d6c0): created DTMF sender: RTCDtmfSender { [11:58:24:INFO] remainingTones: [11:58:24:INFO] duration: 0.100000 sec [11:58:24:INFO] interToneGap: 0.050000 sec [11:58:24:INFO] } [11:58:24:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 3 [11:58:24:INFO] [001:692] [6403] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60000230c060): created sender: RTCRtpSender { [11:58:24:INFO] senderId: ARDAMSa0 [11:58:24:INFO] } [11:58:24:INFO] [001:692] [79107] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [11:58:24:INFO] [001:692] [6403] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60000230a400): created sender: RTCRtpSender { [11:58:24:INFO] senderId: ARDAMSv0 [11:58:24:INFO] } [11:58:24:INFO] [001:694] [31499] (jsep_transport_controller.cc:318): Not adding candidate because the JsepTransport doesn't exist. Ignore it. [11:58:24:INFO] [001:699] [50183] (rtc_event_log_impl.cc:211): Stopping WebRTC event log. [11:58:24:INFO] [001:699] [50183] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped. [11:58:24:INFO] [001:701] [50183] (webrtc_video_engine.cc:2234): VideoSendStream stats: 3909872, {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: 2757358605, 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} [11:58:24:INFO] [001:701] [50183] (webrtc_video_engine.cc:2731): VideoReceiveStream stats: 3909872, {ssrc: 715828121, 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} [11:58:24:INFO] [001:701] [50183] (webrtc_video_engine.cc:1349): Call stats: 3909872, {send_bw_bps: 300000, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1} [11:58:24:INFO] [001:704] [79107] (peer_connection.cc:4173): Session: 4729406347323590376 Old state: kStable New state: kClosed [11:58:24:INFO] [001:704] [50183] (audio_device_module_ios.mm:563): StopRecording [11:58:24:INFO] [001:704] [50183] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording [11:58:24:INFO] [001:704] [50183] (audio_device_buffer.cc:153): StopRecording [11:58:24:INFO] [001:704] [50183] (audio_device_buffer.cc:176): total recording time: 160 [11:58:24:INFO] [001:704] [50183] (audio_device_module_ios.mm:567): output: 0 [11:58:24:INFO] [001:704] [50183] (webrtc_voice_engine.cc:1962): SetOutputVolume() to 0 for recv stream with ssrc 2481983948 [11:58:24:INFO] [001:704] [50183] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 2757358605, options: nullptr, source = nullptr) [11:58:24:INFO] [001:704] [50183] (webrtc_video_engine.cc:1304): SetSink: ssrc:715828121 nullptr [11:58:24:INFO] [001:705] [50183] (channel.cc:563): Channel disabled [11:58:24:INFO] [001:705] [50183] (video_send_stream.cc:160): VideoSendStream::Stop [11:58:24:INFO] [001:705] [50183] (channel.cc:996): Changing video state, send=0 [11:58:24:INFO] [001:705] [36103] (video_send_stream_impl.cc:417): VideoSendStream::Stop [11:58:24:INFO] [001:705] [36103] (rtp_rtcp_impl.cc:360): Failed to send RTCP BYE [11:58:24:INFO] [001:705] [50183] (video_send_stream.cc:160): VideoSendStream::Stop [11:58:24:INFO] [001:715] [36103] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps [11:58:24:INFO] [001:715] [11011] (video_stream_encoder.cc:1568): Video suspend state changed to: suspended [11:58:24:INFO] [001:715] [36103] (video_send_stream_impl.cc:417): VideoSendStream::Stop [11:58:24:INFO] [001:715] [36103] (video_send_stream_impl.cc:417): VideoSendStream::Stop [11:58:24:INFO] [001:715] [36103] (video_send_stream_impl.cc:338): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [2757358605], 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: [2295916538], payload_type: 97}, c_name: QRMy96WNfUpu1Re7}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [11:58:24:INFO] [001:716] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [11:58:24:INFO] [001:716] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [11:58:24:INFO] [001:718] [50183] (send_statistics_proxy.cc:655): Frames encoded 0 [11:58:24:INFO] WebRTC.Video.DroppedFrames.Capturer 0 [11:58:24:INFO] WebRTC.Video.DroppedFrames.EncoderQueue 0 [11:58:24:INFO] WebRTC.Video.DroppedFrames.Encoder 0 [11:58:24:INFO] WebRTC.Video.DroppedFrames.Ratelimiter 0 [11:58:24:INFO] [001:719] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [11:58:24:INFO] [001:719] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [11:58:24:INFO] [001:719] [50183] (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: 715828121, local_ssrc: 2757358605, 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: 993092192, 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} [11:58:24:INFO] [001:729] [50183] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [11:58:24:INFO] [001:730] [50183] (receive_statistics_proxy.cc:487): Frames decoded 0 [11:58:24:INFO] WebRTC.Video.DroppedFrames.Receiver 0 [11:58:24:INFO] [11:58:24:INFO] [001:730] [50183] (channel.cc:162): Destroyed channel: 1 [11:58:24:INFO] [001:730] [50183] (channel.cc:563): Channel disabled [11:58:24:INFO] [001:730] [50183] (audio_device_module_ios.mm:532): StopPlayout [11:58:24:INFO] [001:730] [50183] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout [11:58:24:INFO] [001:730] [50183] (audio_device_buffer.cc:139): StopPlayout [11:58:24:INFO] [001:730] [50183] (audio_device_buffer.cc:145): total playout time: 466 [11:58:24:INFO] [001:730] [50183] (audio_device_module_ios.mm:536): output: 0 [11:58:24:INFO] [001:730] [50183] (channel.cc:863): Changing voice state, recv=0 send=0 [11:58:24:INFO] [001:730] [50183] (webrtc_voice_engine.cc:1823): RemoveSendStream: 3459412627 [11:58:24:INFO] [001:730] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [11:58:24:INFO] [001:730] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [11:58:24:INFO] [001:730] [50183] (audio_send_stream.cc:179): ~AudioSendStream: 3459412627 [11:58:24:INFO] [001:731] [50183] (webrtc_voice_engine.cc:1900): RemoveRecvStream: 2481983948 [11:58:24:INFO] [001:731] [50183] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:24:INFO] [001:742] [50183] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:24:INFO] [001:742] [50183] (audio_receive_stream.cc:140): ~AudioReceiveStream: 2481983948 [11:58:24:INFO] [001:742] [36103] (paced_sender.cc:106): PacedSender paused. [11:58:24:INFO] [001:742] [36103] (control_handler.cc:77): Bitrate estimate state changed, BWE: 300 kbps. [11:58:24:INFO] [001:742] [50183] (channel.cc:162): Destroyed channel: 0 [11:58:24:INFO] [001:743] [31499] (openssl_stream_adapter.cc:896): Cleanup [11:58:24:INFO] [001:743] [31499] (openssl_adapter.cc:787): SSL3 alert write:warning:close notify [11:58:24:INFO] [001:743] [49427] (openssl_adapter.cc:787): SSL3 alert read:warning:close notify [11:58:24:INFO] [001:743] [49427] (openssl_stream_adapter.cc:896): Cleanup [11:58:24:INFO] [001:743] [49427] (openssl_adapter.cc:787): SSL3 alert write:warning:close notify [11:58:24:INFO] [001:743] [49427] (dtls_transport.cc:651): DtlsTransport[0|1|_W]: DTLS transport closed [11:58:24:INFO] [001:743] [49427] (jsep_transport_controller.cc:1213): Transport 0 writability changed to 0. [11:58:24:INFO] [001:743] [79375] (peer_connection.cc:4109): Changing IceConnectionState 3 => 5 [11:58:24:INFO] [001:743] [49427] (channel.cc:596): Channel not writable (0) [11:58:24:INFO] [001:745] [49427] (channel.cc:596): Channel not writable (1) [11:58:24:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 5 [11:58:24:INFO] [001:745] [49427] (srtp_transport.cc:364): The params in SRTP transport are reset. [11:58:24:INFO] [001:745] [38691] (channel.cc:863): Changing voice state, recv=1 send=1 [11:58:24:INFO] [001:745] [38691] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [11:58:24:INFO] [001:745] [36103] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers [11:58:24:INFO] [001:745] [38691] (channel.cc:996): Changing video state, send=1 [11:58:24:INFO] [001:756] [50183] (paced_sender.cc:418): ProcessThreadAttached 0x0 [11:58:24:INFO] [001:756] [50183] (paced_sender.cc:418): ProcessThreadAttached 0x0 [11:58:24:INFO] [001:756] [50183] (rtc_event_log_impl.cc:211): Stopping WebRTC event log. [11:58:24:INFO] [001:757] [50183] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped. [11:58:24:INFO] [001:757] [79107] (peer_connection.cc:6776): Usage signature is 4088 [11:58:24:INFO] [001:757] [79107] (message_queue.cc:517): Message took 58ms to dispatch. Posted from: Close@../../api/peer_connection_proxy.h:140 [11:58:24:INFO] [001:757] [79107] (peer_connection.cc:915): Session: 4729406347323590376 is destroyed. [11:58:24:INFO] [001:757] [79107] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue. [11:58:24:INFO] [001:758] [50183] (webrtc_video_engine.cc:474): WebRtcVideoEngine::~WebRtcVideoEngine [11:58:24:INFO] [001:758] [50183] (webrtc_voice_engine.cc:205): WebRtcVoiceEngine::~WebRtcVoiceEngine [11:58:24:INFO] [001:758] [50183] (audio_device_module_ios.mm:532): StopPlayout [11:58:24:INFO] [001:758] [50183] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout [11:58:24:INFO] [001:758] [50183] (audio_device_module_ios.mm:536): output: 0 [11:58:24:INFO] [001:758] [50183] (audio_device_module_ios.mm:563): StopRecording [11:58:24:INFO] [001:758] [50183] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording [11:58:24:INFO] [001:758] [50183] (audio_device_module_ios.mm:567): output: 0 [11:58:24:INFO] [001:758] [50183] (audio_device_module_ios.mm:581): RegisterAudioCallback [11:58:24:INFO] [001:758] [50183] (audio_device_buffer.cc:81): RegisterAudioCallback [11:58:24:INFO] [001:758] [50183] (audio_device_module_ios.mm:92): Terminate [11:58:24:INFO] [001:758] [50183] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate [11:58:24:INFO] [001:758] [50183] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout [11:58:24:INFO] [001:758] [50183] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording [11:58:24:INFO] [001:760] [50183] (audio_device_module_ios.mm:55): ~AudioDeviceModuleIOS [11:58:24:INFO] [001:760] [50183] (audio_device_buffer.cc:75): AudioDeviceBuffer::~dtor [11:58:24:INFO] [001:760] [50183] (audio_device_ios.mm:127): AudioDeviceIOS::~dtor<NSThread: 0x6000039e4700>{number = 4, name = (null)} [11:58:24:INFO] [001:760] [50183] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate [11:58:24:INFO] [001:760] [50183] (fine_audio_buffer.cc:46): ~FineAudioBuffer [11:58:24:INFO] [001:760] [50183] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:347 Uninitialize): Unintializing audio unit. [11:58:24:INFO] [001:761] [50183] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:354 Uninitialize): Uninitialized audio unit. [11:58:24:INFO] [001:761] [50183] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:459 DisposeAudioUnit): Disposing audio unit. [11:58:24:INFO] Test Case '-[ARDAppClientTest testSession]' passed (1.773 seconds). [11:58:24:INFO] Test Suite 'ARDAppClientTest' passed at 2019-04-17 23:58:23.897. [11:58:24:INFO] Executed 1 test, with 0 failures (0 unexpected) in 1.773 (1.774) seconds [11:58:24:INFO] Test Suite 'ARDFileCaptureControllerTests' started at 2019-04-17 23:58:23.897 [11:58:24:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStarted]' started. [11:58:24:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStarted]' passed (0.001 seconds). [11:58:24:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStoped]' started. [11:58:24:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStoped]' passed (0.001 seconds). [11:58:24:INFO] Test Suite 'ARDFileCaptureControllerTests' passed at 2019-04-17 23:58:23.908. [11:58:24:INFO] Executed 2 tests, with 0 failures (0 unexpected) in 0.002 (0.010) seconds [11:58:24:INFO] Test Suite 'ARDSettingsModelTests' started at 2019-04-17 23:58:23.908 [11:58:24:INFO] Test Case '-[ARDSettingsModelTests testConstraintComponentIsNilWhenInvalidConstraintString]' started. [11:58:24:INFO] Test Case '-[ARDSettingsModelTests testConstraintComponentIsNilWhenInvalidConstraintString]' passed (0.001 seconds). [11:58:24:INFO] Test Case '-[ARDSettingsModelTests testHeightConstraintFromStore]' started. [11:58:24:INFO] Test Case '-[ARDSettingsModelTests testHeightConstraintFromStore]' passed (0.001 seconds). [11:58:24:INFO] Test Case '-[ARDSettingsModelTests testRetrievingSetting]' started. [11:58:24:INFO] Test Case '-[ARDSettingsModelTests testRetrievingSetting]' passed (0.001 seconds). [11:58:24:INFO] Test Case '-[ARDSettingsModelTests testReturningDefaultCallOption]' started. [11:58:24:INFO] Test Case '-[ARDSettingsModelTests testReturningDefaultCallOption]' passed (0.001 seconds). [11:58:24:INFO] Test Case '-[ARDSettingsModelTests testStoringAudioSetting]' started. [11:58:24:INFO] Test Case '-[ARDSettingsModelTests testStoringAudioSetting]' passed (0.001 seconds). [11:58:24:INFO] Test Case '-[ARDSettingsModelTests testStoringInvalidConstraintReturnsNo]' started. [11:58:24:INFO] Test Case '-[ARDSettingsModelTests testStoringInvalidConstraintReturnsNo]' passed (0.001 seconds). [11:58:24:INFO] Test Case '-[ARDSettingsModelTests testWidthConstraintFromStore]' started. [11:58:24:INFO] Test Case '-[ARDSettingsModelTests testWidthConstraintFromStore]' passed (0.001 seconds). [11:58:24:INFO] Test Suite 'ARDSettingsModelTests' passed at 2019-04-17 23:58:23.940. [11:58:24:INFO] Executed 7 tests, with 0 failures (0 unexpected) in 0.007 (0.032) seconds [11:58:24:INFO] Test Suite 'apprtcmobile_tests.app' passed at 2019-04-17 23:58:23.940. [11:58:24:INFO] Executed 10 tests, with 0 failures (0 unexpected) in 1.781 (1.817) seconds [11:58:24:INFO] Test Suite 'All tests' passed at 2019-04-17 23:58:23.941. [11:58:24:INFO] Executed 10 tests, with 0 failures (0 unexpected) in 1.781 (1.819) seconds [11:58:24:INFO] [001:808] [38691] (rtc_event_log_impl.cc:211): Stopping WebRTC event log. [11:58:24:INFO] [001:808] [38691] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped. [11:58:24:INFO] [001:810] [79375] (peer_connection.cc:4173): Session: 3593490849194371550 Old state: kStable New state: kClosed [11:58:24:INFO] [001:810] [38691] (audio_device_module_ios.mm:563): StopRecording [11:58:24:INFO] [001:810] [38691] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording [11:58:24:INFO] [001:810] [38691] (audio_device_buffer.cc:153): StopRecording [11:58:24:INFO] [001:816] [38691] (audio_device_buffer.cc:176): total recording time: 310 [11:58:24:INFO] [001:816] [38691] (audio_device_module_ios.mm:567): output: 0 [11:58:24:INFO] [001:816] [38691] (webrtc_voice_engine.cc:1962): SetOutputVolume() to 0 for recv stream with ssrc 3459412627 [11:58:24:INFO] [001:816] [38691] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 715828121, options: nullptr, source = nullptr) [11:58:24:INFO] [001:816] [38691] (webrtc_video_engine.cc:1304): SetSink: ssrc:2757358605 nullptr [11:58:24:INFO] [001:816] [38691] (channel.cc:563): Channel disabled [11:58:24:INFO] [001:816] [38691] (video_send_stream.cc:160): VideoSendStream::Stop [11:58:24:INFO] [001:816] [38691] (channel.cc:996): Changing video state, send=0 [11:58:24:INFO] [001:816] [35083] (video_send_stream_impl.cc:417): VideoSendStream::Stop [11:58:24:INFO] [001:816] [35083] (rtp_rtcp_impl.cc:360): Failed to send RTCP BYE [11:58:24:INFO] [001:816] [35083] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps [11:58:24:INFO] [001:816] [38691] (video_send_stream.cc:160): VideoSendStream::Stop [11:58:24:INFO] [001:817] [11011] (video_stream_encoder.cc:1568): Video suspend state changed to: suspended [11:58:24:INFO] [001:823] [35083] (video_send_stream_impl.cc:417): VideoSendStream::Stop [11:58:24:INFO] [001:823] [35083] (video_send_stream_impl.cc:417): VideoSendStream::Stop [11:58:24:INFO] [001:823] [35083] (video_send_stream_impl.cc:338): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [715828121], 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: [993092192], payload_type: 97}, c_name: evBadRFH64bGTYuc}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [11:58:24:INFO] [001:842] [38691] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [11:58:24:INFO] [001:842] [38691] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [11:58:24:INFO] [001:842] [38691] (send_statistics_proxy.cc:655): Frames encoded 0 [11:58:24:INFO] WebRTC.Video.DroppedFrames.Capturer 0 [11:58:24:INFO] WebRTC.Video.DroppedFrames.EncoderQueue 0 [11:58:24:INFO] WebRTC.Video.DroppedFrames.Encoder 0 [11:58:24:INFO] WebRTC.Video.DroppedFrames.Ratelimiter 0 [11:58:24:INFO] [001:842] [38691] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [11:58:24:INFO] [001:842] [38691] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [11:58:24:INFO] [001:843] [38691] (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: 2757358605, local_ssrc: 715828121, 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: 2295916538, 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} [11:58:24:INFO] [001:871] [38691] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [11:58:24:INFO] [001:871] [38691] (receive_statistics_proxy.cc:487): Frames decoded 0 [11:58:24:INFO] WebRTC.Video.DroppedFrames.Receiver 0 [11:58:24:INFO] [11:58:24:INFO] [001:872] [38691] (channel.cc:162): Destroyed channel: 1 [11:58:24:INFO] [001:872] [38691] (message_queue.cc:517): Message took 56ms to dispatch. Posted from: DestroyVideoChannel@../../pc/channel_manager.cc:279 [11:58:24:INFO] [001:872] [38691] (channel.cc:563): Channel disabled [11:58:24:INFO] [001:872] [38691] (audio_device_module_ios.mm:532): StopPlayout [11:58:24:INFO] [001:872] [38691] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout [11:58:24:INFO] [001:872] [38691] (audio_device_buffer.cc:139): StopPlayout [11:58:24:INFO] [001:872] [38691] (audio_device_buffer.cc:145): total playout time: 512 [11:58:24:INFO] [001:872] [38691] (audio_device_module_ios.mm:536): output: 0 [11:58:24:INFO] [001:872] [38691] (channel.cc:863): Changing voice state, recv=0 send=0 [11:58:24:INFO] [001:872] [38691] (webrtc_voice_engine.cc:1823): RemoveSendStream: 2481983948 [11:58:24:INFO] [001:872] [38691] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [11:58:24:INFO] [001:872] [38691] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [11:58:24:INFO] [001:872] [38691] (audio_send_stream.cc:179): ~AudioSendStream: 2481983948 [11:58:24:INFO] [001:872] [38691] (webrtc_voice_engine.cc:1900): RemoveRecvStream: 3459412627 [11:58:24:INFO] [001:893] [38691] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [11:58:24:INFO] [001:893] [38691] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [11:58:24:INFO] [001:893] [38691] (audio_receive_stream.cc:140): ~AudioReceiveStream: 3459412627 [11:58:24:INFO] [001:893] [12551] (paced_sender.cc:106): PacedSender paused. [11:58:24:INFO] [001:894] [12551] (control_handler.cc:77): Bitrate estimate state changed, BWE: 300 kbps. [11:58:24:INFO] [001:894] [38691] (channel.cc:162): Destroyed channel: 0 [11:58:24:INFO] [001:894] [49427] (openssl_stream_adapter.cc:896): Cleanup [11:58:24:INFO] [001:904] [38691] (paced_sender.cc:418): ProcessThreadAttached 0x0 [11:58:24:INFO] [001:904] [38691] (paced_sender.cc:418): ProcessThreadAttached 0x0 [11:58:24:INFO] [001:905] [38691] (rtc_event_log_impl.cc:211): Stopping WebRTC event log. [11:58:24:INFO] [001:905] [38691] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped. [11:58:24:INFO] [001:905] [79375] (peer_connection.cc:6776): Usage signature is 4088 [11:58:24:INFO] [001:905] [79375] (message_queue.cc:517): Message took 97ms to dispatch. Posted from: Close@../../api/peer_connection_proxy.h:140 [11:58:24:INFO] [001:905] [79375] (peer_connection.cc:915): Session: 3593490849194371550 is destroyed. [11:58:24:INFO] [001:905] [79375] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue. [11:58:24:INFO] [001:906] [38691] (webrtc_video_engine.cc:474): WebRtcVideoEngine::~WebRtcVideoEngine [11:58:24:INFO] [001:906] [38691] (webrtc_voice_engine.cc:205): WebRtcVoiceEngine::~WebRtcVoiceEngine [11:58:24:INFO] [001:906] [38691] (audio_device_module_ios.mm:532): StopPlayout [11:58:24:INFO] [001:906] [38691] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout [11:58:24:INFO] [001:906] [38691] (audio_device_module_ios.mm:536): output: 0 [11:58:24:INFO] [001:908] [38691] (audio_device_module_ios.mm:563): StopRecording [11:58:24:INFO] [001:908] [38691] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording [11:58:24:INFO] [001:908] [38691] (audio_device_module_ios.mm:567): output: 0 [11:58:24:INFO] [001:908] [38691] (audio_device_module_ios.mm:581): RegisterAudioCallback [11:58:24:INFO] [001:908] [38691] (audio_device_buffer.cc:81): RegisterAudioCallback [11:58:24:INFO] [001:908] [38691] (audio_device_module_ios.mm:92): Terminate [11:58:24:INFO] [001:908] [38691] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate [11:58:24:INFO] [001:908] [38691] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout [11:58:24:INFO] [001:908] [38691] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording [11:58:24:INFO] [001:910] [38691] (audio_device_module_ios.mm:55): ~AudioDeviceModuleIOS [11:58:24:INFO] [001:910] [38691] (audio_device_buffer.cc:75): AudioDeviceBuffer::~dtor [11:58:24:INFO] [001:910] [38691] (audio_device_ios.mm:127): AudioDeviceIOS::~dtor<NSThread: 0x6000039e4900>{number = 3, name = (null)} [11:58:24:INFO] [001:910] [38691] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate [11:58:24:INFO] [001:910] [38691] (fine_audio_buffer.cc:46): ~FineAudioBuffer [11:58:24:INFO] [001:917] [38691] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:347 Uninitialize): Unintializing audio unit. [11:58:24:INFO] [001:918] [38691] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:354 Uninitialize): Uninitialized audio unit. [11:58:24:INFO] [001:918] [38691] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:459 DisposeAudioUnit): Disposing audio unit. [11:58:24:INFO] [11:58:24:INFO] [11:58:24:INFO] Test session results and logs: [11:58:24:INFO] /Users/chrome-bot/Library/Developer/Xcode/DerivedData/temporary-cfnhutywnctkhyghcnwsgfzltxjy/Logs/Test/Test-Transient Testing-2019.04.17_23-58-03--0700.xcresult [11:58:24:INFO] [11:58:24:INFO] 2019-04-17 23:58:24.372 xcodebuild[5961:141305] [MT] IDETestOperationsObserverDebug: 21.192 elapsed -- Testing started completed. [11:58:24:INFO] 2019-04-17 23:58:24.372 xcodebuild[5961:141305] [MT] IDETestOperationsObserverDebug: 0.000 sec, +0.000 sec -- start [11:58:24:INFO] 2019-04-17 23:58:24.372 xcodebuild[5961:141305] [MT] IDETestOperationsObserverDebug: 21.192 sec, +21.192 sec -- end [11:58:24:INFO] ** TEST EXECUTE SUCCEEDED ** [11:58:24:INFO] [11:58:30:DEBUG] Finished print_process_output. [11:58:30:INFO] Waiting for test process to terminate. [11:58:30:INFO] Test process terminated. [11:58:30:DEBUG] Setting sigterm handler. [11:58:30:DEBUG] Stdout flushed after test process. [11:58:30:DEBUG] Processing test results. [11:58:30:INFO] /b/s/w/ir/src/out/Debug-iphonesimulator/iossim returned 0 [11:58:30:DEBUG] Extracting test data. [11:58:30:DEBUG] Retrieving crash reports. [11:58:30:DEBUG] Retrieving derived data. [11:58:30:DEBUG] Making desktop screenshots. [11:58:30:DEBUG] Killing simulators. [11:58:35:DEBUG] Wiping simulator. Device wiped. [11:58:37:DEBUG] End of tear_down. 2019-04-17 23:59:10,321 - root: [DEBUG] Contents of task_output_dir: ['/b/s/w/ir/k/recipe_cleanup/0_BfJvP/44480e293dba6910'] 2019-04-17 23:59:10,322 - root: [WARNING] Expected output.json file missing: set(['/b/s/w/ir/k/recipe_cleanup/0_BfJvP/44480e293dba6910/output.json']) Found: [] Expected: ['/b/s/w/ir/k/recipe_cleanup/0_BfJvP/44480e293dba6910/output.json'] 2019-04-17 23:59:10,322 - root: [WARNING] No shard json files found in task_output_dir: '/b/s/w/ir/k/recipe_cleanup/0_BfJvP' Found ['/b/s/w/ir/k/recipe_cleanup/0_BfJvP/44480e293dba6910'] 2019-04-17 23:59:10,322 - root: [DEBUG] Found shard_json_files: ['/b/s/w/ir/k/recipe_cleanup/0_BfJvP/44480e293dba6910/output.json'] 2019-04-17 23:59:10,323 - root: [INFO] merge_cmd: /b/s/w/ir/cache/vpython/5b0713/bin/python /b/s/w/ir/kitchen-checkout/build/scripts/slave/recipe_modules/chromium_swarming/resources/noop_merge.py --summary-json /b/s/w/ir/tmp/t/tmptoSpiZ.json --task-output-dir /b/s/w/ir/k/recipe_cleanup/0_BfJvP -o /b/s/w/ir/tmp/t/tmpginBlC.json Running ['/b/s/w/ir/cache/vpython/5b0713/bin/python', '/b/s/w/ir/kitchen-checkout/build/scripts/slave/recipe_modules/chromium_swarming/resources/noop_merge.py', '--summary-json', '/b/s/w/ir/tmp/t/tmptoSpiZ.json', '--task-output-dir', '/b/s/w/ir/k/recipe_cleanup/0_BfJvP', '-o', '/b/s/w/ir/tmp/t/tmpginBlC.json'] in None (env: None) Command ['/b/s/w/ir/cache/vpython/5b0713/bin/python', '/b/s/w/ir/kitchen-checkout/build/scripts/slave/recipe_modules/chromium_swarming/resources/noop_merge.py', '--summary-json', '/b/s/w/ir/tmp/t/tmptoSpiZ.json', '--task-output-dir', '/b/s/w/ir/k/recipe_cleanup/0_BfJvP', '-o', '/b/s/w/ir/tmp/t/tmpginBlC.json'] returned exit code 0