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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/tmpqYi_g6.json --task-output-dir /b/s/w/ir/k/recipe_cleanup/0xm25_5 --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/tmpvsmiDamerge_script_log --merge-additional-args '[]' --summary-json-file /b/s/w/ir/tmp/t/tmpYm4t4M.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/tmpZmeDUv.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/tmpqYi_g6.json', '--task-output-dir', '/b/s/w/ir/k/recipe_cleanup/0xm25_5', '--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/tmpvsmiDamerge_script_log', '--merge-additional-args', '[]', '--summary-json-file', '/b/s/w/ir/tmp/t/tmpYm4t4M.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/tmpZmeDUv.json'] env: {} env_prefixes: {'PATH': '/b/s/w/ir/cache/swarming_client'} env_suffixes: {} infra_step: False name_tokens: ('apprtcmobile_tests (iPhone 6s iOS 10.3)',) 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.z4oIGtMacA/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: 52641 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/8915922857259736720 LOGDOG_STREAM_PROJECT: webrtc LOGDOG_STREAM_SERVER_PATH: unix:/b/s/w/ir/tmp/ld.sock LOGNAME: chrome-bot LUCI_CONTEXT: /b/s/w/ith7Gv6D/luci_context.454417475 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.nkF5yGwNVn/Listeners SWARMING_BOT_ID: build188-m9 SWARMING_HEADLESS: 1 SWARMING_SERVER: https://chromium-swarm.appspot.com SWARMING_TASK_ID: 444476d8da039f11 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 07:16:48,935 - root: [DEBUG] Using task_output_dir: '/b/s/w/ir/k/recipe_cleanup/0xm25_5' 2019-04-17 07:16:48,936 - root: [WARNING] task_output_dir '/b/s/w/ir/k/recipe_cleanup/0xm25_5' already exists! 2019-04-17 07:16:48,936 - root: [WARNING] task_output_dir existing content: [] 2019-04-17 07:16:48,936 - 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/tmpZmeDUv.json -output-dir /b/s/w/ir/k/recipe_cleanup/0xm25_5 -task-summary-json /b/s/w/ir/tmp/t/tmpYm4t4M.json [D2019-04-17T07:16:48.950438-07:00 55407 0 auth.go:1265] Minting a new token {"key":"luci_ctx/36ec53da5d60cbf0376e059f17f79b1d2dd85d7815e623df4f893b8935daf76b", "scopes":"https://www.googleapis.com/auth/userinfo.email"} [D2019-04-17T07:16:48.950515-07:00 55407 0 luci_ctx.go:138] POST http://127.0.0.1:52630/rpc/LuciLocalAuthService.GetOAuthToken {"key":"luci_ctx/36ec53da5d60cbf0376e059f17f79b1d2dd85d7815e623df4f893b8935daf76b", "scopes":"https://www.googleapis.com/auth/userinfo.email"} [D2019-04-17T07:16:48.952092-07:00 55407 0 auth.go:1222] Token expires in 21m45.047934s {"key":"luci_ctx/36ec53da5d60cbf0376e059f17f79b1d2dd85d7815e623df4f893b8935daf76b", "scopes":"https://www.googleapis.com/auth/userinfo.email"} [D2019-04-17T07:16:50.371282-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:16:51.788146-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:16:53.317462-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:16:54.856510-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:16:56.697519-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:16:58.218326-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:16:59.847890-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:01.463590-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:03.141214-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:05.333865-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:07.258177-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:09.249200-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:11.254605-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:13.281807-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:15.454957-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:17.724228-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:20.216273-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:22.729878-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:25.139729-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:28.250635-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:30.923165-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:33.817950-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:37.473579-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:40.463269-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:43.531917-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:46.715088-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:50.004051-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:53.359959-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:17:56.827893-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 [D2019-04-17T07:18:00.371605-07:00 55407 0 collect.go:344] Waiting task_id: 44447987e87a1210 44447987e87a1210: exit 0 [I 2019-04-17 07:16:39] About to install Xcode 10l232m in Xcode.app for ios [07:16:41:INFO] Using Xcode version 10.0 build 10L232m at /b/s/w/ir/Xcode.app/Contents/Developer Device wiped. [07:16:50:DEBUG] Setting sigterm handler. [07:16:51:INFO] User defaults from command line: [07:16:51:INFO] IDETestRunSpecificationPath = /var/folders/2j/22s2gz0s7hn48k32d47clxf80000gm/T/5C0352E5-956C-4764-A71D-FF81C8EC1882 [07:16:51:INFO] [07:16:51:INFO] Testing started on 'iPhone 6s' [07:16:51:INFO] 2019-04-17 07:16:51.797 xcodebuild[18734:512759] IDETestOperationsObserverDebug: Writing diagnostic log for test session to: [07:16:51:INFO] /Users/chrome-bot/Library/Developer/Xcode/DerivedData/temporary-cysxxxcydjplqwhhkapbtbilmqwj/Logs/Test/Test-Transient Testing-2019.04.17_07-16-51--0700.xcresult/1_Test/Diagnostics/apprtcmobile_tests_module-AD349138-467A-4773-AD3B-01F467DDA5BD/apprtcmobile_tests_module-0E482AE9-4D25-4F4D-BBB8-A12301C21FA9/Session-apprtcmobile_tests_module-2019-04-17_071651-u6fswA.log [07:16:51:INFO] 2019-04-17 07:16:51.797 xcodebuild[18734:512735] [MT] IDETestOperationsObserverDebug: (A635FF87-4DF3-4B3C-8174-4736B3982DDB) Beginning test session apprtcmobile_tests_module-A635FF87-4DF3-4B3C-8174-4736B3982DDB at 2019-04-17 07:16:51.797 with Xcode 10L232m on target <DVTiPhoneSimulator: 0x7fe7a8749150> { [07:16:51:INFO] SimDevice: iPhone 6s (6C66BB4F-DE72-417F-B3FA-71F63674AA99, iOS 10.3, Shutdown) [07:16:51:INFO] } (10.3.1 (14E8301)) [07:17:09:INFO] objc[18833]: Class RTCDispatcher is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x1064fda50) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/apprtcmobile_tests (0x1018ea330). One of the two will be used. Which one is undefined. [07:17:09:INFO] objc[18833]: Class RTCCameraPreviewView is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x1064fda78) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/apprtcmobile_tests (0x1018ea358). One of the two will be used. Which one is undefined. [07:17:09:INFO] objc[18833]: Class RTCEncodedImage is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x1064fc920) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/apprtcmobile_tests (0x1018ea3d0). One of the two will be used. Which one is undefined. [07:17:09:INFO] objc[18833]: Class RTCRtpFragmentationHeader is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x1064fc970) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/apprtcmobile_tests (0x1018ea420). One of the two will be used. Which one is undefined. [07:17:09:INFO] objc[18833]: Class RTCVideoCapturer is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x1064fc998) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/apprtcmobile_tests (0x1018ea448). One of the two will be used. Which one is undefined. [07:17:09:INFO] objc[18833]: Class RTCVideoCodecInfo is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x1064fc9e8) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/apprtcmobile_tests (0x1018ea498). One of the two will be used. Which one is undefined. [07:17:09:INFO] objc[18833]: Class RTCVideoEncoderQpThresholds is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x1064fca38) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/apprtcmobile_tests (0x1018ea4e8). One of the two will be used. Which one is undefined. [07:17:10:INFO] objc[18833]: Class RTCVideoEncoderSettings is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x1064fcab0) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/apprtcmobile_tests (0x1018ea560). One of the two will be used. Which one is undefined. [07:17:10:INFO] objc[18833]: Class RTCVideoFrame is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC (0x1064fcad8) and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/6C66BB4F-DE72-417F-B3FA-71F63674AA99/data/Containers/Bundle/Application/CE3E57B5-5986-46BF-9C0F-9E2E902E6F15/apprtcmobile_tests.app/apprtcmobile_tests (0x1018ea588). One of the two will be used. Which one is undefined. [07:17:11:INFO] Test Suite 'All tests' started at 2019-04-17 14:17:11.115 [07:17:11:INFO] Test Suite 'apprtcmobile_tests.app' started at 2019-04-17 14:17:11.116 [07:17:11:INFO] Test Suite 'ARDAppClientTest' started at 2019-04-17 14:17:11.116 [07:17:11:INFO] Test Case '-[ARDAppClientTest testSession]' started. [07:17:11:INFO] [000:000] [1027] (audio_device_module.mm:21): CreateAudioDeviceModule [07:17:11:INFO] [000:001] [1027] (audio_device_module_ios.mm:44): current platform is IOS [07:17:11:INFO] [000:001] [1027] (audio_device_module_ios.mm:45): iPhone Audio APIs will be utilized. [07:17:11:INFO] [000:013] [1027] (audio_processing_impl.cc:438): Capture analyzer activated: 0 [07:17:11:INFO] Capture post processor activated: 0 [07:17:11:INFO] Render pre processor activated: 0 [07:17:11:INFO] [000:015] [1027] (webrtc_video_engine.cc:470): WebRtcVideoEngine::WebRtcVideoEngine() [07:17:11:INFO] [000:018] [1027] (webrtc_voice_engine.cc:196): WebRtcVoiceEngine::WebRtcVoiceEngine [07:17:11:INFO] [000:023] [25603] (webrtc_voice_engine.cc:219): WebRtcVoiceEngine::Init [07:17:11:INFO] [000:023] [25603] (webrtc_voice_engine.cc:227): Supported send codecs in order of preference: [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:230): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:230): ISAC/16000/1 (103) [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:230): ISAC/32000/1 (104) [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:230): G722/8000/1 (9) [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:230): ILBC/8000/1 (102) [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:230): PCMU/8000/1 (0) [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:230): PCMA/8000/1 (8) [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:230): CN/32000/1 (106) [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:230): CN/16000/1 (105) [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:230): CN/8000/1 (13) [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:230): telephone-event/48000/1 (110) [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:230): telephone-event/32000/1 (112) [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:230): telephone-event/16000/1 (113) [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:230): telephone-event/8000/1 (126) [07:17:11:INFO] [000:030] [25603] (webrtc_voice_engine.cc:233): Supported recv codecs in order of preference: [07:17:11:INFO] [000:031] [25603] (webrtc_voice_engine.cc:236): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [07:17:11:INFO] [000:031] [25603] (webrtc_voice_engine.cc:236): ISAC/16000/1 (103) [07:17:11:INFO] [000:031] [25603] (webrtc_voice_engine.cc:236): ISAC/32000/1 (104) [07:17:11:INFO] [000:031] [25603] (webrtc_voice_engine.cc:236): G722/8000/1 (9) [07:17:11:INFO] [000:031] [25603] (webrtc_voice_engine.cc:236): ILBC/8000/1 (102) [07:17:11:INFO] [000:031] [25603] (webrtc_voice_engine.cc:236): PCMU/8000/1 (0) [07:17:11:INFO] [000:031] [25603] (webrtc_voice_engine.cc:236): PCMA/8000/1 (8) [07:17:11:INFO] [000:031] [25603] (webrtc_voice_engine.cc:236): CN/32000/1 (106) [07:17:11:INFO] [000:031] [25603] (webrtc_voice_engine.cc:236): CN/16000/1 (105) [07:17:11:INFO] [000:031] [25603] (webrtc_voice_engine.cc:236): CN/8000/1 (13) [07:17:11:INFO] [000:031] [25603] (webrtc_voice_engine.cc:236): telephone-event/48000/1 (110) [07:17:11:INFO] [000:031] [25603] (webrtc_voice_engine.cc:236): telephone-event/32000/1 (112) [07:17:11:INFO] [000:031] [25603] (webrtc_voice_engine.cc:236): telephone-event/16000/1 (113) [07:17:11:INFO] [000:031] [25603] (webrtc_voice_engine.cc:236): telephone-event/8000/1 (126) [07:17:11:INFO] [000:031] [25603] (audio_device_module_ios.mm:69): Init [07:17:11:INFO] [000:032] [25603] (audio_device_buffer.cc:63): AudioDeviceBuffer::ctor [07:17:11:INFO] [000:033] [25603] (audio_device_ios.mm:117): AudioDeviceIOS::ctor<NSThread: 0x600001278c80>{number = 3, name = (null)} [07:17:11:INFO] [000:033] [25603] (audio_device_module_ios.mm:49): AttachAudioBuffer [07:17:11:INFO] [000:033] [25603] (audio_device_ios.mm:133): AudioDeviceIOS::AttachAudioBuffer [07:17:11:INFO] [000:033] [25603] (audio_device_ios.mm:140): AudioDeviceIOS::Init [07:17:11:INFO] [000:033] [25603] (audio_device_ios.mm:85): LogDeviceInfo [07:17:11:INFO] [000:033] [25603] (audio_device_ios.mm:87): system name: iOS [07:17:11:INFO] [000:033] [25603] (audio_device_ios.mm:88): system version: 10.3.1 [07:17:11:INFO] [000:033] [25603] (audio_device_ios.mm:89): device type: iPhone [07:17:11:INFO] [000:033] [25603] (audio_device_ios.mm:90): device name: x86_64 [07:17:11:INFO] [000:033] [25603] (audio_device_ios.mm:91): process name: apprtcmobile_tests [07:17:11:INFO] [000:033] [25603] (audio_device_ios.mm:92): process ID: 18833 [07:17:11:INFO] [000:034] [25603] (audio_device_ios.mm:93): OS version: Version 10.3.1 (Build 14E8301) [07:17:11:INFO] [000:034] [25603] (audio_device_ios.mm:94): processing cores: 4 [07:17:11:INFO] [000:034] [25603] (audio_device_ios.mm:95): low power mode: 0 [07:17:11:INFO] [000:034] [25603] (audio_device_ios.mm:97): TARGET_IPHONE_SIMULATOR is defined [07:17:11:INFO] [000:034] [25603] (audio_device_ios.mm:99): DeviceIsSimulator: 1 [07:17:11:INFO] [000:034] [25603] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer [07:17:11:INFO] [000:034] [25603] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000) [07:17:11:INFO] [000:034] [25603] (audio_device_buffer.cc:206): SetPlayoutChannels(1) [07:17:11:INFO] [000:035] [25603] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000) [07:17:11:INFO] [000:035] [25603] (audio_device_buffer.cc:200): SetRecordingChannels(1) [07:17:11:INFO] [000:035] [25603] (audio_device_module_ios.mm:406): SetPlayoutDevice(0) [07:17:11:INFO] [000:035] [25603] (audio_device_ios.mm:1006): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetPlayoutDevice(uint16_t): Not implemented [07:17:11:INFO] [000:035] [25603] (audio_device_module_ios.mm:108): InitSpeaker [07:17:11:INFO] [000:035] [25603] (audio_device_module_ios.mm:310): StereoPlayoutIsAvailable [07:17:11:INFO] [000:035] [25603] (audio_device_module_ios.mm:317): output: 0 [07:17:11:INFO] [000:035] [25603] (audio_device_module_ios.mm:322): SetStereoPlayout(0) [07:17:11:INFO] [000:035] [25603] (audio_device_ios.mm:1059): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetStereoPlayout(bool): Not implemented [07:17:11:INFO] [000:035] [25603] (audio_device_module_ios.mm:330): stereo playout is not supported [07:17:11:INFO] [000:035] [25603] (adm_helpers.cc:57): Failed to set stereo playout mode. [07:17:11:INFO] [000:035] [25603] (audio_device_module_ios.mm:468): SetRecordingDevice(0) [07:17:11:INFO] [000:035] [25603] (audio_device_ios.mm:1108): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetRecordingDevice(uint16_t): Not implemented [07:17:11:INFO] [000:035] [25603] (audio_device_module_ios.mm:114): InitMicrophone [07:17:11:INFO] [000:036] [25603] (audio_device_module_ios.mm:277): StereoRecordingIsAvailable [07:17:11:INFO] [000:036] [25603] (audio_device_module_ios.mm:284): output: 0 [07:17:11:INFO] [000:036] [25603] (audio_device_module_ios.mm:289): SetStereoRecording(0) [07:17:11:INFO] [000:036] [25603] (adm_helpers.cc:77): Failed to set stereo recording mode. [07:17:11:INFO] [000:036] [25603] (apm_helpers.cc:32): Setting AGC mode to 2 [07:17:11:INFO] [000:036] [25603] (audio_processing_impl.cc:727): Highpass filter activated: 0 [07:17:11:INFO] [000:036] [25603] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:11:INFO] [000:036] [25603] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:11:INFO] [000:037] [25603] (audio_device_module_ios.mm:581): RegisterAudioCallback [07:17:11:INFO] [000:037] [25603] (audio_device_buffer.cc:81): RegisterAudioCallback [07:17:11:INFO] [000:038] [25603] (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, } [07:17:11:INFO] [000:038] [25603] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [07:17:11:INFO] [000:038] [25603] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [07:17:11:INFO] [000:038] [25603] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [07:17:11:INFO] [000:038] [25603] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [07:17:11:INFO] [000:038] [25603] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [07:17:11:INFO] [000:038] [25603] (audio_device_module_ios.mm:600): output: 0 [07:17:11:INFO] [000:038] [25603] (audio_processing_impl.cc:727): Highpass filter activated: 0 [07:17:11:INFO] [000:038] [25603] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:11:INFO] [000:038] [25603] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:11:INFO] [000:038] [25603] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [07:17:11:INFO] [000:038] [25603] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [07:17:11:INFO] [000:038] [25603] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [07:17:11:INFO] [000:038] [25603] (audio_device_module_ios.mm:616): output: 0 [07:17:11:INFO] [000:039] [25603] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [07:17:11:INFO] [000:039] [25603] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [07:17:11:INFO] [000:039] [25603] (audio_device_module_ios.mm:632): output: 0 [07:17:11:INFO] [000:039] [25603] (audio_processing_impl.cc:727): Highpass filter activated: 0 [07:17:11:INFO] [000:039] [25603] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:11:INFO] [000:039] [25603] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:11:INFO] [000:039] [25603] (apm_helpers.cc:62): NS set to 0 [07:17:11:INFO] [000:039] [25603] (webrtc_voice_engine.cc:447): Stereo swapping enabled? 0 [07:17:11:INFO] [000:039] [25603] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [07:17:11:INFO] [000:039] [25603] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [07:17:11:INFO] [000:039] [25603] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [07:17:11:INFO] [000:039] [25603] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [07:17:11:INFO] [000:039] [25603] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [07:17:11:INFO] [000:039] [25603] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [07:17:11:INFO] [000:039] [25603] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [07:17:11:INFO] [000:039] [25603] (webrtc_voice_engine.cc:511): Setting AGC to 0 [07:17:11:INFO] [000:039] [25603] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [07:17:11:INFO] [000:039] [25603] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:11:INFO] [000:040] [25603] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:11:INFO] [000:040] [25603] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:11:INFO] (RTCLogging.mm:33): (ARDAppClient.m:270 -[ARDAppClient connectToRoomWithId:settings:isLoopback:]_block_invoke): Joined room:testRoom on room server. [07:17:11:INFO] [000:048] [1027] (openssl_identity.cc:44): Making key pair [07:17:11:INFO] [000:391] [1027] (openssl_identity.cc:92): Returning key pair [07:17:11:INFO] [000:392] [1027] (openssl_certificate.cc:58): Making certificate for WebRTC [07:17:11:INFO] [000:400] [1027] (openssl_certificate.cc:108): Returning certificate [07:17:11:INFO] [000:402] [1027] (RTCCertificate.mm:63): CERT PEM [07:17:11:INFO] [000:402] [1027] (RTCCertificate.mm:64): -----BEGIN CERTIFICATE----- [07:17:11:INFO] MIIBnjCCAQegAwIBAgIJAPmHU65cQQ5BMA0GCSqGSIb3DQEBCwUAMBExDzANBgNV [07:17:11:INFO] BAMMBldlYlJUQzAeFw0xOTA0MTYxNDE3MTFaFw0xOTA0MTcxNDE4NTFaMBExDzAN [07:17:11:INFO] BgNVBAMMBldlYlJUQzCBnzANBgkqhkiG9w0BAQEFAAOBjQAwgYkCgYEAvBfoPNkb [07:17:11:INFO] MFyv2WNjdMAKJOY6eY+iLo3HrcAr+BQNss2YuE9FwFX6l2HxxMj6c14iyvqlJFVV [07:17:11:INFO] /qF5yoBQnSvarv8pDWlhnO7lQXAzlz7GpGB4OR6zimtZMmdLP0NeAMNu+rYCfz3S [07:17:11:INFO] jxb8XnmK+QBDExiWUd9FEjqBQDSXakP5Vv8CAwEAATANBgkqhkiG9w0BAQsFAAOB [07:17:11:INFO] gQBmptyeGgnnNt1fX6hMUguFA8qjVbyQf1AqQVkW/XNLTro456U7XqvTnHb00mgg [07:17:11:INFO] tuPj4FTeQ9dODlzo8hFo571XGj/pC9pF9h6u73YDPEw7vS5aZHgNX8PLghwgBp8X [07:17:11:INFO] 6JQyxATGbe5/mpOMVdvdiZjmqiYc9++D5nv8ipmE2AMr9A== [07:17:11:INFO] -----END CERTIFICATE----- [07:17:11:INFO] [07:17:11:INFO] [000:402] [1027] (RTCConfiguration.mm:213): Have configured cert - using it. [07:17:11:INFO] [000:404] [1027] (RTCConfiguration.mm:218): Created cert from PEM strings. [07:17:11:INFO] [000:408] [25603] (rtc_event_log_impl.cc:71): Creating legacy encoder for RTC event log. [07:17:11:INFO] [000:409] [25603] (peer_connection_factory.cc:407): Using default network controller factory [07:17:11:INFO] [000:413] [25603] (bitrate_prober.cc:66): Bandwidth probing enabled, set to inactive [07:17:11:INFO] [000:415] [25603] (paced_sender.cc:418): ProcessThreadAttached 0x29f13ed0 [07:17:11:INFO] [000:415] [25603] (cpu_info.cc:49): Available number of cores: 4 [07:17:11:INFO] [000:418] [25603] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85 [07:17:11:INFO] [000:418] [25603] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating. [07:17:11:INFO] [000:431] [38403] (webrtc_session_description_factory.cc:314): Using certificate supplied to the constructor. [07:17:11:INFO] [000:433] [38403] (peer_connection.cc:1405): Adding audio transceiver in response to a call to AddTrack. [07:17:11:INFO] (RTCLogging.mm:33): (ARDAppClient.m:396 -[ARDAppClient peerConnectionShouldNegotiate:]): WARNING: Renegotiation needed but unimplemented. [07:17:11:INFO] [000:437] [1027] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x600000202550): created DTMF sender: RTCDtmfSender { [07:17:11:INFO] remainingTones: [07:17:11:INFO] duration: 0.100000 sec [07:17:11:INFO] interToneGap: 0.050000 sec [07:17:11:INFO] } [07:17:11:INFO] [000:438] [1027] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600000226200): created sender: RTCRtpSender { [07:17:11:INFO] senderId: ARDAMSa0 [07:17:11:INFO] } [07:17:11:INFO] [000:439] [1027] (RTCLogging.mm:33): (RTCFileVideoCapturer.m:103 -[RTCFileVideoCapturer stopCapture]): File capturer stopped. [07:17:11:INFO] [000:439] [38403] (peer_connection.cc:1405): Adding video transceiver in response to a call to AddTrack. [07:17:11:INFO] [000:440] [38403] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [07:17:11:INFO] [000:440] [1027] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600000226080): created sender: RTCRtpSender { [07:17:11:INFO] senderId: ARDAMSv0 [07:17:11:INFO] } [07:17:11:INFO] [000:441] [1027] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x6080002028f0): created DTMF sender: RTCDtmfSender { [07:17:11:INFO] remainingTones: [07:17:11:INFO] duration: 0.100000 sec [07:17:11:INFO] interToneGap: 0.050000 sec [07:17:11:INFO] } [07:17:11:INFO] [000:441] [1027] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60800023e9c0): created sender: RTCRtpSender { [07:17:11:INFO] senderId: ARDAMSa0 [07:17:11:INFO] } [07:17:11:INFO] [000:441] [1027] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x60000005d5e0): created receiver: RTCRtpReceiver { [07:17:11:INFO] receiverId: bcfa7252-dd67-479d-8b38-7be9a4d8d932 [07:17:11:INFO] } [07:17:11:INFO] [000:441] [1027] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x60800005adc0): created transceiver: RTCRtpTransceiver { [07:17:11:INFO] sender: RTCRtpSender { [07:17:11:INFO] senderId: ARDAMSa0 [07:17:11:INFO] } [07:17:11:INFO] receiver: RTCRtpReceiver { [07:17:11:INFO] receiverId: bcfa7252-dd67-479d-8b38-7be9a4d8d932 [07:17:11:INFO] } [07:17:11:INFO] } [07:17:11:INFO] [000:441] [38403] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [07:17:11:INFO] [000:442] [1027] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60800023e9e0): created sender: RTCRtpSender { [07:17:11:INFO] senderId: ARDAMSv0 [07:17:11:INFO] } [07:17:11:INFO] [000:442] [1027] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x60000005d640): created receiver: RTCRtpReceiver { [07:17:11:INFO] receiverId: ce5f386a-9d17-4ef2-b677-7dccf8793da3 [07:17:11:INFO] } [07:17:11:INFO] [000:442] [1027] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x60800005ae20): created transceiver: RTCRtpTransceiver { [07:17:11:INFO] sender: RTCRtpSender { [07:17:11:INFO] senderId: ARDAMSv0 [07:17:11:INFO] } [07:17:11:INFO] receiver: RTCRtpReceiver { [07:17:11:INFO] receiverId: ce5f386a-9d17-4ef2-b677-7dccf8793da3 [07:17:11:INFO] } [07:17:11:INFO] } [07:17:11:INFO] [000:448] [38403] (media_session.cc:335): Duplicate id found. Reassigning from 102 to 127 [07:17:11:INFO] [000:448] [38403] (media_session.cc:335): Duplicate id found. Reassigning from 104 to 125 [07:17:11:INFO] [000:448] [38403] (media_session.cc:335): Duplicate id found. Reassigning from 106 to 124 [07:17:11:INFO] [000:448] [38403] (media_session.cc:335): Duplicate id found. Reassigning from 103 to 123 [07:17:11:INFO] [000:448] [38403] (media_session.cc:335): Duplicate id found. Reassigning from 105 to 122 [07:17:11:INFO] [000:449] [38403] (media_session.cc:335): Duplicate id found. Reassigning from 1 to 14 [07:17:11:INFO] [000:449] [38403] (media_session.cc:335): Duplicate id found. Reassigning from 2 to 13 [07:17:11:INFO] [000:449] [38403] (media_session.cc:335): Duplicate id found. Reassigning from 3 to 12 [07:17:11:INFO] [000:449] [38403] (media_session.cc:335): Duplicate id found. Reassigning from 5 to 11 [07:17:11:INFO] [000:455] [25603] (rtc_event_log_impl.cc:186): Starting WebRTC event log. (Timestamp, UTC) = (12617242394, 1555510631581976). [07:17:11:INFO] [000:456] [1027] (audio_device_module.mm:21): CreateAudioDeviceModule [07:17:11:INFO] [000:456] [1027] (audio_device_module_ios.mm:44): current platform is IOS [07:17:11:INFO] [000:456] [1027] (audio_device_module_ios.mm:45): iPhone Audio APIs will be utilized. [07:17:11:INFO] [000:457] [1027] (audio_processing_impl.cc:438): Capture analyzer activated: 0 [07:17:11:INFO] Capture post processor activated: 0 [07:17:11:INFO] Render pre processor activated: 0 [07:17:11:INFO] [000:458] [1027] (webrtc_video_engine.cc:470): WebRtcVideoEngine::WebRtcVideoEngine() [07:17:11:INFO] [000:458] [1027] (webrtc_voice_engine.cc:196): WebRtcVoiceEngine::WebRtcVoiceEngine [07:17:11:INFO] [000:458] [37379] (webrtc_voice_engine.cc:219): WebRtcVoiceEngine::Init [07:17:11:INFO] [000:458] [37379] (webrtc_voice_engine.cc:227): Supported send codecs in order of preference: [07:17:11:INFO] [000:458] [37379] (webrtc_voice_engine.cc:230): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [07:17:11:INFO] [000:458] [37379] (webrtc_voice_engine.cc:230): ISAC/16000/1 (103) [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:230): ISAC/32000/1 (104) [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:230): G722/8000/1 (9) [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:230): ILBC/8000/1 (102) [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:230): PCMU/8000/1 (0) [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:230): PCMA/8000/1 (8) [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:230): CN/32000/1 (106) [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:230): CN/16000/1 (105) [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:230): CN/8000/1 (13) [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:230): telephone-event/48000/1 (110) [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:230): telephone-event/32000/1 (112) [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:230): telephone-event/16000/1 (113) [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:230): telephone-event/8000/1 (126) [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:233): Supported recv codecs in order of preference: [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:236): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [07:17:11:INFO] [000:459] [37379] (webrtc_voice_engine.cc:236): ISAC/16000/1 (103) [07:17:11:INFO] [000:460] [37379] (webrtc_voice_engine.cc:236): ISAC/32000/1 (104) [07:17:11:INFO] [000:460] [37379] (webrtc_voice_engine.cc:236): G722/8000/1 (9) [07:17:11:INFO] [000:460] [37379] (webrtc_voice_engine.cc:236): ILBC/8000/1 (102) [07:17:11:INFO] [000:460] [37379] (webrtc_voice_engine.cc:236): PCMU/8000/1 (0) [07:17:11:INFO] [000:460] [37379] (webrtc_voice_engine.cc:236): PCMA/8000/1 (8) [07:17:11:INFO] [000:460] [37379] (webrtc_voice_engine.cc:236): CN/32000/1 (106) [07:17:11:INFO] [000:460] [37379] (webrtc_voice_engine.cc:236): CN/16000/1 (105) [07:17:11:INFO] [000:460] [37379] (webrtc_voice_engine.cc:236): CN/8000/1 (13) [07:17:11:INFO] [000:460] [37379] (webrtc_voice_engine.cc:236): telephone-event/48000/1 (110) [07:17:11:INFO] [000:460] [37379] (webrtc_voice_engine.cc:236): telephone-event/32000/1 (112) [07:17:11:INFO] [000:460] [37379] (webrtc_voice_engine.cc:236): telephone-event/16000/1 (113) [07:17:11:INFO] [000:460] [37379] (webrtc_voice_engine.cc:236): telephone-event/8000/1 (126) [07:17:11:INFO] [000:460] [37379] (audio_device_module_ios.mm:69): Init [07:17:11:INFO] [000:461] [37379] (audio_device_buffer.cc:63): AudioDeviceBuffer::ctor [07:17:11:INFO] [000:461] [37379] (audio_device_ios.mm:117): AudioDeviceIOS::ctor<NSThread: 0x608000273480>{number = 4, name = (null)} [07:17:11:INFO] [000:461] [37379] (audio_device_module_ios.mm:49): AttachAudioBuffer [07:17:11:INFO] [000:461] [37379] (audio_device_ios.mm:133): AudioDeviceIOS::AttachAudioBuffer [07:17:11:INFO] [000:461] [37379] (audio_device_ios.mm:140): AudioDeviceIOS::Init [07:17:11:INFO] [000:461] [37379] (audio_device_ios.mm:85): LogDeviceInfo [07:17:11:INFO] [000:461] [37379] (audio_device_ios.mm:87): system name: iOS [07:17:11:INFO] [000:461] [37379] (audio_device_ios.mm:88): system version: 10.3.1 [07:17:11:INFO] [000:461] [37379] (audio_device_ios.mm:89): device type: iPhone [07:17:11:INFO] [000:461] [37379] (audio_device_ios.mm:90): device name: x86_64 [07:17:11:INFO] [000:461] [37379] (audio_device_ios.mm:91): process name: apprtcmobile_tests [07:17:11:INFO] [000:461] [37379] (audio_device_ios.mm:92): process ID: 18833 [07:17:11:INFO] [000:461] [37379] (audio_device_ios.mm:93): OS version: Version 10.3.1 (Build 14E8301) [07:17:11:INFO] [000:461] [37379] (audio_device_ios.mm:94): processing cores: 4 [07:17:11:INFO] [000:462] [37379] (audio_device_ios.mm:95): low power mode: 0 [07:17:11:INFO] [000:462] [37379] (audio_device_ios.mm:97): TARGET_IPHONE_SIMULATOR is defined [07:17:11:INFO] [000:462] [37379] (audio_device_ios.mm:99): DeviceIsSimulator: 1 [07:17:11:INFO] [000:462] [37379] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer [07:17:11:INFO] [000:462] [37379] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000) [07:17:11:INFO] [000:462] [37379] (audio_device_buffer.cc:206): SetPlayoutChannels(1) [07:17:11:INFO] [000:462] [37379] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000) [07:17:11:INFO] [000:462] [37379] (audio_device_buffer.cc:200): SetRecordingChannels(1) [07:17:11:INFO] [000:462] [37379] (audio_device_module_ios.mm:406): SetPlayoutDevice(0) [07:17:11:INFO] [000:462] [37379] (audio_device_ios.mm:1006): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetPlayoutDevice(uint16_t): Not implemented [07:17:11:INFO] [000:462] [37379] (audio_device_module_ios.mm:108): InitSpeaker [07:17:11:INFO] [000:462] [37379] (audio_device_module_ios.mm:310): StereoPlayoutIsAvailable [07:17:11:INFO] [000:462] [37379] (audio_device_module_ios.mm:317): output: 0 [07:17:11:INFO] [000:462] [37379] (audio_device_module_ios.mm:322): SetStereoPlayout(0) [07:17:11:INFO] [000:462] [37379] (audio_device_ios.mm:1059): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetStereoPlayout(bool): Not implemented [07:17:11:INFO] [000:462] [37379] (audio_device_module_ios.mm:330): stereo playout is not supported [07:17:11:INFO] [000:462] [37379] (adm_helpers.cc:57): Failed to set stereo playout mode. [07:17:11:INFO] [000:462] [37379] (audio_device_module_ios.mm:468): SetRecordingDevice(0) [07:17:11:INFO] [000:462] [37379] (audio_device_ios.mm:1108): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetRecordingDevice(uint16_t): Not implemented [07:17:11:INFO] [000:462] [37379] (audio_device_module_ios.mm:114): InitMicrophone [07:17:11:INFO] [000:462] [37379] (audio_device_module_ios.mm:277): StereoRecordingIsAvailable [07:17:11:INFO] [000:462] [37379] (audio_device_module_ios.mm:284): output: 0 [07:17:11:INFO] [000:462] [37379] (audio_device_module_ios.mm:289): SetStereoRecording(0) [07:17:11:INFO] [000:462] [37379] (adm_helpers.cc:77): Failed to set stereo recording mode. [07:17:11:INFO] [000:462] [37379] (apm_helpers.cc:32): Setting AGC mode to 2 [07:17:11:INFO] [000:463] [37379] (audio_processing_impl.cc:727): Highpass filter activated: 0 [07:17:11:INFO] [000:463] [37379] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:11:INFO] [000:463] [37379] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:11:INFO] [000:463] [37379] (audio_device_module_ios.mm:581): RegisterAudioCallback [07:17:11:INFO] [000:463] [37379] (audio_device_buffer.cc:81): RegisterAudioCallback [07:17:11:INFO] [000:463] [37379] (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, } [07:17:11:INFO] [000:463] [37379] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [07:17:11:INFO] [000:463] [37379] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [07:17:11:INFO] [000:463] [37379] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [07:17:11:INFO] [000:463] [37379] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [07:17:11:INFO] [000:463] [37379] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [07:17:11:INFO] [000:463] [37379] (audio_device_module_ios.mm:600): output: 0 [07:17:11:INFO] [000:463] [37379] (audio_processing_impl.cc:727): Highpass filter activated: 0 [07:17:11:INFO] [000:464] [37379] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:11:INFO] [000:464] [37379] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:11:INFO] [000:464] [37379] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [07:17:11:INFO] [000:464] [37379] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [07:17:11:INFO] [000:464] [37379] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [07:17:11:INFO] [000:464] [37379] (audio_device_module_ios.mm:616): output: 0 [07:17:11:INFO] [000:464] [37379] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [07:17:11:INFO] [000:464] [37379] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [07:17:11:INFO] [000:464] [37379] (audio_device_module_ios.mm:632): output: 0 [07:17:11:INFO] [000:464] [37379] (audio_processing_impl.cc:727): Highpass filter activated: 0 [07:17:11:INFO] [000:464] [37379] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:11:INFO] [000:464] [37379] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:11:INFO] [000:464] [37379] (apm_helpers.cc:62): NS set to 0 [07:17:11:INFO] [000:464] [37379] (webrtc_voice_engine.cc:447): Stereo swapping enabled? 0 [07:17:11:INFO] [000:464] [37379] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [07:17:11:INFO] [000:464] [37379] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [07:17:11:INFO] [000:464] [37379] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [07:17:11:INFO] [000:464] [37379] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [07:17:11:INFO] [000:464] [37379] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [07:17:11:INFO] [000:464] [37379] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [07:17:11:INFO] [000:464] [37379] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [07:17:11:INFO] [000:464] [37379] (webrtc_voice_engine.cc:511): Setting AGC to 0 [07:17:11:INFO] [000:464] [37379] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [07:17:11:INFO] [000:464] [37379] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:11:INFO] [000:464] [37379] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:11:INFO] [000:464] [37379] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:11:INFO] (RTCLogging.mm:33): (ARDAppClient.m:270 -[ARDAppClient connectToRoomWithId:settings:isLoopback:]_block_invoke): Joined room:testRoom on room server. [07:17:11:INFO] [000:465] [1027] (openssl_identity.cc:44): Making key pair [07:17:11:INFO] [000:824] [1027] (openssl_identity.cc:92): Returning key pair [07:17:11:INFO] [000:824] [1027] (openssl_certificate.cc:58): Making certificate for WebRTC [07:17:11:INFO] [000:828] [1027] (openssl_certificate.cc:108): Returning certificate [07:17:11:INFO] [000:828] [1027] (RTCCertificate.mm:63): CERT PEM [07:17:11:INFO] [000:828] [1027] (RTCCertificate.mm:64): -----BEGIN CERTIFICATE----- [07:17:11:INFO] MIIBnjCCAQegAwIBAgIJAPzFTcKXR9LtMA0GCSqGSIb3DQEBCwUAMBExDzANBgNV [07:17:11:INFO] BAMMBldlYlJUQzAeFw0xOTA0MTYxNDE3MTFaFw0xOTA0MTcxNDE4NTFaMBExDzAN [07:17:11:INFO] BgNVBAMMBldlYlJUQzCBnzANBgkqhkiG9w0BAQEFAAOBjQAwgYkCgYEAoovsqlRy [07:17:11:INFO] n/pkfG3lLpOvdbUmWUtcQqWJqUlBFwY5I894AMHrBpPkcSInnYsDvZMsBxdXrWBa [07:17:12:INFO] rgz6XhR+kwfehArSgQ7cveQ9jUOQgFA7g9YNbJcJmtBg42YuacYZCbM7pvxOSd35 [07:17:12:INFO] BDu6HVfFJ+rxT9i1ufE/7NQQL7Gs23Es3XMCAwEAATANBgkqhkiG9w0BAQsFAAOB [07:17:12:INFO] gQAfajXFQ6HGbpz/mEOYC84d8jZnBTZlHCNu1Ij/pGP3oZtySeIHiJsQq4xQboQ+ [07:17:12:INFO] T+MSzdJNH825cKoydubOjcv+sZ62m+rSz/khepAjZ4j1qZP7Fb5+yW+xHkCKB6fU [07:17:12:INFO] Ovv3tdlvodwPpwLTE4Z7Q2+sEy92gZ63Thf1KsLDuEerSw== [07:17:12:INFO] -----END CERTIFICATE----- [07:17:12:INFO] [07:17:12:INFO] [000:828] [1027] (RTCConfiguration.mm:213): Have configured cert - using it. [07:17:12:INFO] [000:830] [1027] (RTCConfiguration.mm:218): Created cert from PEM strings. [07:17:12:INFO] [000:830] [37379] (rtc_event_log_impl.cc:71): Creating legacy encoder for RTC event log. [07:17:12:INFO] [000:831] [37379] (peer_connection_factory.cc:407): Using default network controller factory [07:17:12:INFO] [000:831] [37379] (bitrate_prober.cc:66): Bandwidth probing enabled, set to inactive [07:17:12:INFO] [000:831] [37379] (paced_sender.cc:418): ProcessThreadAttached 0x29d1ea50 [07:17:12:INFO] [000:831] [37379] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85 [07:17:12:INFO] [000:831] [37379] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating. [07:17:12:INFO] [000:833] [37123] (webrtc_session_description_factory.cc:314): Using certificate supplied to the constructor. [07:17:12:INFO] [000:833] [37123] (peer_connection.cc:1405): Adding audio transceiver in response to a call to AddTrack. [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:396 -[ARDAppClient peerConnectionShouldNegotiate:]): WARNING: Renegotiation needed but unimplemented. [07:17:12:INFO] [000:834] [1027] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x600000202a30): created DTMF sender: RTCDtmfSender { [07:17:12:INFO] remainingTones: [07:17:12:INFO] duration: 0.100000 sec [07:17:12:INFO] interToneGap: 0.050000 sec [07:17:12:INFO] } [07:17:12:INFO] [000:834] [1027] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x6000002281a0): created sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSa0 [07:17:12:INFO] } [07:17:12:INFO] [000:834] [1027] (RTCLogging.mm:33): (RTCFileVideoCapturer.m:103 -[RTCFileVideoCapturer stopCapture]): File capturer stopped. [07:17:12:INFO] [000:835] [37123] (peer_connection.cc:1405): Adding video transceiver in response to a call to AddTrack. [07:17:12:INFO] [000:835] [37123] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [07:17:12:INFO] [000:835] [1027] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600000228680): created sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSv0 [07:17:12:INFO] } [07:17:12:INFO] [000:835] [1027] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x608000202aa0): created DTMF sender: RTCDtmfSender { [07:17:12:INFO] remainingTones: [07:17:12:INFO] duration: 0.100000 sec [07:17:12:INFO] interToneGap: 0.050000 sec [07:17:12:INFO] } [07:17:12:INFO] [000:835] [1027] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x6000002286a0): created sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSa0 [07:17:12:INFO] } [07:17:12:INFO] [000:838] [1027] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x60000005ebd0): created receiver: RTCRtpReceiver { [07:17:12:INFO] receiverId: e5d742a8-8500-4691-9c01-27eb66284eb0 [07:17:12:INFO] } [07:17:12:INFO] [000:839] [1027] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x60000005eb70): created transceiver: RTCRtpTransceiver { [07:17:12:INFO] sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSa0 [07:17:12:INFO] } [07:17:12:INFO] receiver: RTCRtpReceiver { [07:17:12:INFO] receiverId: e5d742a8-8500-4691-9c01-27eb66284eb0 [07:17:12:INFO] } [07:17:12:INFO] } [07:17:12:INFO] [000:839] [37123] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [07:17:12:INFO] [000:839] [1027] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x6080000398e0): created sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSv0 [07:17:12:INFO] } [07:17:12:INFO] [000:839] [1027] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x60000005ec30): created receiver: RTCRtpReceiver { [07:17:12:INFO] receiverId: b8e8fa8f-7ec3-4479-9f11-cdf30ad385eb [07:17:12:INFO] } [07:17:12:INFO] [000:840] [1027] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x60800005b390): created transceiver: RTCRtpTransceiver { [07:17:12:INFO] sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSv0 [07:17:12:INFO] } [07:17:12:INFO] receiver: RTCRtpReceiver { [07:17:12:INFO] receiverId: b8e8fa8f-7ec3-4479-9f11-cdf30ad385eb [07:17:12:INFO] } [07:17:12:INFO] } [07:17:12:INFO] [000:841] [37379] (rtc_event_log_impl.cc:186): Starting WebRTC event log. (Timestamp, UTC) = (12617628346, 1555510631967939). [07:17:12:INFO] [000:843] [1027] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [07:17:12:INFO] [000:845] [1027] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [07:17:12:INFO] [000:852] [25347] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [07:17:12:INFO] [000:852] [25347] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [07:17:12:INFO] [000:852] [25347] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [07:17:12:INFO] [000:852] [25347] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [07:17:12:INFO] [000:852] [25347] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [07:17:12:INFO] [000:852] [25347] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [07:17:12:INFO] [000:852] [25347] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [07:17:12:INFO] [000:853] [25347] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0 [07:17:12:INFO] [000:853] [25347] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 29d20630 [07:17:12:INFO] [000:853] [25347] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [07:17:12:INFO] [000:853] [25347] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [07:17:12:INFO] [000:853] [25347] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [07:17:12:INFO] [000:853] [25347] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [07:17:12:INFO] [000:853] [25347] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [07:17:12:INFO] [000:853] [25347] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [07:17:12:INFO] [000:853] [25347] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [07:17:12:INFO] [000:854] [25347] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 1 transport 0 [07:17:12:INFO] [000:854] [25347] (dtls_srtp_transport.cc:66): Setting RTP Transport on 1 transport 29c141c0 [07:17:12:INFO] [000:855] [25347] (p2p_transport_channel.cc:432): Set ICE ufrag: trVv pwd: apTVJECXl8ccsmrvX3dGAQ3d on transport 0 [07:17:12:INFO] [000:855] [25347] (p2p_transport_channel.cc:432): Set ICE ufrag: trVv pwd: apTVJECXl8ccsmrvX3dGAQ3d on transport 1 [07:17:12:INFO] [000:855] [25603] (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, } [07:17:12:INFO] [000:856] [25603] (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, } [07:17:12:INFO] [000:856] [25603] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [07:17:12:INFO] [000:856] [25603] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [07:17:12:INFO] [000:856] [25603] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [07:17:12:INFO] [000:857] [25603] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [07:17:12:INFO] [000:857] [25603] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [07:17:12:INFO] [000:857] [25603] (audio_device_module_ios.mm:600): output: 0 [07:17:12:INFO] [000:857] [25603] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [000:858] [25603] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [000:858] [25603] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [000:858] [25603] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [07:17:12:INFO] [000:858] [25603] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [07:17:12:INFO] [000:858] [25603] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [07:17:12:INFO] [000:858] [25603] (audio_device_module_ios.mm:616): output: 0 [07:17:12:INFO] [000:858] [25603] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [07:17:12:INFO] [000:858] [25603] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [07:17:12:INFO] [000:858] [25603] (audio_device_module_ios.mm:632): output: 0 [07:17:12:INFO] [000:858] [25603] (apm_helpers.cc:62): NS set to 0 [07:17:12:INFO] [000:858] [25603] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [07:17:12:INFO] [000:858] [25603] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [07:17:12:INFO] [000:858] [25603] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [07:17:12:INFO] [000:858] [25603] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [07:17:12:INFO] [000:858] [25603] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [07:17:12:INFO] [000:858] [25603] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [07:17:12:INFO] [000:858] [25603] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [07:17:12:INFO] [000:858] [25603] (webrtc_voice_engine.cc:511): Setting AGC to 0 [07:17:12:INFO] [000:858] [25603] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [07:17:12:INFO] [000:858] [25603] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [000:858] [25603] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [000:858] [25603] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [000:858] [25603] (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, } [07:17:12:INFO] [000:859] [25603] (channel.cc:144): Created channel for 0 [07:17:12:INFO] [000:860] [25603] (channel.cc:212): BaseChannel::Init_w, media_transport=0 [07:17:12:INFO] [000:861] [25603] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [000:861] [25603] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [000:861] [25603] (webrtc_video_engine.cc:483): CreateMediaChannel. Options: VideoOptions {} [07:17:12:INFO] [000:862] [25603] (channel.cc:144): Created channel for 1 [07:17:12:INFO] [000:863] [25603] (channel.cc:212): BaseChannel::Init_w, media_transport=0 [07:17:12:INFO] [000:863] [25603] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [000:863] [25603] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [000:863] [38403] (peer_connection.cc:4173): Session: 6407474718852373401 Old state: kStable New state: kHaveLocalOffer [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 1 [07:17:12:INFO] [000:864] [25603] (channel.cc:871): Setting local voice description [07:17:12:INFO] [000:864] [25603] (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}]} [07:17:12:INFO] [000:865] [25603] (webrtc_voice_engine.cc:1513): Setting receive voice codecs. [07:17:12:INFO] [000:866] [25603] (webrtc_voice_engine.cc:1784): AddSendStream: {id:ARDAMSa0;ssrcs:[2766236256];ssrc_groups:;cname:1JnqTdn92iBG4iL0;stream_ids:ARDAMS;} [07:17:12:INFO] [000:870] [25603] (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 [07:17:12:INFO] [000:872] [25603] (audio_coding_module.cc:333): Created [07:17:12:INFO] [000:875] [25603] (channel_send.cc:708): Not setting media_transport_ rate observers. [07:17:12:INFO] [000:875] [25603] (audio_send_stream.cc:142): AudioSendStream: 2766236256 [07:17:12:INFO] [000:875] [25603] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2766236256, extmap-allow-mixed: false, extensions: [], c_name: 1JnqTdn92iBG4iL0}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: <unset>} [07:17:12:INFO] [000:877] [25603] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [000:877] [25603] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [000:877] [25603] (channel.cc:676): Add send stream ssrc: 2766236256 [07:17:12:INFO] [000:877] [25603] (channel.cc:863): Changing voice state, recv=0 send=0 [07:17:12:INFO] [000:877] [25603] (channel.cc:1009): Setting local video description [07:17:12:INFO] [000:877] [25603] (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}]} [07:17:12:INFO] [000:879] [25603] (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]} [07:17:12:INFO] [000:880] [25603] (webrtc_video_engine.cc:1072): AddSendStream: {id:ARDAMSv0;ssrcs:[934729385,2198679567];ssrc_groups:{semantics:FID;ssrcs:[934729385,2198679567]};cname:1JnqTdn92iBG4iL0;stream_ids:ARDAMS;} [07:17:12:INFO] [000:880] [25603] (webrtc_video_engine.cc:1111): SetLocalSsrc on all the receive streams because we added a send stream. [07:17:12:INFO] [000:880] [25603] (channel.cc:676): Add send stream ssrc: 934729385 [07:17:12:INFO] [000:880] [25603] (channel.cc:996): Changing video state, send=0 [07:17:12:INFO] [000:880] [38403] (peer_connection.cc:5512): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport. [07:17:12:INFO] [000:881] [25603] (webrtc_voice_engine.cc:1485): Setting voice channel options: AudioOptions {} [07:17:12:INFO] [000:881] [25603] (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, } [07:17:12:INFO] [000:881] [25603] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [07:17:12:INFO] [000:882] [25603] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [07:17:12:INFO] [000:882] [25603] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [07:17:12:INFO] [000:882] [25603] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [07:17:12:INFO] [000:882] [25603] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [07:17:12:INFO] [000:882] [25603] (audio_device_module_ios.mm:600): output: 0 [07:17:12:INFO] [000:882] [25603] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [000:882] [25603] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [000:882] [25603] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [000:882] [25603] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [07:17:12:INFO] [000:882] [25603] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [07:17:12:INFO] [000:882] [25603] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [07:17:12:INFO] [000:883] [25603] (audio_device_module_ios.mm:616): output: 0 [07:17:12:INFO] [000:883] [25603] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [07:17:12:INFO] [000:883] [25603] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [07:17:12:INFO] [000:883] [25603] (audio_device_module_ios.mm:632): output: 0 [07:17:12:INFO] [000:883] [25603] (apm_helpers.cc:62): NS set to 0 [07:17:12:INFO] [000:883] [25603] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [07:17:12:INFO] [000:883] [25603] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [07:17:12:INFO] [000:883] [25603] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [07:17:12:INFO] [000:883] [25603] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [07:17:12:INFO] [000:883] [25603] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [07:17:12:INFO] [000:883] [25603] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [07:17:12:INFO] [000:883] [25603] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [07:17:12:INFO] [000:883] [25603] (webrtc_voice_engine.cc:511): Setting AGC to 0 [07:17:12:INFO] [000:883] [25603] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [07:17:12:INFO] [000:883] [25603] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [000:883] [25603] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [000:883] [25603] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [000:883] [25603] (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, } [07:17:12:INFO] [000:884] [25603] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 934729385, options: VideoOptions {noise reduction: false, is_screencast : false, }, source = (source)) [07:17:12:INFO] [000:884] [25347] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [07:17:12:INFO] [000:884] [25347] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [07:17:12:INFO] [000:885] [1027] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 1 [07:17:12:INFO] [000:886] [25347] (basic_port_allocator.cc:108): Filtered out ignored networks: [07:17:12:INFO] [000:886] [1027] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [07:17:12:INFO] [000:887] [25347] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [07:17:12:INFO] [000:887] [25347] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [07:17:12:INFO] [000:888] [25347] (basic_port_allocator.cc:851): Network manager has started [07:17:12:INFO] [000:888] [25347] (basic_port_allocator.cc:108): Filtered out ignored networks: [07:17:12:INFO] [000:888] [25347] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [07:17:12:INFO] [000:888] [25347] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [07:17:12:INFO] [000:888] [25347] (basic_port_allocator.cc:851): Network manager has started [07:17:12:INFO] [000:888] [25347] (basic_port_allocator.cc:108): Filtered out ignored networks: [07:17:12:INFO] [000:888] [25347] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [07:17:12:INFO] [000:888] [25347] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [07:17:12:INFO] [000:888] [25347] (basic_port_allocator.cc:766): Allocate ports on 1 networks [07:17:12:INFO] [000:889] [25347] (basic_port_allocator.cc:108): Filtered out ignored networks: [07:17:12:INFO] [000:889] [25347] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [07:17:12:INFO] [000:889] [25347] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [07:17:12:INFO] [000:889] [25347] (basic_port_allocator.cc:766): Allocate ports on 1 networks [07:17:12:INFO] [000:890] [37635] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [07:17:12:INFO] [000:890] [37635] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [07:17:12:INFO] [000:890] [37635] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [07:17:12:INFO] [000:890] [37635] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [07:17:12:INFO] [000:890] [37635] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [07:17:12:INFO] [000:890] [25347] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Udp [07:17:12:INFO] [000:890] [37635] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [07:17:12:INFO] [000:890] [37635] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [07:17:12:INFO] [000:890] [37635] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0 [07:17:12:INFO] [000:890] [37635] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 29e2d740 [07:17:12:INFO] [000:890] [37635] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [07:17:12:INFO] [000:891] [25347] (port.cc:320): Port[2a880400::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10 [07:17:12:INFO] [000:892] [37635] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [07:17:12:INFO] [000:892] [37635] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [07:17:12:INFO] [000:892] [37635] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [07:17:12:INFO] [000:892] [37635] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [07:17:12:INFO] [000:892] [37635] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [07:17:12:INFO] [000:892] [37635] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [07:17:12:INFO] [000:892] [37635] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 1 transport 0 [07:17:12:INFO] [000:892] [37635] (dtls_srtp_transport.cc:66): Setting RTP Transport on 1 transport 29f58e10 [07:17:12:INFO] [000:892] [37635] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=trVv, renomination enabled [07:17:12:INFO] [000:893] [37635] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=trVv, renomination enabled [07:17:12:INFO] [000:893] [37379] (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, } [07:17:12:INFO] [000:893] [25347] (basic_port_allocator.cc:875): Adding allocated port for 0 [07:17:12:INFO] [000:894] [37379] (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, } [07:17:12:INFO] [000:894] [37379] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [07:17:12:INFO] [000:894] [37379] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [07:17:12:INFO] [000:894] [37379] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [07:17:12:INFO] [000:894] [37379] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [07:17:12:INFO] [000:894] [37379] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [07:17:12:INFO] [000:894] [25347] (basic_port_allocator.cc:894): Port[2a880400:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator [07:17:12:INFO] [000:894] [37379] (audio_device_module_ios.mm:600): output: 0 [07:17:12:INFO] [000:894] [37379] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [000:895] [25347] (basic_port_allocator.cc:912): Port[2a880400:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:1435151450:1:udp:2122260223:192.168.147.82:64902:local::0:trVv:apTVJECXl8ccsmrvX3dGAQ3d:1:10:0] [07:17:12:INFO] [000:895] [37379] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [000:895] [37379] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [000:895] [25347] (basic_port_allocator.cc:940): Port[2a880400:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready. [07:17:12:INFO] [000:895] [37379] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [07:17:12:INFO] [000:895] [37379] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [07:17:12:INFO] [000:895] [37379] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [07:17:12:INFO] [000:895] [25347] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [07:17:12:INFO] [000:895] [37379] (audio_device_module_ios.mm:616): output: 0 [07:17:12:INFO] [000:895] [25347] (p2p_transport_channel.cc:799): Port[2a880400:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0 [07:17:12:INFO] [000:895] [37379] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [07:17:12:INFO] [000:895] [37379] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [07:17:12:INFO] [000:896] [37379] (audio_device_module_ios.mm:632): output: 0 [07:17:12:INFO] [000:896] [37379] (apm_helpers.cc:62): NS set to 0 [07:17:12:INFO] [000:896] [37379] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [07:17:12:INFO] [000:896] [37379] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [07:17:12:INFO] [000:896] [37379] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [07:17:12:INFO] [000:896] [37379] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [07:17:12:INFO] [000:896] [37379] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [07:17:12:INFO] [000:896] [25347] (basic_port_allocator.cc:1017): Port[2a880400:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates. [07:17:12:INFO] [000:896] [37379] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [07:17:12:INFO] [000:896] [25347] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Udp [07:17:12:INFO] [000:896] [37379] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [07:17:12:INFO] [000:897] [25347] (port.cc:320): Port[2b80dc00::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10 [07:17:12:INFO] [000:897] [37379] (webrtc_voice_engine.cc:511): Setting AGC to 0 [07:17:12:INFO] [000:897] [37379] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [07:17:12:INFO] [000:897] [25347] (basic_port_allocator.cc:875): Adding allocated port for 1 [07:17:12:INFO] [000:897] [37379] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [000:897] [25347] (basic_port_allocator.cc:894): Port[2b80dc00:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator [07:17:12:INFO] [000:897] [25347] (basic_port_allocator.cc:912): Port[2b80dc00:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:1435151450:1:udp:2122260223:192.168.147.82:50915:local::0:trVv:apTVJECXl8ccsmrvX3dGAQ3d:1:10:0] [07:17:12:INFO] [000:897] [25347] (basic_port_allocator.cc:940): Port[2b80dc00:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready. [07:17:12:INFO] [000:897] [25347] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [07:17:12:INFO] [000:897] [25347] (p2p_transport_channel.cc:799): Port[2b80dc00:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0 [07:17:12:INFO] [000:897] [25347] (basic_port_allocator.cc:1017): Port[2b80dc00:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates. [07:17:12:INFO] [000:897] [37379] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [000:897] [37379] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [000:897] [37379] (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, } [07:17:12:INFO] [000:897] [37379] (channel.cc:144): Created channel for 0 [07:17:12:INFO] [000:898] [37379] (channel.cc:212): BaseChannel::Init_w, media_transport=0 [07:17:12:INFO] [000:898] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [000:898] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [000:898] [37379] (webrtc_video_engine.cc:483): CreateMediaChannel. Options: VideoOptions {} [07:17:12:INFO] [000:898] [37379] (channel.cc:144): Created channel for 1 [07:17:12:INFO] [000:899] [37379] (channel.cc:212): BaseChannel::Init_w, media_transport=0 [07:17:12:INFO] [000:899] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [000:899] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [000:899] [37123] (peer_connection.cc:4173): Session: 7386157372738723833 Old state: kStable New state: kHaveRemoteOffer [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 3 [07:17:12:INFO] [000:899] [37379] (channel.cc:923): Setting remote voice description [07:17:12:INFO] [000:899] [37379] (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 {}} [07:17:12:INFO] [000:899] [37379] (webrtc_voice_engine.cc:1702): Recreate all the receive streams because the send codec has changed. [07:17:12:INFO] [000:899] [37379] (webrtc_voice_engine.cc:2163): WebRtcVoiceMediaChannel::SetMaxSendBitrate. [07:17:12:INFO] [000:899] [37379] (webrtc_voice_engine.cc:1485): Setting voice channel options: AudioOptions {} [07:17:12:INFO] [000:899] [37379] (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, } [07:17:12:INFO] [000:900] [37379] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [07:17:12:INFO] [000:900] [37379] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [07:17:12:INFO] [000:900] [37379] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [07:17:12:INFO] [000:900] [37379] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [07:17:12:INFO] [000:900] [37379] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [07:17:12:INFO] [000:900] [37379] (audio_device_module_ios.mm:600): output: 0 [07:17:12:INFO] [000:900] [37379] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [000:900] [37379] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [000:900] [37379] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [000:900] [37379] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [07:17:12:INFO] [000:900] [37379] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [07:17:12:INFO] [000:900] [37379] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [07:17:12:INFO] [000:900] [37379] (audio_device_module_ios.mm:616): output: 0 [07:17:12:INFO] [000:900] [37379] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [07:17:12:INFO] [000:900] [37379] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [07:17:12:INFO] [000:901] [37379] (audio_device_module_ios.mm:632): output: 0 [07:17:12:INFO] [000:901] [37379] (apm_helpers.cc:62): NS set to 0 [07:17:12:INFO] [000:901] [37379] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [07:17:12:INFO] [000:901] [37379] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [07:17:12:INFO] [000:901] [37379] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [07:17:12:INFO] [000:901] [37379] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [07:17:12:INFO] [000:901] [37379] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [07:17:12:INFO] [000:901] [37379] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [07:17:12:INFO] [000:901] [37379] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [07:17:12:INFO] [000:901] [37379] (webrtc_voice_engine.cc:511): Setting AGC to 0 [07:17:12:INFO] [000:901] [37379] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [07:17:12:INFO] [000:901] [37379] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [000:901] [37379] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [000:901] [37379] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [000:901] [37379] (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, } [07:17:12:INFO] [000:902] [37379] (webrtc_voice_engine.cc:1849): AddRecvStream: {id:ARDAMSa0;ssrcs:[2766236256];ssrc_groups:;cname:1JnqTdn92iBG4iL0;stream_ids:ARDAMS;} [07:17:12:INFO] [000:903] [37379] (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 [07:17:12:INFO] [000:903] [37379] (audio_coding_module.cc:333): Created [07:17:12:INFO] [000:903] [37379] (audio_receive_stream.cc:117): AudioReceiveStream: 2766236256 [07:17:12:INFO] [000:903] [37379] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2766236256, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS} [07:17:12:INFO] [000:903] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [000:903] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [000:904] [37379] (channel.cc:719): Add remote ssrc: 2766236256 [07:17:12:INFO] [000:904] [37379] (channel.cc:863): Changing voice state, recv=0 send=0 [07:17:12:INFO] [000:904] [37379] (channel.cc:1061): Setting remote video description [07:17:12:INFO] [000:904] [37379] (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} [07:17:12:INFO] [000:905] [37379] (webrtc_video_engine.cc:690): Using codec: VideoCodec[96:H264] [07:17:12:INFO] [000:905] [37379] (webrtc_video_engine.cc:758): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. [07:17:12:INFO] [000:905] [37379] (webrtc_video_engine.cc:1174): AddRecvStream: {id:ARDAMSv0;ssrcs:[934729385,2198679567];ssrc_groups:{semantics:FID;ssrcs:[934729385,2198679567]};cname:1JnqTdn92iBG4iL0;stream_ids:ARDAMS;} [07:17:12:INFO] [000:912] [37379] (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: 934729385, 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: 2198679567, 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} [07:17:12:INFO] [000:912] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [000:912] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [000:916] [37379] (channel.cc:719): Add remote ssrc: 934729385 [07:17:12:INFO] [000:916] [37379] (channel.cc:996): Changing video state, send=0 [07:17:12:INFO] [000:916] [37123] (peer_connection.cc:5512): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport. [07:17:12:INFO] [000:916] [37123] (peer_connection.cc:2745): Processing the MSIDs for MID=0 (streams=[ARDAMS]). [07:17:12:INFO] [000:917] [37123] (peer_connection.cc:2756): Processing the addition of a remote track for MID=0. [07:17:12:INFO] [000:917] [37379] (webrtc_voice_engine.cc:1962): SetOutputVolume() to 1 for recv stream with ssrc 2766236256 [07:17:12:INFO] [000:917] [37123] (peer_connection.cc:2745): Processing the MSIDs for MID=1 (streams=[ARDAMS]). [07:17:12:INFO] [000:917] [37123] (peer_connection.cc:2756): Processing the addition of a remote track for MID=1. [07:17:12:INFO] [000:917] [37379] (webrtc_video_engine.cc:1304): SetSink: ssrc:934729385 (ptr) [07:17:12:INFO] [000:917] [37123] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x608000202d80): created DTMF sender: RTCDtmfSender { [07:17:12:INFO] remainingTones: [07:17:12:INFO] duration: 0.100000 sec [07:17:12:INFO] interToneGap: 0.050000 sec [07:17:12:INFO] } [07:17:12:INFO] [000:917] [37123] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60800023fdc0): created sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSa0 [07:17:12:INFO] } [07:17:12:INFO] [000:918] [37123] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x60800005f1d0): created receiver: RTCRtpReceiver { [07:17:12:INFO] receiverId: e5d742a8-8500-4691-9c01-27eb66284eb0 [07:17:12:INFO] } [07:17:12:INFO] [000:918] [37123] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x60800005ee10): created transceiver: RTCRtpTransceiver { [07:17:12:INFO] sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSa0 [07:17:12:INFO] } [07:17:12:INFO] receiver: RTCRtpReceiver { [07:17:12:INFO] receiverId: e5d742a8-8500-4691-9c01-27eb66284eb0 [07:17:12:INFO] } [07:17:12:INFO] } [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving audio on track e5d742a8-8500-4691-9c01-27eb66284eb0. [07:17:12:INFO] [000:918] [37123] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [07:17:12:INFO] [000:918] [37123] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60800023ff20): created sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSv0 [07:17:12:INFO] } [07:17:12:INFO] [000:919] [37123] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x60800005f470): created receiver: RTCRtpReceiver { [07:17:12:INFO] receiverId: b8e8fa8f-7ec3-4479-9f11-cdf30ad385eb [07:17:12:INFO] } [07:17:12:INFO] [000:919] [37123] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x60800005ee10): created transceiver: RTCRtpTransceiver { [07:17:12:INFO] sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSv0 [07:17:12:INFO] } [07:17:12:INFO] receiver: RTCRtpReceiver { [07:17:12:INFO] receiverId: b8e8fa8f-7ec3-4479-9f11-cdf30ad385eb [07:17:12:INFO] } [07:17:12:INFO] } [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving video on track b8e8fa8f-7ec3-4479-9f11-cdf30ad385eb. [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:381 -[ARDAppClient peerConnection:didAddStream:]): Stream with 1 video tracks and 1 audio tracks was added. [07:17:12:INFO] [000:919] [1027] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x608000202dd0): created DTMF sender: RTCDtmfSender { [07:17:12:INFO] remainingTones: [07:17:12:INFO] duration: 0.100000 sec [07:17:12:INFO] interToneGap: 0.050000 sec [07:17:12:INFO] } [07:17:12:INFO] [000:919] [1027] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x600000229480): created sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSa0 [07:17:12:INFO] } [07:17:12:INFO] [000:919] [38403] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [07:17:12:INFO] [000:920] [1027] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60800023fb00): created sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSv0 [07:17:12:INFO] } [07:17:12:INFO] [000:922] [37123] (peer_connection.cc:6210): 0 is not ready to use the remote candidate because the local or remote description is not set. [07:17:12:INFO] [000:922] [37123] (peer_connection.cc:6210): 1 is not ready to use the remote candidate because the local or remote description is not set. [07:17:12:INFO] [000:922] [37123] (peer_connection.cc:2110): CreateAnswer: offer_to_receive_audio is not supported with Unified Plan semantics. Use the RtpTransceiver API instead. [07:17:12:INFO] [000:922] [37123] (peer_connection.cc:2115): CreateAnswer: offer_to_receive_video is not supported with Unified Plan semantics. Use the RtpTransceiver API instead. [07:17:12:INFO] [000:923] [37123] (peer_connection.cc:5540): Local and Remote descriptions must be applied to get the SSL Role of the session. [07:17:12:INFO] [000:923] [37123] (peer_connection.cc:5540): Local and Remote descriptions must be applied to get the SSL Role of the session. [07:17:12:INFO] [000:926] [1027] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [07:17:12:INFO] [000:927] [1027] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [07:17:12:INFO] [000:930] [37635] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0 [07:17:12:INFO] [000:930] [37635] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 29e2d740 [07:17:12:INFO] [000:930] [37635] (p2p_transport_channel.cc:432): Set ICE ufrag: mZbg pwd: QUgTM02OeOJc9Wr8tvv4JS5d on transport 0 [07:17:12:INFO] [000:931] [37635] (dtls_transport.cc:370): DtlsTransport[0|1|__]: DTLS setup complete. [07:17:12:INFO] [000:931] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [000:931] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [000:931] [37123] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv. [07:17:12:INFO] [000:931] [37123] (rtp_transceiver.cc:169): Changing transceiver (MID=1) current direction from <not set> to kSendRecv. [07:17:12:INFO] [000:931] [37379] (channel.cc:553): Channel enabled [07:17:12:INFO] [000:931] [37379] (channel.cc:863): Changing voice state, recv=0 send=0 [07:17:12:INFO] [000:932] [37379] (channel.cc:553): Channel enabled [07:17:12:INFO] [000:932] [37379] (channel.cc:996): Changing video state, send=0 [07:17:12:INFO] [000:932] [37123] (peer_connection.cc:4173): Session: 7386157372738723833 Old state: kHaveRemoteOffer New state: kStable [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 0 [07:17:12:INFO] [000:932] [37379] (channel.cc:871): Setting local voice description [07:17:12:INFO] [000:932] [37379] (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}]} [07:17:12:INFO] [000:932] [37379] (webrtc_voice_engine.cc:1513): Setting receive voice codecs. [07:17:12:INFO] [000:932] [37379] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2766236256, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS} [07:17:12:INFO] [000:933] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [000:933] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [000:933] [37379] (audio_receive_stream.cc:140): ~AudioReceiveStream: 2766236256 [07:17:12:INFO] [000:933] [37379] (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 [07:17:12:INFO] [000:933] [37379] (audio_coding_module.cc:333): Created [07:17:12:INFO] [000:933] [37379] (audio_receive_stream.cc:117): AudioReceiveStream: 2766236256 [07:17:12:INFO] [000:933] [37379] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2766236256, 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} [07:17:12:INFO] [000:933] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [000:933] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [000:934] [37379] (webrtc_voice_engine.cc:1784): AddSendStream: {id:ARDAMSa0;ssrcs:[2391384973];ssrc_groups:;cname:uLPegaWZdV596BVM;stream_ids:ARDAMS;} [07:17:12:INFO] [000:934] [37379] (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 [07:17:12:INFO] [000:934] [37379] (audio_coding_module.cc:333): Created [07:17:12:INFO] [000:934] [37379] (channel_send.cc:708): Not setting media_transport_ rate observers. [07:17:12:INFO] [000:934] [37379] (audio_send_stream.cc:142): AudioSendStream: 2391384973 [07:17:12:INFO] [000:934] [37379] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2391384973, 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: uLPegaWZdV596BVM}, 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}}}} [07:17:12:INFO] [000:938] [37379] (audio_encoder_opus.cc:758): Set Opus bitrate to 32000 bps. [07:17:12:INFO] [000:938] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [000:938] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [000:938] [37379] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2766236256, local_ssrc: 2391384973, 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} [07:17:12:INFO] [000:938] [37379] (channel.cc:676): Add send stream ssrc: 2391384973 [07:17:12:INFO] [000:939] [37379] (audio_device_module_ios.mm:543): Playing [07:17:12:INFO] [000:939] [37379] (audio_device_module_ios.mm:480): InitPlayout [07:17:12:INFO] [000:939] [37379] (audio_device_module_ios.mm:506): PlayoutIsInitialized [07:17:12:INFO] [000:939] [37379] (audio_device_ios.mm:185): AudioDeviceIOS::InitPlayout [07:17:12:INFO] [000:939] [37379] (audio_device_ios.mm:859): AudioDeviceIOS::InitPlayOrRecord [07:17:12:INFO] [000:948] [25347] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Relay [07:17:12:INFO] [000:948] [25347] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Relay [07:17:12:INFO] [000:962] [37379] (RTCLogging.mm:33): (RTCAudioSession.mm:99 -[RTCAudioSession initWithAudioSession:]): RTCAudioSession (0x608000110e90): init. [07:17:12:INFO] [000:962] [37379] (RTCLogging.mm:33): (audio_device_ios.mm:824 ConfigureAudioSession): Configuring audio session. [07:17:12:INFO] [000:962] [37379] (RTCLogging.mm:33): (RTCAudioSession.mm:699 -[RTCAudioSession configureWebRTCSession:]): Configuring audio session for WebRTC. [07:17:12:INFO] [000:963] [37379] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:64 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set category to: AVAudioSessionCategoryPlayAndRecord [07:17:12:INFO] [000:963] [37379] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:75 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set mode to: AVAudioSessionModeVoiceChat [07:17:12:INFO] [000:963] [37379] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:103 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred sample rate to: 48000.00 [07:17:12:INFO] [000:963] [37379] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:116 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred IO buffer duration to: 0.020000 [07:17:12:INFO] [000:963] [37379] (RTCLogging.mm:33): (RTCAudioSession.mm:624 -[RTCAudioSession incrementActivationCount]): Incrementing activation count. [07:17:12:INFO] [000:963] [37379] (RTCLogging.mm:33): (RTCAudioSession.mm:377 -[RTCAudioSession setActive:error:]): Number of current activations: 1 [07:17:12:INFO] [000:963] [37379] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:145 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set input number of channels to: 1 [07:17:12:INFO] [000:963] [37379] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:158 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set output number of channels to: 1 [07:17:12:INFO] [000:963] [37379] (RTCLogging.mm:33): (audio_device_ios.mm:835 ConfigureAudioSession): Configured audio session. [07:17:12:INFO] [000:963] [37379] (audio_device_ios.mm:678): AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession [07:17:12:INFO] [000:963] [37379] (RTCLogging.mm:33): (audio_device_ios.mm:683 SetupAudioBuffersForActiveAudioSession): RTCAudioSession: { [07:17:12:INFO] category: AVAudioSessionCategoryPlayAndRecord [07:17:12:INFO] categoryOptions: 4 [07:17:12:INFO] mode: AVAudioSessionModeVoiceChat [07:17:12:INFO] isActive: 1 [07:17:12:INFO] sampleRate: 48000.00 [07:17:12:INFO] IOBufferDuration: 0.010667 [07:17:12:INFO] outputNumberOfChannels: 2 [07:17:12:INFO] inputNumberOfChannels: 2 [07:17:12:INFO] outputLatency: 0.010000 [07:17:12:INFO] inputLatency: 0.010000 [07:17:12:INFO] outputVolume: 1.000000 [07:17:12:INFO] } [07:17:12:INFO] [000:963] [37379] (audio_device_ios.mm:714): frames per I/O buffer: 512 [07:17:12:INFO] [000:963] [37379] (audio_device_ios.mm:715): bytes per I/O buffer: 1024 [07:17:12:INFO] [000:964] [37379] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer [07:17:12:INFO] [000:964] [37379] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000) [07:17:12:INFO] [000:964] [37379] (audio_device_buffer.cc:206): SetPlayoutChannels(1) [07:17:12:INFO] [000:964] [37379] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000) [07:17:12:INFO] [000:964] [37379] (audio_device_buffer.cc:200): SetRecordingChannels(1) [07:17:12:INFO] [000:964] [37379] (fine_audio_buffer.cc:32): FineAudioBuffer [07:17:12:INFO] [000:964] [37379] (fine_audio_buffer.cc:34): playout_samples_per_channel_10ms: 480 [07:17:12:INFO] [000:964] [37379] (fine_audio_buffer.cc:36): playout_channels: 1 [07:17:12:INFO] [000:964] [37379] (fine_audio_buffer.cc:39): record_samples_per_channel_10ms: 480 [07:17:12:INFO] [000:964] [37379] (fine_audio_buffer.cc:41): record_channels: 1 [07:17:12:INFO] [000:964] [37379] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:196 Initialize): Initializing audio unit with sample rate: 48000.000000 [07:17:12:INFO] [000:964] [37379] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:43 LogStreamDescription): AudioStreamBasicDescription: { [07:17:12:INFO] mSampleRate: 48000.00 [07:17:12:INFO] formatIDString: lpcm [07:17:12:INFO] mFormatFlags: 0xC [07:17:12:INFO] mBytesPerPacket: 2 [07:17:12:INFO] mFramesPerPacket: 1 [07:17:12:INFO] mBytesPerFrame: 2 [07:17:12:INFO] mChannelsPerFrame: 1 [07:17:12:INFO] mBitsPerChannel: 16 [07:17:12:INFO] mReserved: 28672 [07:17:12:INFO] } [07:17:12:INFO] [000:998] [37379] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:250 Initialize): Voice Processing I/O unit is now initialized. [07:17:12:INFO] [000:999] [25347] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Tcp [07:17:12:INFO] [001:000] [25347] (port.cc:320): Port[2a09fc00::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10 [07:17:12:INFO] [001:000] [25347] (basic_port_allocator.cc:875): Adding allocated port for 0 [07:17:12:INFO] [001:000] [25347] (basic_port_allocator.cc:894): Port[2a09fc00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator [07:17:12:INFO] [001:000] [25347] (basic_port_allocator.cc:912): Port[2a09fc00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:453595306:1:tcp:1518280447:192.168.147.82:50612:local::0:trVv:apTVJECXl8ccsmrvX3dGAQ3d:1:10:0] [07:17:12:INFO] [001:000] [25347] (basic_port_allocator.cc:940): Port[2a09fc00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready. [07:17:12:INFO] [001:000] [25347] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [07:17:12:INFO] [001:000] [25347] (p2p_transport_channel.cc:799): Port[2a09fc00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0 [07:17:12:INFO] [001:000] [25347] (basic_port_allocator.cc:1017): Port[2a09fc00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates. [07:17:12:INFO] [001:000] [25347] (basic_port_allocator.cc:1119): All candidates gathered for 0:1:0 [07:17:12:INFO] [001:000] [25347] (p2p_transport_channel.cc:854): P2PTransportChannel: 0, component 1 gathering complete [07:17:12:INFO] [001:000] [25347] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Tcp [07:17:12:INFO] [001:001] [25347] (port.cc:320): Port[2a0a0200::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10 [07:17:12:INFO] [001:001] [25347] (basic_port_allocator.cc:875): Adding allocated port for 1 [07:17:12:INFO] [001:001] [25347] (basic_port_allocator.cc:894): Port[2a0a0200:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator [07:17:12:INFO] [001:001] [25347] (basic_port_allocator.cc:912): Port[2a0a0200:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:453595306:1:tcp:1518280447:192.168.147.82:50613:local::0:trVv:apTVJECXl8ccsmrvX3dGAQ3d:1:10:0] [07:17:12:INFO] [001:001] [25347] (basic_port_allocator.cc:940): Port[2a0a0200:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready. [07:17:12:INFO] [001:001] [25347] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [07:17:12:INFO] [001:001] [25347] (p2p_transport_channel.cc:799): Port[2a0a0200:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0 [07:17:12:INFO] [001:001] [25347] (basic_port_allocator.cc:1017): Port[2a0a0200:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates. [07:17:12:INFO] [001:001] [25347] (basic_port_allocator.cc:1119): All candidates gathered for 1:1:0 [07:17:12:INFO] [001:001] [25347] (p2p_transport_channel.cc:854): P2PTransportChannel: 1, component 1 gathering complete [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 2 [07:17:12:INFO] [001:120] [37379] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:71 GetAGCState): VPIO unit AGC: 0 [07:17:12:INFO] [001:121] [37379] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:264 Initialize): Failed to get AGC state (1st attempt). Error=-66631. [07:17:12:INFO] [001:121] [37379] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:302 Initialize): WebRTC.Audio.BuiltInAGCWasEnabledByDefault: 0 [07:17:12:INFO] [001:121] [37379] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:308 Initialize): WebRTC.Audio.BuiltInAGCIsEnabled: 0 [07:17:12:INFO] [001:121] [37379] (audio_device_module_ios.mm:486): output: 0 [07:17:12:INFO] [001:121] [37379] (audio_device_module_ios.mm:518): StartPlayout [07:17:12:INFO] [001:121] [37379] (audio_device_module_ios.mm:543): Playing [07:17:12:INFO] [001:121] [37379] (audio_device_buffer.cc:98): StartPlayout [07:17:12:INFO] [001:121] [37379] (audio_device_ios.mm:227): AudioDeviceIOS::StartPlayout [07:17:12:INFO] [001:121] [37379] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:316 Start): Starting audio unit. [07:17:12:INFO] [001:131] [37379] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:320 Start): Failed to start audio unit. Error=-66637 [07:17:12:INFO] [001:131] [37379] (RTCLogging.mm:33): (audio_device_ios.mm:237 StartPlayout): StartPlayout failed to start audio unit. [07:17:12:INFO] [001:131] [37379] (audio_device_module_ios.mm:525): output: -1 [07:17:12:INFO] [001:132] [37379] (channel.cc:863): Changing voice state, recv=1 send=0 [07:17:12:INFO] [001:132] [37379] (message_queue.cc:517): Message took 200ms to dispatch. Posted from: SetLocalContent@../../pc/channel.cc:293 [07:17:12:INFO] [001:132] [37379] (channel.cc:1009): Setting local video description [07:17:12:INFO] [001:132] [37379] (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}]} [07:17:12:INFO] [001:133] [37379] (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]} [07:17:12:INFO] [001:133] [37379] (webrtc_video_engine.cc:2542): MaybeRecreateWebRtcFlexfecStream (recv) because of SetRecvParameters [07:17:12:INFO] [001:133] [37379] (webrtc_video_engine.cc:2547): RecreateWebRtcVideoStream (recv) because of SetRecvParameters [07:17:12:INFO] [001:133] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [001:133] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [001:133] [37379] (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: 934729385, 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: 2198679567, 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} [07:17:12:INFO] [001:134] [37379] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [07:17:12:INFO] [001:134] [37379] (receive_statistics_proxy.cc:487): Frames decoded 0 [07:17:12:INFO] WebRTC.Video.DroppedFrames.Receiver 0 [07:17:12:INFO] [07:17:12:INFO] [001:135] [37379] (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: 934729385, 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: 2198679567, 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} [07:17:12:INFO] [001:135] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [001:135] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [001:135] [37379] (webrtc_video_engine.cc:1072): AddSendStream: {id:ARDAMSv0;ssrcs:[3428412742,4010987472];ssrc_groups:{semantics:FID;ssrcs:[3428412742,4010987472]};cname:uLPegaWZdV596BVM;stream_ids:ARDAMS;} [07:17:12:INFO] [001:135] [37379] (webrtc_video_engine.cc:1883): RecreateWebRtcStream (send) because of SetCodec. [07:17:12:INFO] [001:136] [37379] (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 [07:17:12:INFO] [001:137] [9219] (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. [07:17:12:INFO] [001:138] [9219] (video_send_stream_impl.cc:250): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [3428412742], 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: [4010987472], payload_type: 97}, c_name: uLPegaWZdV596BVM}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [07:17:12:INFO] [001:138] [9219] (video_send_stream_impl.cc:278): ERROR: Initial encoder max bitrate = -1 which is <= 0! [07:17:12:INFO] [001:138] [9219] (video_send_stream_impl.cc:280): Using default encoder max bitrate = 10 Mbps [07:17:12:INFO] [001:138] [9219] (video_stream_encoder.cc:628): ConfigureEncoder requested. [07:17:12:INFO] [001:138] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [001:138] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [001:138] [37379] (video_send_stream.cc:160): VideoSendStream::Stop [07:17:12:INFO] [001:138] [37379] (webrtc_video_engine.cc:1111): SetLocalSsrc on all the receive streams because we added a send stream. [07:17:12:INFO] [001:138] [37379] (webrtc_video_engine.cc:2483): RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc=3428412742 [07:17:12:INFO] [001:138] [9219] (video_send_stream_impl.cc:417): VideoSendStream::Stop [07:17:12:INFO] [001:138] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [001:138] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [001:139] [37379] (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: 934729385, 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: 2198679567, 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} [07:17:12:INFO] [001:139] [37379] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [07:17:12:INFO] [001:139] [37379] (receive_statistics_proxy.cc:487): Frames decoded 0 [07:17:12:INFO] WebRTC.Video.DroppedFrames.Receiver 0 [07:17:12:INFO] [07:17:12:INFO] [001:140] [37379] (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: 934729385, local_ssrc: 3428412742, 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: 2198679567, 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} [07:17:12:INFO] [001:140] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [001:140] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [001:140] [37379] (channel.cc:676): Add send stream ssrc: 3428412742 [07:17:12:INFO] [001:140] [37379] (video_send_stream.cc:160): VideoSendStream::Stop [07:17:12:INFO] [001:140] [37379] (channel.cc:996): Changing video state, send=0 [07:17:12:INFO] [001:140] [9219] (video_send_stream_impl.cc:417): VideoSendStream::Stop [07:17:12:INFO] [001:140] [37123] (peer_connection.cc:4109): Changing IceConnectionState 0 => 1 [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 1 [07:17:12:INFO] [001:141] [37635] (jsep_transport_controller.cc:318): Not adding candidate because the JsepTransport doesn't exist. Ignore it. [07:17:12:INFO] [001:141] [37123] (peer_connection.cc:5518): Non-rejected SCTP m= section is needed to get the SSL Role of the SCTP transport. [07:17:12:INFO] [001:141] [37379] (webrtc_voice_engine.cc:1485): Setting voice channel options: AudioOptions {} [07:17:12:INFO] [001:141] [37379] (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, } [07:17:12:INFO] [001:141] [37379] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [07:17:12:INFO] [001:141] [37379] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [07:17:12:INFO] [001:141] [37379] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [07:17:12:INFO] [001:141] [37379] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [07:17:12:INFO] [001:141] [37379] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [07:17:12:INFO] [001:141] [37379] (audio_device_module_ios.mm:600): output: 0 [07:17:12:INFO] [001:141] [37379] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [001:141] [37379] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [001:141] [37379] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [001:141] [37379] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [07:17:12:INFO] [001:141] [37379] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [07:17:12:INFO] [001:141] [37379] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [07:17:12:INFO] [001:141] [37379] (audio_device_module_ios.mm:616): output: 0 [07:17:12:INFO] [001:141] [37379] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [07:17:12:INFO] [001:141] [37379] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [07:17:12:INFO] [001:141] [37379] (audio_device_module_ios.mm:632): output: 0 [07:17:12:INFO] [001:141] [37379] (apm_helpers.cc:62): NS set to 0 [07:17:12:INFO] [001:141] [37379] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [07:17:12:INFO] [001:141] [37379] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [07:17:12:INFO] [001:141] [37379] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [07:17:12:INFO] [001:141] [37379] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [07:17:12:INFO] [001:141] [37379] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [07:17:12:INFO] [001:141] [37379] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [07:17:12:INFO] [001:141] [37379] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [07:17:12:INFO] [001:141] [37379] (webrtc_voice_engine.cc:511): Setting AGC to 0 [07:17:12:INFO] [001:141] [37379] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [07:17:12:INFO] [001:142] [37379] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [001:142] [37379] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [001:142] [37379] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [001:142] [37379] (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, } [07:17:12:INFO] [001:142] [37379] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 3428412742, options: VideoOptions {noise reduction: false, is_screencast : false, }, source = (source)) [07:17:12:INFO] [001:142] [9219] (video_stream_encoder.cc:628): ConfigureEncoder requested. [07:17:12:INFO] [001:142] [37635] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [07:17:12:INFO] [001:143] [37635] (basic_port_allocator.cc:108): Filtered out ignored networks: [07:17:12:INFO] [001:143] [37635] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [07:17:12:INFO] [001:143] [37635] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [07:17:12:INFO] [001:143] [37635] (basic_port_allocator.cc:851): Network manager has started [07:17:12:INFO] [001:143] [37635] (basic_port_allocator.cc:108): Filtered out ignored networks: [07:17:12:INFO] [001:143] [37635] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [07:17:12:INFO] [001:143] [37635] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [07:17:12:INFO] [001:143] [37635] (basic_port_allocator.cc:766): Allocate ports on 1 networks [07:17:12:INFO] [001:143] [37635] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Udp [07:17:12:INFO] [001:143] [37635] (port.cc:320): Port[2a89ba00::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10 [07:17:12:INFO] [001:143] [37635] (basic_port_allocator.cc:875): Adding allocated port for 0 [07:17:12:INFO] [001:143] [37635] (basic_port_allocator.cc:894): Port[2a89ba00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator [07:17:12:INFO] [001:143] [37123] (message_queue.cc:517): Message took 213ms to dispatch. Posted from: SetLocalDescription@../../api/peer_connection_proxy.h:100 [07:17:12:INFO] [001:143] [37635] (basic_port_allocator.cc:912): Port[2a89ba00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:1435151450:1:udp:2122260223:192.168.147.82:58048:local::0:mZbg:QUgTM02OeOJc9Wr8tvv4JS5d:1:10:0] [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 1 [07:17:12:INFO] [001:143] [37635] (basic_port_allocator.cc:940): Port[2a89ba00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready. [07:17:12:INFO] [001:143] [37635] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [07:17:12:INFO] [001:143] [1027] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [07:17:12:INFO] [001:143] [37635] (p2p_transport_channel.cc:799): Port[2a89ba00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0 [07:17:12:INFO] [001:144] [37635] (port.cc:1128): Conn[2a909600:0:Net[en0:192.168.144.0/22:Wifi:id=1]:fo7M2zeU:1:0:local:udp:192.168.147.82:58048->4Fga3Nc5:1:2122260223:local:udp:192.168.147.82:64902|C--W|-|0|0|9115038255631187454|-]: Connection created [07:17:12:INFO] [001:144] [37635] (p2p_transport_channel.cc:1264): Channel[0|1|__]: Created connection with origin: 2, total: 1 [07:17:12:INFO] [001:144] [1027] (webrtc_sdp.cc:3278): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [07:17:12:INFO] [001:144] [37635] (p2p_transport_channel.cc:1914): Channel[0|1|__]: Transport channel state changed from 0 to 2 [07:17:12:INFO] [001:144] [37635] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete. [07:17:12:INFO] [001:144] [37635] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete. [07:17:12:INFO] [001:144] [37635] (p2p_transport_channel.cc:1476): Channel[0|1|__]: Have a pingable connection for the first time; starting to ping. [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 1 [07:17:12:INFO] [001:144] [37635] (basic_port_allocator.cc:1017): Port[2a89ba00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates. [07:17:12:INFO] [001:145] [37635] (port.cc:1776): Conn[2a909600:0:Net[en0:192.168.144.0/22:Wifi:id=1]:fo7M2zeU:1:0:local:udp:192.168.147.82:58048->4Fga3Nc5:1:2122260223:local:udp:192.168.147.82:64902|C--W|-|0|0|9115038255631187454|-]: Sent STUN ping, id=653252634d3677636a616252, use_candidate=0, nomination=0 [07:17:12:INFO] [001:146] [25347] (port.cc:530): Received STUN ping id=653252634d3677636a616252 from unknown address 192.168.147.82:58048 [07:17:12:INFO] [001:146] [25347] (port.cc:1128): Conn[2b8ec400:0:Net[en0:192.168.144.0/22:Wifi:id=1]:7bZO7ZO0:1:0:local:udp:192.168.147.82:64902->uEdF/5iv:1:1853824767:prflx:udp:192.168.147.82:58048|C--W|-|0|0|7962116751024340479|-]: Connection created [07:17:12:INFO] [001:146] [25347] (p2p_transport_channel.cc:988): Adding connection from peer reflexive candidate: Cand[:81749522:1:udp:1853824767:192.168.147.82:58048:prflx::0:mZbg::1:10:0] [07:17:12:INFO] [001:146] [25347] (port.cc:855): Port[2a880400:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Sent STUN ping response, to=192.168.147.82:58048, id=653252634d3677636a616252 [07:17:12:INFO] [001:146] [25347] (p2p_transport_channel.cc:1914): Channel[0|1|R_]: Transport channel state changed from 0 to 2 [07:17:12:INFO] [001:146] [25347] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete. [07:17:12:INFO] [001:146] [25347] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete. [07:17:12:INFO] [001:146] [37635] (port.cc:1724): Conn[2a909600:0:Net[en0:192.168.144.0/22:Wifi:id=1]:fo7M2zeU:1:0:local:udp:192.168.147.82:58048->4Fga3Nc5:1:2122260223:local:udp:192.168.147.82:64902|C--I|-|0|0|9115038255631187454|-]: Received STUN ping response, id=653252634d3677636a616252, code=0, rtt=1, pings_since_last_response=653252634d3677636a616252 [07:17:12:INFO] [001:147] [37635] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed [07:17:12:INFO] [001:147] [37635] (p2p_transport_channel.cc:1852): Channel[0|1|__]: New selected connection: Conn[2a909600:0:Net[en0:192.168.144.0/22:Wifi:id=1]:fo7M2zeU:1:0:local:udp:192.168.147.82:58048->4Fga3Nc5:1:2122260223:local:udp:192.168.147.82:64902|CRWS|S|0|0|9115038255631187454|1] [07:17:12:INFO] [001:147] [37635] (channel.cc:376): Network route was changed. [07:17:12:INFO] [001:147] [37635] (channel.cc:376): Network route was changed. [07:17:12:INFO] [001:147] [37635] (dtls_transport.cc:811): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 1 [07:17:12:INFO] [001:147] [37635] (openssl_stream_adapter.cc:777): BeginSSL with peer. [07:17:12:INFO] [001:149] [25603] (webrtc_video_engine.cc:1349): Call stats: 12617936, {send_bw_bps: 0, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1} [07:17:12:INFO] [001:150] [25347] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0 [07:17:12:INFO] [001:150] [25347] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 29d20630 [07:17:12:INFO] [001:150] [25347] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=mZbg, renomination enabled [07:17:12:INFO] [001:150] [25347] (dtls_transport.cc:370): DtlsTransport[0|1|__]: DTLS setup complete. [07:17:12:INFO] [001:150] [25603] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [001:150] [25603] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [001:151] [25347] (p2p_transport_channel.cc:1476): Channel[0|1|R_]: Have a pingable connection for the first time; starting to ping. [07:17:12:INFO] [001:151] [25603] (channel.cc:553): Channel enabled [07:17:12:INFO] [001:151] [25347] (p2p_transport_channel.cc:2440): Selecting connection for triggered check: Conn[2b8ec400:0:Net[en0:192.168.144.0/22:Wifi:id=1]:7bZO7ZO0:1:0:local:udp:192.168.147.82:64902->uEdF/5iv:1:1853824767:prflx:udp:192.168.147.82:58048|CR-W|-|0|0|7962116751024340479|-] [07:17:12:INFO] [001:151] [25347] (port.cc:1776): Conn[2b8ec400:0:Net[en0:192.168.144.0/22:Wifi:id=1]:7bZO7ZO0:1:0:local:udp:192.168.147.82:64902->uEdF/5iv:1:1853824767:prflx:udp:192.168.147.82:58048|CR-W|-|0|0|7962116751024340479|-]: Sent STUN ping, id=436933494773704561794a5a, use_candidate=0, nomination=0 [07:17:12:INFO] [001:151] [25603] (channel.cc:863): Changing voice state, recv=1 send=0 [07:17:12:INFO] [001:151] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client enter_early_data [07:17:12:INFO] [001:151] [25603] (channel.cc:553): Channel enabled [07:17:12:INFO] [001:151] [25603] (channel.cc:996): Changing video state, send=0 [07:17:12:INFO] [001:151] [38403] (peer_connection.cc:4173): Session: 6407474718852373401 Old state: kHaveLocalOffer New state: kStable [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 0 [07:17:12:INFO] [001:152] [25603] (channel.cc:923): Setting remote voice description [07:17:12:INFO] [001:152] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client read_hello_verify_request [07:17:12:INFO] [001:152] [37635] (openssl_adapter.cc:794): SSL_connect:error in TLS client read_hello_verify_request [07:17:12:INFO] [001:152] [25347] (dtls_transport.cc:546): DtlsTransport[0|1|__]: Packet received before DTLS started. [07:17:12:INFO] [001:152] [37635] (dtls_transport.cc:698): DtlsTransport[0|1|__]: DtlsTransport: Started DTLS handshake [07:17:12:INFO] [001:152] [25347] (dtls_transport.cc:555): DtlsTransport[0|1|__]: Caching DTLS ClientHello packet until DTLS is started. [07:17:12:INFO] [001:152] [37635] (srtp_transport.cc:364): The params in SRTP transport are reset. [07:17:12:INFO] [001:152] [25603] (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 {}} [07:17:12:INFO] [001:152] [37635] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete. [07:17:12:INFO] [001:152] [37635] (basic_port_allocator.cc:1119): All candidates gathered for 0:1:0 [07:17:12:INFO] [001:152] [37635] (p2p_transport_channel.cc:854): P2PTransportChannel: 0, component 1 gathering complete [07:17:12:INFO] [001:152] [25603] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2766236256, extmap-allow-mixed: false, extensions: [], c_name: 1JnqTdn92iBG4iL0}, 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}}}} [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 2 [07:17:12:INFO] [001:152] [25603] (audio_encoder_opus.cc:758): Set Opus bitrate to 32000 bps. [07:17:12:INFO] [001:152] [25603] (webrtc_voice_engine.cc:1702): Recreate all the receive streams because the send codec has changed. [07:17:12:INFO] [001:152] [25347] (port.cc:1724): Conn[2b8ec400:0:Net[en0:192.168.144.0/22:Wifi:id=1]:7bZO7ZO0:1:0:local:udp:192.168.147.82:64902->uEdF/5iv:1:1853824767:prflx:udp:192.168.147.82:58048|CR-I|-|0|0|7962116751024340479|-]: Received STUN ping response, id=436933494773704561794a5a, code=0, rtt=1, pings_since_last_response=436933494773704561794a5a [07:17:12:INFO] [001:152] [25603] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2766236256, 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: 1JnqTdn92iBG4iL0}, 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}}}} [07:17:12:INFO] [001:153] [25347] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed [07:17:12:INFO] [001:153] [25347] (p2p_transport_channel.cc:1852): Channel[0|1|R_]: New selected connection: Conn[2b8ec400:0:Net[en0:192.168.144.0/22:Wifi:id=1]:7bZO7ZO0:1:0:local:udp:192.168.147.82:64902->uEdF/5iv:1:1853824767:prflx:udp:192.168.147.82:58048|CRWS|S|0|0|7962116751024340479|1] [07:17:12:INFO] [001:153] [25347] (channel.cc:376): Network route was changed. [07:17:12:INFO] [001:153] [25603] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 2766236256, 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: 1JnqTdn92iBG4iL0}, 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}}}} [07:17:12:INFO] [001:153] [25347] (channel.cc:376): Network route was changed. [07:17:12:INFO] [001:153] [25603] (webrtc_voice_engine.cc:2163): WebRtcVoiceMediaChannel::SetMaxSendBitrate. [07:17:12:INFO] [001:153] [25603] (webrtc_voice_engine.cc:1485): Setting voice channel options: AudioOptions {} [07:17:12:INFO] [001:153] [25347] (dtls_transport.cc:811): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 1 [07:17:12:INFO] [001:153] [25603] (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, } [07:17:12:INFO] [001:153] [25347] (openssl_stream_adapter.cc:777): BeginSSL with peer. [07:17:12:INFO] [001:153] [25603] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [07:17:12:INFO] [001:153] [25603] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [07:17:12:INFO] [001:153] [25603] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [07:17:12:INFO] [001:154] [25603] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [07:17:12:INFO] [001:154] [25603] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [07:17:12:INFO] [001:154] [25603] (audio_device_module_ios.mm:600): output: 0 [07:17:12:INFO] [001:154] [25603] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [001:154] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_hello [07:17:12:INFO] [001:154] [25347] (openssl_adapter.cc:794): SSL_accept:error in TLS server read_client_hello [07:17:12:INFO] [001:154] [25347] (dtls_transport.cc:698): DtlsTransport[0|1|__]: DtlsTransport: Started DTLS handshake [07:17:12:INFO] [001:154] [25603] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [001:154] [25347] (srtp_transport.cc:364): The params in SRTP transport are reset. [07:17:12:INFO] [001:154] [25603] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [001:154] [25347] (dtls_transport.cc:704): DtlsTransport[0|1|__]: Handling cached DTLS ClientHello packet. [07:17:12:INFO] [001:154] [25603] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [07:17:12:INFO] [001:154] [25603] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [07:17:12:INFO] [001:154] [25603] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [07:17:12:INFO] [001:154] [25603] (audio_device_module_ios.mm:616): output: 0 [07:17:12:INFO] [001:154] [25603] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [07:17:12:INFO] [001:154] [25603] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [07:17:12:INFO] [001:154] [25603] (audio_device_module_ios.mm:632): output: 0 [07:17:12:INFO] [001:154] [25603] (apm_helpers.cc:62): NS set to 0 [07:17:12:INFO] [001:154] [25603] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [07:17:12:INFO] [001:154] [25603] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [07:17:12:INFO] [001:154] [25603] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [07:17:12:INFO] [001:154] [25603] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [07:17:12:INFO] [001:154] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server select_certificate [07:17:12:INFO] [001:155] [25603] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [07:17:12:INFO] [001:155] [25603] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [07:17:12:INFO] [001:155] [25603] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [07:17:12:INFO] [001:155] [25603] (webrtc_voice_engine.cc:511): Setting AGC to 0 [07:17:12:INFO] [001:155] [25603] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [07:17:12:INFO] [001:155] [25603] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [001:155] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server select_parameters [07:17:12:INFO] [001:155] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_hello [07:17:12:INFO] [001:155] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_certificate [07:17:12:INFO] [001:155] [25603] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [001:155] [25603] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [001:155] [25603] (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, } [07:17:12:INFO] [001:155] [25603] (webrtc_voice_engine.cc:1849): AddRecvStream: {id:ARDAMSa0;ssrcs:[2391384973];ssrc_groups:;cname:uLPegaWZdV596BVM;stream_ids:ARDAMS;} [07:17:12:INFO] [001:155] [25603] (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 [07:17:12:INFO] [001:155] [25603] (audio_coding_module.cc:333): Created [07:17:12:INFO] [001:155] [25603] (audio_receive_stream.cc:117): AudioReceiveStream: 2391384973 [07:17:12:INFO] [001:155] [25603] (audio_receive_stream.cc:354): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 2391384973, local_ssrc: 2766236256, 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} [07:17:12:INFO] [001:156] [25603] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [001:156] [25603] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [001:156] [25603] (audio_device_module_ios.mm:543): Playing [07:17:12:INFO] [001:156] [25603] (audio_device_module_ios.mm:480): InitPlayout [07:17:12:INFO] [001:156] [25603] (audio_device_module_ios.mm:506): PlayoutIsInitialized [07:17:12:INFO] [001:156] [25603] (audio_device_ios.mm:185): AudioDeviceIOS::InitPlayout [07:17:12:INFO] [001:156] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_key_exchange [07:17:12:INFO] [001:156] [25603] (audio_device_ios.mm:859): AudioDeviceIOS::InitPlayOrRecord [07:17:12:INFO] [001:156] [25603] (RTCLogging.mm:33): (audio_device_ios.mm:824 ConfigureAudioSession): Configuring audio session. [07:17:12:INFO] [001:156] [25603] (RTCLogging.mm:33): (RTCAudioSession.mm:699 -[RTCAudioSession configureWebRTCSession:]): Configuring audio session for WebRTC. [07:17:12:INFO] [001:156] [25603] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:116 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred IO buffer duration to: 0.020000 [07:17:12:INFO] [001:156] [25603] (RTCLogging.mm:33): (RTCAudioSession.mm:624 -[RTCAudioSession incrementActivationCount]): Incrementing activation count. [07:17:12:INFO] [001:156] [25603] (RTCLogging.mm:33): (RTCAudioSession.mm:377 -[RTCAudioSession setActive:error:]): Number of current activations: 2 [07:17:12:INFO] [001:157] [25603] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:145 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set input number of channels to: 1 [07:17:12:INFO] [001:157] [25603] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:158 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set output number of channels to: 1 [07:17:12:INFO] [001:157] [25603] (RTCLogging.mm:33): (audio_device_ios.mm:835 ConfigureAudioSession): Configured audio session. [07:17:12:INFO] [001:157] [25603] (audio_device_ios.mm:678): AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession [07:17:12:INFO] [001:157] [25603] (RTCLogging.mm:33): (audio_device_ios.mm:683 SetupAudioBuffersForActiveAudioSession): RTCAudioSession: { [07:17:12:INFO] category: AVAudioSessionCategoryPlayAndRecord [07:17:12:INFO] categoryOptions: 4 [07:17:12:INFO] mode: AVAudioSessionModeVoiceChat [07:17:12:INFO] isActive: 1 [07:17:12:INFO] sampleRate: 48000.00 [07:17:12:INFO] IOBufferDuration: 0.010667 [07:17:12:INFO] outputNumberOfChannels: 2 [07:17:12:INFO] inputNumberOfChannels: 2 [07:17:12:INFO] outputLatency: 0.010000 [07:17:12:INFO] inputLatency: 0.010000 [07:17:12:INFO] outputVolume: 1.000000 [07:17:12:INFO] } [07:17:12:INFO] [001:157] [25603] (audio_device_ios.mm:714): frames per I/O buffer: 512 [07:17:12:INFO] [001:157] [25603] (audio_device_ios.mm:715): bytes per I/O buffer: 1024 [07:17:12:INFO] [001:157] [25603] (audio_device_ios.mm:662): AudioDeviceIOS::UpdateAudioDevicebuffer [07:17:12:INFO] [001:157] [25603] (audio_device_buffer.cc:186): SetPlayoutSampleRate(48000) [07:17:12:INFO] [001:157] [25603] (audio_device_buffer.cc:206): SetPlayoutChannels(1) [07:17:12:INFO] [001:157] [25603] (audio_device_buffer.cc:180): SetRecordingSampleRate(48000) [07:17:12:INFO] [001:157] [25603] (audio_device_buffer.cc:200): SetRecordingChannels(1) [07:17:12:INFO] [001:157] [25603] (fine_audio_buffer.cc:32): FineAudioBuffer [07:17:12:INFO] [001:157] [25603] (fine_audio_buffer.cc:34): playout_samples_per_channel_10ms: 480 [07:17:12:INFO] [001:157] [25603] (fine_audio_buffer.cc:36): playout_channels: 1 [07:17:12:INFO] [001:157] [25603] (fine_audio_buffer.cc:39): record_samples_per_channel_10ms: 480 [07:17:12:INFO] [001:157] [25603] (fine_audio_buffer.cc:41): record_channels: 1 [07:17:12:INFO] [001:157] [25603] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:196 Initialize): Initializing audio unit with sample rate: 48000.000000 [07:17:12:INFO] [001:157] [25603] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:43 LogStreamDescription): AudioStreamBasicDescription: { [07:17:12:INFO] mSampleRate: 48000.00 [07:17:12:INFO] formatIDString: lpcm [07:17:12:INFO] mFormatFlags: 0xC [07:17:12:INFO] mBytesPerPacket: 2 [07:17:12:INFO] mFramesPerPacket: 1 [07:17:12:INFO] mBytesPerFrame: 2 [07:17:12:INFO] mChannelsPerFrame: 1 [07:17:12:INFO] mBitsPerChannel: 16 [07:17:12:INFO] mReserved: 28672 [07:17:12:INFO] } [07:17:12:INFO] [001:157] [25603] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:250 Initialize): Voice Processing I/O unit is now initialized. [07:17:12:INFO] [001:158] [25603] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:71 GetAGCState): VPIO unit AGC: 0 [07:17:12:INFO] [001:158] [25603] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:264 Initialize): Failed to get AGC state (1st attempt). Error=-66631. [07:17:12:INFO] [001:158] [25603] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:302 Initialize): WebRTC.Audio.BuiltInAGCWasEnabledByDefault: 0 [07:17:12:INFO] [001:158] [25603] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:308 Initialize): WebRTC.Audio.BuiltInAGCIsEnabled: 0 [07:17:12:INFO] [001:158] [25603] (audio_device_module_ios.mm:486): output: 0 [07:17:12:INFO] [001:158] [25603] (audio_device_module_ios.mm:518): StartPlayout [07:17:12:INFO] [001:158] [25603] (audio_device_module_ios.mm:543): Playing [07:17:12:INFO] [001:158] [25603] (audio_device_buffer.cc:98): StartPlayout [07:17:12:INFO] [001:158] [25603] (audio_device_ios.mm:227): AudioDeviceIOS::StartPlayout [07:17:12:INFO] [001:158] [25603] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:316 Start): Starting audio unit. [07:17:12:INFO] [001:160] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_hello_done [07:17:12:INFO] [001:160] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_certificate [07:17:12:INFO] [001:160] [25347] (openssl_adapter.cc:794): SSL_accept:error in TLS server read_client_certificate [07:17:12:INFO] [001:160] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_hello [07:17:12:INFO] [001:160] [25347] (jsep_transport_controller.cc:1276): 0 Transport 1 state changed. Check if state is complete. [07:17:12:INFO] [001:160] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_certificate [07:17:12:INFO] [001:160] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client read_certificate_status [07:17:12:INFO] [001:160] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client verify_server_certificate [07:17:12:INFO] [001:160] [37635] (openssl_stream_adapter.cc:1049): Accepted peer certificate. [07:17:12:INFO] [001:160] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_key_exchange [07:17:12:INFO] [001:160] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client read_certificate_request [07:17:12:INFO] [001:160] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_hello_done [07:17:12:INFO] [001:160] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_certificate [07:17:12:INFO] [001:160] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_key_exchange [07:17:12:INFO] [001:162] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_certificate_verify [07:17:12:INFO] [001:164] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_finished [07:17:12:INFO] [001:165] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client finish_flight [07:17:12:INFO] [001:165] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client read_session_ticket [07:17:12:INFO] [001:165] [37635] (openssl_adapter.cc:794): SSL_connect:error in TLS client read_session_ticket [07:17:12:INFO] [001:165] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server verify_client_certificate [07:17:12:INFO] [001:165] [25347] (openssl_stream_adapter.cc:1049): Accepted peer certificate. [07:17:12:INFO] [001:165] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_key_exchange [07:17:12:INFO] [001:166] [25603] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:320 Start): Failed to start audio unit. Error=-66637 [07:17:12:INFO] [001:166] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_certificate_verify [07:17:12:INFO] [001:166] [25603] (RTCLogging.mm:33): (audio_device_ios.mm:237 StartPlayout): StartPlayout failed to start audio unit. [07:17:12:INFO] [001:166] [25603] (audio_device_module_ios.mm:525): output: -1 [07:17:12:INFO] [001:166] [25603] (channel.cc:719): Add remote ssrc: 2391384973 [07:17:12:INFO] [001:166] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server read_change_cipher_spec [07:17:12:INFO] [001:167] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server process_change_cipher_spec [07:17:12:INFO] [001:167] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server read_next_proto [07:17:12:INFO] [001:167] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server read_channel_id [07:17:12:INFO] [001:167] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_finished [07:17:12:INFO] [001:167] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_finished [07:17:12:INFO] [001:167] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server finish_server_handshake [07:17:12:INFO] [001:167] [25347] (openssl_adapter.cc:784): SSL_accept:TLS server done [07:17:12:INFO] [001:167] [25347] (dtls_transport.cc:630): DtlsTransport[0|1|__]: DTLS handshake complete. [07:17:12:INFO] [001:167] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client process_change_cipher_spec [07:17:12:INFO] [001:167] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_finished [07:17:12:INFO] [001:167] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client finish_client_handshake [07:17:12:INFO] [001:167] [25347] (jsep_transport_controller.cc:1213): Transport 0 writability changed to 1. [07:17:12:INFO] [001:167] [37635] (openssl_adapter.cc:784): SSL_connect:TLS client done [07:17:12:INFO] [001:167] [37635] (dtls_transport.cc:630): DtlsTransport[0|1|__]: DTLS handshake complete. [07:17:12:INFO] [001:167] [25347] (dtls_srtp_transport.cc:219): Extracting keys from transport: 0 [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 2 [07:17:12:INFO] [001:168] [37635] (jsep_transport_controller.cc:1213): Transport 0 writability changed to 1. [07:17:12:INFO] [001:168] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [07:17:12:INFO] [001:168] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [07:17:12:INFO] [001:168] [37123] (peer_connection.cc:6044): Changing to ICE connected state because all transports are writable. [07:17:12:INFO] [001:168] [37635] (dtls_srtp_transport.cc:219): Extracting keys from transport: 0 [07:17:12:INFO] [001:168] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [07:17:12:INFO] [001:168] [37123] (peer_connection.cc:4109): Changing IceConnectionState 1 => 2 [07:17:12:INFO] [001:168] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 2 [07:17:12:INFO] [001:168] [13083] (rtp_transport_controller_send.cc:459): Creating fallback congestion controller [07:17:12:INFO] [001:169] [13083] (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 [07:17:12:INFO] [001:169] [13083] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85 [07:17:12:INFO] [001:169] [13083] (delay_based_bwe.cc:107): Using Trendline filter for delay change estimation with window size 20 [07:17:12:INFO] [001:169] [13083] (delay_based_bwe.cc:297): BWE Setting start bitrate to: 300 kbps [07:17:12:INFO] [001:170] [13083] (probe_controller.cc:266): Measured bitrate: 300000 Minimum to probe further: 1260000 [07:17:12:INFO] [001:170] [13083] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (900000:1687:5) [07:17:12:INFO] [001:170] [13083] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5) [07:17:12:INFO] [001:170] [13083] (bitrate_allocator.cc:115): Current BWE 300000 [07:17:12:INFO] [001:174] [37635] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [07:17:12:INFO] [001:174] [25347] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [07:17:12:INFO] [001:174] [37635] (channel.cc:583): Channel writable (0) for the first time [07:17:12:INFO] [001:174] [25347] (channel.cc:583): Channel writable (0) for the first time [07:17:12:INFO] [001:174] [37635] (channel.cc:583): Channel writable (1) for the first time [07:17:12:INFO] [001:174] [25347] (channel.cc:583): Channel writable (1) for the first time [07:17:12:INFO] [001:174] [37379] (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, } [07:17:12:INFO] [001:174] [37379] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [07:17:12:INFO] [001:174] [37379] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [07:17:12:INFO] [001:174] [37379] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [07:17:12:INFO] [001:174] [37379] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [07:17:12:INFO] [001:174] [37379] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [07:17:12:INFO] [001:174] [25603] (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, } [07:17:12:INFO] [001:174] [37379] (audio_device_module_ios.mm:600): output: 0 [07:17:12:INFO] [001:174] [25603] (webrtc_voice_engine.cc:330): Always disable AEC on iOS. Use built-in instead. [07:17:12:INFO] [001:174] [37379] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [001:174] [25603] (webrtc_voice_engine.cc:357): Always disable NS on iOS. Use built-in instead. [07:17:12:INFO] [001:174] [25603] (webrtc_voice_engine.cc:368): Always disable AGC on iOS. Use built-in instead. [07:17:12:INFO] [001:174] [25603] (audio_device_module_ios.mm:597): BuiltInAECIsAvailable [07:17:12:INFO] [001:174] [25603] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [07:17:12:INFO] [001:174] [25603] (audio_device_module_ios.mm:600): output: 0 [07:17:12:INFO] [001:175] [37379] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [001:175] [25603] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [001:175] [37379] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [001:175] [37379] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [07:17:12:INFO] [001:175] [37379] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [07:17:12:INFO] [001:175] [37379] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [07:17:12:INFO] [001:175] [37379] (audio_device_module_ios.mm:616): output: 0 [07:17:12:INFO] [001:175] [37379] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [07:17:12:INFO] [001:175] [37379] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [07:17:12:INFO] [001:175] [37379] (audio_device_module_ios.mm:632): output: 0 [07:17:12:INFO] [001:175] [37379] (apm_helpers.cc:62): NS set to 0 [07:17:12:INFO] [001:175] [37379] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [07:17:12:INFO] [001:175] [37379] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [07:17:12:INFO] [001:175] [25603] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [001:175] [37379] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [07:17:12:INFO] [001:175] [25603] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [001:175] [37379] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [07:17:12:INFO] [001:175] [25603] (apm_helpers.cc:48): Echo control set to 0 with mode 0 [07:17:12:INFO] [001:175] [37379] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [07:17:12:INFO] [001:175] [25603] (audio_device_module_ios.mm:613): BuiltInAGCIsAvailable [07:17:12:INFO] [001:175] [37379] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [07:17:12:INFO] [001:175] [25603] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [07:17:12:INFO] [001:175] [37379] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [07:17:12:INFO] [001:175] [25603] (audio_device_module_ios.mm:616): output: 0 [07:17:12:INFO] [001:175] [37379] (webrtc_voice_engine.cc:511): Setting AGC to 0 [07:17:12:INFO] [001:175] [25603] (audio_device_module_ios.mm:629): BuiltInNSIsAvailable [07:17:12:INFO] [001:175] [37379] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [07:17:12:INFO] [001:175] [25603] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [07:17:12:INFO] [001:175] [37379] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [001:175] [25603] (audio_device_module_ios.mm:632): output: 0 [07:17:12:INFO] [001:175] [25603] (apm_helpers.cc:62): NS set to 0 [07:17:12:INFO] [001:175] [25603] (webrtc_voice_engine.cc:452): NetEq capacity is 200 [07:17:12:INFO] [001:175] [25603] (webrtc_voice_engine.cc:458): NetEq fast mode? 0 [07:17:12:INFO] [001:175] [25603] (webrtc_voice_engine.cc:464): NetEq minimum delay is 0 [07:17:12:INFO] [001:175] [25603] (webrtc_voice_engine.cc:470): NetEq handle reordered packets? 0 [07:17:12:INFO] [001:175] [25603] (webrtc_voice_engine.cc:481): Delay agnostic aec is enabled? 0 [07:17:12:INFO] [001:175] [25603] (webrtc_voice_engine.cc:491): Extended filter aec is enabled? 0 [07:17:12:INFO] [001:175] [25603] (webrtc_voice_engine.cc:501): Experimental ns is enabled? 0 [07:17:12:INFO] [001:176] [37379] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [001:176] [25603] (webrtc_voice_engine.cc:511): Setting AGC to 0 [07:17:12:INFO] [001:176] [37379] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [001:176] [25603] (webrtc_voice_engine.cc:533): Typing detection is enabled? 0 [07:17:12:INFO] [001:176] [37379] (audio_device_module_ios.mm:512): RecordingIsInitialized [07:17:12:INFO] [001:176] [25603] (audio_processing_impl.cc:727): Highpass filter activated: 1 [07:17:12:INFO] [001:176] [37379] (audio_device_module_ios.mm:574): Recording [07:17:12:INFO] [001:176] [37379] (audio_device_module_ios.mm:493): InitRecording [07:17:12:INFO] [001:176] [37379] (audio_device_module_ios.mm:512): RecordingIsInitialized [07:17:12:INFO] [001:176] [37379] (audio_device_module_ios.mm:549): StartRecording [07:17:12:INFO] [001:176] [25603] (audio_processing_impl.cc:745): Gain Controller 2 activated: 0 [07:17:12:INFO] [001:176] [37379] (audio_device_module_ios.mm:574): Recording [07:17:12:INFO] [001:176] [25603] (audio_processing_impl.cc:747): Pre-amplifier activated: 0 [07:17:12:INFO] [001:176] [37379] (audio_device_buffer.cc:117): StartRecording [07:17:12:INFO] [001:176] [25603] (audio_device_module_ios.mm:512): RecordingIsInitialized [07:17:12:INFO] [001:176] [37379] (audio_device_ios.mm:280): AudioDeviceIOS::StartRecording [07:17:12:INFO] [001:176] [25603] (audio_device_module_ios.mm:574): Recording [07:17:12:INFO] [001:176] [37379] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:316 Start): Starting audio unit. [07:17:12:INFO] [001:176] [25603] (audio_device_module_ios.mm:493): InitRecording [07:17:12:INFO] [001:176] [25603] (audio_device_module_ios.mm:512): RecordingIsInitialized [07:17:12:INFO] [001:177] [25603] (audio_device_module_ios.mm:549): StartRecording [07:17:12:INFO] [001:177] [25603] (audio_device_module_ios.mm:574): Recording [07:17:12:INFO] [001:177] [25603] (audio_device_buffer.cc:117): StartRecording [07:17:12:INFO] [001:177] [25603] (audio_device_ios.mm:280): AudioDeviceIOS::StartRecording [07:17:12:INFO] [001:177] [25603] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:316 Start): Starting audio unit. [07:17:12:INFO] [001:189] [37379] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:320 Start): Failed to start audio unit. Error=-66637 [07:17:12:INFO] [001:189] [37379] (RTCLogging.mm:33): (audio_device_ios.mm:290 StartRecording): StartRecording failed to start audio unit. [07:17:12:INFO] [001:189] [37379] (audio_device_module_ios.mm:556): output: -1 [07:17:12:INFO] [001:189] [37379] (channel.cc:863): Changing voice state, recv=1 send=1 [07:17:12:INFO] [001:190] [37379] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [07:17:12:INFO] [001:190] [13083] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers [07:17:12:INFO] [001:191] [13083] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps [07:17:12:INFO] [001:191] [37379] (channel.cc:996): Changing video state, send=1 [07:17:12:INFO] [001:191] [9219] (video_stream_encoder.cc:1568): Video suspend state changed to: not suspended [07:17:12:INFO] [001:201] [25603] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:320 Start): Failed to start audio unit. Error=-66637 [07:17:12:INFO] [001:201] [25603] (RTCLogging.mm:33): (audio_device_ios.mm:290 StartRecording): StartRecording failed to start audio unit. [07:17:12:INFO] [001:201] [25603] (audio_device_module_ios.mm:556): output: -1 [07:17:12:INFO] [001:201] [25603] (channel.cc:863): Changing voice state, recv=1 send=1 [07:17:12:INFO] [001:201] [25603] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [07:17:12:INFO] [001:201] [25603] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [07:17:12:INFO] [001:201] [25603] (channel.cc:1061): Setting remote video description [07:17:12:INFO] [001:201] [13083] (rtp_transport_controller_send.cc:459): Creating fallback congestion controller [07:17:12:INFO] [001:201] [25603] (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} [07:17:12:INFO] [001:201] [13083] (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 [07:17:12:INFO] [001:202] [25603] (webrtc_video_engine.cc:690): Using codec: VideoCodec[96:H264] [07:17:12:INFO] [001:202] [13083] (aimd_rate_control.cc:105): Using aimd rate control with back off factor 0.85 [07:17:12:INFO] [001:202] [13083] (delay_based_bwe.cc:107): Using Trendline filter for delay change estimation with window size 20 [07:17:12:INFO] [001:202] [25603] (webrtc_video_engine.cc:1883): RecreateWebRtcStream (send) because of SetCodec. [07:17:12:INFO] [001:202] [13083] (delay_based_bwe.cc:297): BWE Setting start bitrate to: 300 kbps [07:17:12:INFO] [001:202] [13083] (probe_controller.cc:266): Measured bitrate: 300000 Minimum to probe further: 1260000 [07:17:12:INFO] [001:202] [13083] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (900000:1687:5) [07:17:12:INFO] [001:202] [13083] (bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5) [07:17:12:INFO] [001:202] [25603] (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 [07:17:12:INFO] [001:202] [13083] (bitrate_allocator.cc:115): Current BWE 300000 [07:17:12:INFO] [001:203] [13083] (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. [07:17:12:INFO] [001:203] [13083] (video_send_stream_impl.cc:250): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [934729385], 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: [2198679567], payload_type: 97}, c_name: 1JnqTdn92iBG4iL0}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [07:17:12:INFO] [001:203] [13083] (video_send_stream_impl.cc:278): ERROR: Initial encoder max bitrate = -1 which is <= 0! [07:17:12:INFO] [001:203] [13083] (video_send_stream_impl.cc:280): Using default encoder max bitrate = 10 Mbps [07:17:12:INFO] [001:203] [25603] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [07:17:12:INFO] [001:203] [13083] (video_stream_encoder.cc:628): ConfigureEncoder requested. [07:17:12:INFO] [001:203] [25603] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [07:17:12:INFO] [001:204] [25603] (video_send_stream.cc:160): VideoSendStream::Stop [07:17:12:INFO] [001:204] [25603] (webrtc_video_engine.cc:758): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. [07:17:12:INFO] [001:204] [13083] (video_send_stream_impl.cc:417): VideoSendStream::Stop [07:17:12:INFO] [001:204] [25603] (webrtc_video_engine.cc:1174): AddRecvStream: {id:ARDAMSv0;ssrcs:[3428412742,4010987472];ssrc_groups:{semantics:FID;ssrcs:[3428412742,4010987472]};cname:uLPegaWZdV596BVM;stream_ids:ARDAMS;} [07:17:12:INFO] [001:205] [25603] (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: 3428412742, local_ssrc: 934729385, 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: 4010987472, 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} [07:17:12:INFO] [001:205] [25603] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [07:17:12:INFO] [001:205] [25603] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [07:17:12:INFO] [001:205] [25603] (channel.cc:719): Add remote ssrc: 3428412742 [07:17:12:INFO] [001:206] [25603] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [07:17:12:INFO] [001:206] [13083] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers [07:17:12:INFO] [001:206] [13083] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps [07:17:12:INFO] [001:206] [9219] (video_stream_encoder.cc:1568): Video suspend state changed to: not suspended [07:17:12:INFO] [001:206] [25603] (channel.cc:996): Changing video state, send=1 [07:17:12:INFO] [001:206] [25603] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [07:17:12:INFO] [001:206] [25603] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [07:17:12:INFO] [001:206] [38403] (peer_connection.cc:4109): Changing IceConnectionState 0 => 1 [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 1 [07:17:12:INFO] [001:206] [38403] (peer_connection.cc:5518): Non-rejected SCTP m= section is needed to get the SSL Role of the SCTP transport. [07:17:12:INFO] [001:206] [38403] (peer_connection.cc:2745): Processing the MSIDs for MID=0 (streams=[ARDAMS]). [07:17:12:INFO] [001:206] [25603] (channel.cc:863): Changing voice state, recv=1 send=1 [07:17:12:INFO] [001:206] [38403] (peer_connection.cc:2756): Processing the addition of a remote track for MID=0. [07:17:12:INFO] [001:206] [25603] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [07:17:12:INFO] [001:206] [38403] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv. [07:17:12:INFO] [001:206] [9219] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers [07:17:12:INFO] [001:206] [25603] (channel.cc:996): Changing video state, send=1 [07:17:12:INFO] [001:206] [25603] (webrtc_voice_engine.cc:1962): SetOutputVolume() to 1 for recv stream with ssrc 2391384973 [07:17:12:INFO] [001:206] [38403] (peer_connection.cc:2745): Processing the MSIDs for MID=1 (streams=[ARDAMS]). [07:17:12:INFO] [001:207] [38403] (peer_connection.cc:2756): Processing the addition of a remote track for MID=1. [07:17:12:INFO] [001:207] [38403] (rtp_transceiver.cc:169): Changing transceiver (MID=1) current direction from <not set> to kSendRecv. [07:17:12:INFO] [001:207] [25603] (webrtc_video_engine.cc:1304): SetSink: ssrc:3428412742 (ptr) [07:17:12:INFO] [001:207] [38403] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x6080002036b0): created DTMF sender: RTCDtmfSender { [07:17:12:INFO] remainingTones: [07:17:12:INFO] duration: 0.100000 sec [07:17:12:INFO] interToneGap: 0.050000 sec [07:17:12:INFO] } [07:17:12:INFO] [001:207] [38403] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x6080004220e0): created sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSa0 [07:17:12:INFO] } [07:17:12:INFO] [001:207] [38403] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x60800005d8b0): created receiver: RTCRtpReceiver { [07:17:12:INFO] receiverId: bcfa7252-dd67-479d-8b38-7be9a4d8d932 [07:17:12:INFO] } [07:17:12:INFO] [001:207] [38403] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x60800005e2a0): created transceiver: RTCRtpTransceiver { [07:17:12:INFO] sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSa0 [07:17:12:INFO] } [07:17:12:INFO] receiver: RTCRtpReceiver { [07:17:12:INFO] receiverId: bcfa7252-dd67-479d-8b38-7be9a4d8d932 [07:17:12:INFO] } [07:17:12:INFO] } [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving audio on track bcfa7252-dd67-479d-8b38-7be9a4d8d932. [07:17:12:INFO] [001:207] [38403] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [07:17:12:INFO] [001:207] [38403] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x608000421fa0): created sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSv0 [07:17:12:INFO] } [07:17:12:INFO] [001:207] [38403] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x60800005cec0): created receiver: RTCRtpReceiver { [07:17:12:INFO] receiverId: ce5f386a-9d17-4ef2-b677-7dccf8793da3 [07:17:12:INFO] } [07:17:12:INFO] [001:207] [38403] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x60800005e2a0): created transceiver: RTCRtpTransceiver { [07:17:12:INFO] sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSv0 [07:17:12:INFO] } [07:17:12:INFO] receiver: RTCRtpReceiver { [07:17:12:INFO] receiverId: ce5f386a-9d17-4ef2-b677-7dccf8793da3 [07:17:12:INFO] } [07:17:12:INFO] } [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving video on track ce5f386a-9d17-4ef2-b677-7dccf8793da3. [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:381 -[ARDAppClient peerConnection:didAddStream:]): Stream with 1 video tracks and 1 audio tracks was added. [07:17:12:INFO] [001:208] [38403] (message_queue.cc:517): Message took 62ms to dispatch. Posted from: SetRemoteDescription@../../api/peer_connection_proxy.h:104 [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 1 [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 2 [07:17:12:INFO] [001:208] [38403] (peer_connection.cc:6050): Changing to ICE completed state because all transports are complete. [07:17:12:INFO] [001:208] [38403] (peer_connection.cc:4109): Changing IceConnectionState 1 => 2 [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 2 [07:17:12:INFO] [001:208] [1027] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x608000203940): created DTMF sender: RTCDtmfSender { [07:17:12:INFO] remainingTones: [07:17:12:INFO] duration: 0.100000 sec [07:17:12:INFO] interToneGap: 0.050000 sec [07:17:12:INFO] } [07:17:12:INFO] [001:210] [38403] (peer_connection.cc:4109): Changing IceConnectionState 2 => 3 [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 3 [07:17:12:INFO] [001:210] [1027] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x60800023f540): created sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSa0 [07:17:12:INFO] } [07:17:12:INFO] [001:210] [37123] (rtp_sender.cc:572): Tried to get DTMF sender from video sender. [07:17:12:INFO] [001:210] [1027] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x608000421fa0): created sender: RTCRtpSender { [07:17:12:INFO] senderId: ARDAMSv0 [07:17:12:INFO] } [07:17:12:INFO] [001:211] [37635] (jsep_transport_controller.cc:318): Not adding candidate because the JsepTransport doesn't exist. Ignore it. [07:17:12:INFO] [001:258] [37379] (rtc_event_log_impl.cc:211): Stopping WebRTC event log. [07:17:12:INFO] [001:258] [37379] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped. [07:17:12:INFO] [001:262] [37379] (webrtc_video_engine.cc:2234): VideoSendStream stats: 12618049, {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: 3428412742, 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} [07:17:12:INFO] [001:262] [37379] (webrtc_video_engine.cc:2731): VideoReceiveStream stats: 12618049, {ssrc: 934729385, 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} [07:17:12:INFO] [001:263] [37379] (webrtc_video_engine.cc:1349): Call stats: 12618049, {send_bw_bps: 300000, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1} [07:17:12:INFO] [001:264] [37123] (peer_connection.cc:4173): Session: 7386157372738723833 Old state: kStable New state: kClosed [07:17:12:INFO] [001:265] [37379] (audio_device_module_ios.mm:563): StopRecording [07:17:12:INFO] [001:265] [37379] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording [07:17:12:INFO] [001:265] [37379] (audio_device_buffer.cc:153): StopRecording [07:17:12:INFO] [001:265] [37379] (audio_device_buffer.cc:176): total recording time: 89 [07:17:12:INFO] [001:265] [37379] (audio_device_module_ios.mm:567): output: 0 [07:17:12:INFO] [001:265] [37379] (webrtc_voice_engine.cc:1962): SetOutputVolume() to 0 for recv stream with ssrc 2766236256 [07:17:12:INFO] [001:266] [37379] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 3428412742, options: nullptr, source = nullptr) [07:17:12:INFO] [001:266] [37379] (webrtc_video_engine.cc:1304): SetSink: ssrc:934729385 nullptr [07:17:12:INFO] [001:267] [37379] (channel.cc:563): Channel disabled [07:17:12:INFO] [001:267] [37379] (video_send_stream.cc:160): VideoSendStream::Stop [07:17:12:INFO] [001:267] [37379] (channel.cc:996): Changing video state, send=0 [07:17:12:INFO] [001:267] [16907] (video_send_stream_impl.cc:417): VideoSendStream::Stop [07:17:12:INFO] [001:267] [16907] (rtp_rtcp_impl.cc:360): Failed to send RTCP BYE [07:17:12:INFO] [001:267] [16907] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps [07:17:12:INFO] [001:267] [37379] (video_send_stream.cc:160): VideoSendStream::Stop [07:17:12:INFO] [001:271] [16907] (video_send_stream_impl.cc:417): VideoSendStream::Stop [07:17:12:INFO] [001:271] [9731] (video_stream_encoder.cc:1568): Video suspend state changed to: suspended [07:17:12:INFO] [001:272] [9731] (video_send_stream_impl.cc:417): VideoSendStream::Stop [07:17:12:INFO] [001:272] [9731] (video_send_stream_impl.cc:338): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [3428412742], 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: [4010987472], payload_type: 97}, c_name: uLPegaWZdV596BVM}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [07:17:12:INFO] [001:278] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [07:17:12:INFO] [001:278] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [07:17:12:INFO] [001:279] [37379] (send_statistics_proxy.cc:655): Frames encoded 0 [07:17:12:INFO] WebRTC.Video.DroppedFrames.Capturer 0 [07:17:12:INFO] WebRTC.Video.DroppedFrames.EncoderQueue 0 [07:17:12:INFO] WebRTC.Video.DroppedFrames.Encoder 0 [07:17:12:INFO] WebRTC.Video.DroppedFrames.Ratelimiter 0 [07:17:12:INFO] [001:279] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [07:17:12:INFO] [001:279] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [07:17:12:INFO] [001:279] [37379] (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: 934729385, local_ssrc: 3428412742, 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: 2198679567, 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} [07:17:12:INFO] [001:288] [37379] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [07:17:12:INFO] [001:288] [37379] (receive_statistics_proxy.cc:487): Frames decoded 0 [07:17:12:INFO] WebRTC.Video.DroppedFrames.Receiver 0 [07:17:12:INFO] [07:17:12:INFO] [001:288] [37379] (channel.cc:162): Destroyed channel: 1 [07:17:12:INFO] [001:289] [37379] (channel.cc:563): Channel disabled [07:17:12:INFO] [001:289] [37379] (audio_device_module_ios.mm:532): StopPlayout [07:17:12:INFO] [001:289] [37379] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout [07:17:12:INFO] [001:289] [37379] (audio_device_buffer.cc:139): StopPlayout [07:17:12:INFO] [001:289] [37379] (audio_device_buffer.cc:145): total playout time: 168 [07:17:12:INFO] [001:289] [37379] (audio_device_module_ios.mm:536): output: 0 [07:17:12:INFO] [001:289] [37379] (channel.cc:863): Changing voice state, recv=0 send=0 [07:17:12:INFO] [001:289] [37379] (webrtc_voice_engine.cc:1823): RemoveSendStream: 2391384973 [07:17:12:INFO] [001:289] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [07:17:12:INFO] [001:289] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [07:17:12:INFO] [001:289] [37379] (audio_send_stream.cc:179): ~AudioSendStream: 2391384973 [07:17:12:INFO] [001:290] [37379] (webrtc_voice_engine.cc:1900): RemoveRecvStream: 2766236256 [07:17:12:INFO] [001:290] [37379] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [001:300] [37379] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [001:301] [37379] (audio_receive_stream.cc:140): ~AudioReceiveStream: 2766236256 [07:17:12:INFO] [001:301] [16907] (paced_sender.cc:106): PacedSender paused. [07:17:12:INFO] [001:301] [16907] (control_handler.cc:77): Bitrate estimate state changed, BWE: 300 kbps. [07:17:12:INFO] [001:301] [37379] (channel.cc:162): Destroyed channel: 0 [07:17:12:INFO] [001:301] [37635] (openssl_stream_adapter.cc:896): Cleanup [07:17:12:INFO] [001:301] [37635] (openssl_adapter.cc:787): SSL3 alert write:warning:close notify [07:17:12:INFO] [001:302] [25347] (openssl_adapter.cc:787): SSL3 alert read:warning:close notify [07:17:12:INFO] [001:302] [25347] (openssl_stream_adapter.cc:896): Cleanup [07:17:12:INFO] [001:302] [25347] (openssl_adapter.cc:787): SSL3 alert write:warning:close notify [07:17:12:INFO] [001:302] [25347] (dtls_transport.cc:651): DtlsTransport[0|1|_W]: DTLS transport closed [07:17:12:INFO] [001:302] [25347] (jsep_transport_controller.cc:1213): Transport 0 writability changed to 0. [07:17:12:INFO] [001:302] [38403] (peer_connection.cc:4109): Changing IceConnectionState 3 => 5 [07:17:12:INFO] [001:302] [25347] (channel.cc:596): Channel not writable (0) [07:17:12:INFO] [001:302] [37379] (paced_sender.cc:418): ProcessThreadAttached 0x0 [07:17:12:INFO] (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 5 [07:17:12:INFO] [001:305] [37379] (paced_sender.cc:418): ProcessThreadAttached 0x0 [07:17:12:INFO] [001:305] [25347] (channel.cc:596): Channel not writable (1) [07:17:12:INFO] [001:305] [25347] (srtp_transport.cc:364): The params in SRTP transport are reset. [07:17:12:INFO] [001:306] [25603] (channel.cc:863): Changing voice state, recv=1 send=1 [07:17:12:INFO] [001:306] [37379] (rtc_event_log_impl.cc:211): Stopping WebRTC event log. [07:17:12:INFO] [001:306] [25603] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [07:17:12:INFO] [001:306] [37379] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped. [07:17:12:INFO] [001:306] [16907] (video_send_stream_impl.cc:363): VideoSendStream::UpdateActiveSimulcastLayers [07:17:12:INFO] [001:306] [25603] (channel.cc:996): Changing video state, send=1 [07:17:12:INFO] [001:306] [37123] (peer_connection.cc:6776): Usage signature is 4088 [07:17:12:INFO] [001:310] [37123] (message_queue.cc:517): Message took 51ms to dispatch. Posted from: Close@../../api/peer_connection_proxy.h:140 [07:17:12:INFO] [001:310] [37123] (peer_connection.cc:915): Session: 7386157372738723833 is destroyed. [07:17:12:INFO] [001:311] [37123] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue. [07:17:12:INFO] [001:311] [37379] (webrtc_video_engine.cc:474): WebRtcVideoEngine::~WebRtcVideoEngine [07:17:12:INFO] [001:311] [37379] (webrtc_voice_engine.cc:205): WebRtcVoiceEngine::~WebRtcVoiceEngine [07:17:12:INFO] [001:311] [37379] (audio_device_module_ios.mm:532): StopPlayout [07:17:12:INFO] [001:311] [37379] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout [07:17:12:INFO] [001:311] [37379] (audio_device_module_ios.mm:536): output: 0 [07:17:12:INFO] [001:311] [37379] (audio_device_module_ios.mm:563): StopRecording [07:17:12:INFO] [001:312] [37379] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording [07:17:12:INFO] [001:312] [37379] (audio_device_module_ios.mm:567): output: 0 [07:17:12:INFO] [001:312] [37379] (audio_device_module_ios.mm:581): RegisterAudioCallback [07:17:12:INFO] [001:312] [37379] (audio_device_buffer.cc:81): RegisterAudioCallback [07:17:12:INFO] [001:313] [37379] (audio_device_module_ios.mm:92): Terminate [07:17:12:INFO] [001:313] [37379] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate [07:17:12:INFO] [001:313] [37379] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout [07:17:12:INFO] [001:313] [37379] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording [07:17:12:INFO] [001:315] [37379] (audio_device_module_ios.mm:55): ~AudioDeviceModuleIOS [07:17:12:INFO] [001:315] [37379] (audio_device_buffer.cc:75): AudioDeviceBuffer::~dtor [07:17:12:INFO] [001:315] [37379] (audio_device_ios.mm:127): AudioDeviceIOS::~dtor<NSThread: 0x608000273480>{number = 4, name = (null)} [07:17:12:INFO] [001:315] [37379] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate [07:17:12:INFO] [001:315] [37379] (fine_audio_buffer.cc:46): ~FineAudioBuffer [07:17:12:INFO] [001:315] [37379] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:347 Uninitialize): Unintializing audio unit. [07:17:12:INFO] [001:316] [37379] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:354 Uninitialize): Uninitialized audio unit. [07:17:12:INFO] [001:316] [37379] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:459 DisposeAudioUnit): Disposing audio unit. [07:17:12:INFO] Test Case '-[ARDAppClientTest testSession]' passed (1.332 seconds). [07:17:12:INFO] Test Suite 'ARDAppClientTest' passed at 2019-04-17 14:17:12.449. [07:17:12:INFO] Executed 1 test, with 0 failures (0 unexpected) in 1.332 (1.333) seconds [07:17:12:INFO] Test Suite 'ARDFileCaptureControllerTests' started at 2019-04-17 14:17:12.450 [07:17:12:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStarted]' started. [07:17:12:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStarted]' passed (0.001 seconds). [07:17:12:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStoped]' started. [07:17:12:INFO] Test Case '-[ARDFileCaptureControllerTests testCaptureIsStoped]' passed (0.001 seconds). [07:17:12:INFO] Test Suite 'ARDFileCaptureControllerTests' passed at 2019-04-17 14:17:12.453. [07:17:12:INFO] Executed 2 tests, with 0 failures (0 unexpected) in 0.002 (0.003) seconds [07:17:12:INFO] Test Suite 'ARDSettingsModelTests' started at 2019-04-17 14:17:12.454 [07:17:12:INFO] Test Case '-[ARDSettingsModelTests testConstraintComponentIsNilWhenInvalidConstraintString]' started. [07:17:12:INFO] Test Case '-[ARDSettingsModelTests testConstraintComponentIsNilWhenInvalidConstraintString]' passed (0.002 seconds). [07:17:12:INFO] Test Case '-[ARDSettingsModelTests testHeightConstraintFromStore]' started. [07:17:12:INFO] Test Case '-[ARDSettingsModelTests testHeightConstraintFromStore]' passed (0.001 seconds). [07:17:12:INFO] Test Case '-[ARDSettingsModelTests testRetrievingSetting]' started. [07:17:12:INFO] Test Case '-[ARDSettingsModelTests testRetrievingSetting]' passed (0.001 seconds). [07:17:12:INFO] Test Case '-[ARDSettingsModelTests testReturningDefaultCallOption]' started. [07:17:12:INFO] Test Case '-[ARDSettingsModelTests testReturningDefaultCallOption]' passed (0.001 seconds). [07:17:12:INFO] Test Case '-[ARDSettingsModelTests testStoringAudioSetting]' started. [07:17:12:INFO] Test Case '-[ARDSettingsModelTests testStoringAudioSetting]' passed (0.001 seconds). [07:17:12:INFO] Test Case '-[ARDSettingsModelTests testStoringInvalidConstraintReturnsNo]' started. [07:17:12:INFO] Test Case '-[ARDSettingsModelTests testStoringInvalidConstraintReturnsNo]' passed (0.001 seconds). [07:17:12:INFO] Test Case '-[ARDSettingsModelTests testWidthConstraintFromStore]' started. [07:17:12:INFO] Test Case '-[ARDSettingsModelTests testWidthConstraintFromStore]' passed (0.002 seconds). [07:17:12:INFO] Test Suite 'ARDSettingsModelTests' passed at 2019-04-17 14:17:12.471. [07:17:12:INFO] Executed 7 tests, with 0 failures (0 unexpected) in 0.010 (0.017) seconds [07:17:12:INFO] Test Suite 'apprtcmobile_tests.app' passed at 2019-04-17 14:17:12.471. [07:17:12:INFO] Executed 10 tests, with 0 failures (0 unexpected) in 1.344 (1.356) seconds [07:17:12:INFO] Test Suite 'All tests' passed at 2019-04-17 14:17:12.472. [07:17:12:INFO] Executed 10 tests, with 0 failures (0 unexpected) in 1.344 (1.357) seconds [07:17:12:INFO] [001:347] [25603] (rtc_event_log_impl.cc:211): Stopping WebRTC event log. [07:17:12:INFO] [001:351] [25603] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped. [07:17:12:INFO] [001:353] [38403] (peer_connection.cc:4173): Session: 6407474718852373401 Old state: kStable New state: kClosed [07:17:12:INFO] [001:353] [25603] (audio_device_module_ios.mm:563): StopRecording [07:17:12:INFO] [001:353] [25603] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording [07:17:12:INFO] [001:353] [25603] (audio_device_buffer.cc:153): StopRecording [07:17:12:INFO] [001:353] [25603] (audio_device_buffer.cc:176): total recording time: 176 [07:17:12:INFO] [001:353] [25603] (audio_device_module_ios.mm:567): output: 0 [07:17:12:INFO] [001:353] [25603] (webrtc_voice_engine.cc:1962): SetOutputVolume() to 0 for recv stream with ssrc 2391384973 [07:17:12:INFO] [001:354] [25603] (webrtc_video_engine.cc:1031): SetVideoSend (ssrc= 934729385, options: nullptr, source = nullptr) [07:17:12:INFO] [001:354] [25603] (webrtc_video_engine.cc:1304): SetSink: ssrc:3428412742 nullptr [07:17:12:INFO] [001:354] [25603] (channel.cc:563): Channel disabled [07:17:12:INFO] [001:354] [25603] (video_send_stream.cc:160): VideoSendStream::Stop [07:17:12:INFO] [001:354] [25603] (channel.cc:996): Changing video state, send=0 [07:17:12:INFO] [001:354] [9731] (video_send_stream_impl.cc:417): VideoSendStream::Stop [07:17:12:INFO] [001:359] [9731] (rtp_rtcp_impl.cc:360): Failed to send RTCP BYE [07:17:12:INFO] [001:359] [9731] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps [07:17:12:INFO] [001:359] [9731] (video_stream_encoder.cc:1568): Video suspend state changed to: suspended [07:17:12:INFO] [001:359] [25603] (video_send_stream.cc:160): VideoSendStream::Stop [07:17:12:INFO] [001:359] [9731] (video_send_stream_impl.cc:417): VideoSendStream::Stop [07:17:12:INFO] [001:359] [13083] (video_send_stream_impl.cc:417): VideoSendStream::Stop [07:17:12:INFO] [001:359] [13083] (video_send_stream_impl.cc:338): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [934729385], 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: [2198679567], payload_type: 97}, c_name: 1JnqTdn92iBG4iL0}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [07:17:12:INFO] [001:365] [25603] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [07:17:12:INFO] [001:367] [25603] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [07:17:12:INFO] [001:367] [25603] (send_statistics_proxy.cc:655): Frames encoded 0 [07:17:12:INFO] WebRTC.Video.DroppedFrames.Capturer 0 [07:17:12:INFO] WebRTC.Video.DroppedFrames.EncoderQueue 0 [07:17:12:INFO] WebRTC.Video.DroppedFrames.Encoder 0 [07:17:12:INFO] WebRTC.Video.DroppedFrames.Ratelimiter 0 [07:17:12:INFO] [001:367] [25603] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [07:17:12:INFO] [001:367] [25603] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [07:17:12:INFO] [001:368] [25603] (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: 3428412742, local_ssrc: 934729385, 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: 4010987472, 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} [07:17:12:INFO] [001:376] [25603] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [07:17:12:INFO] [001:376] [25603] (receive_statistics_proxy.cc:487): Frames decoded 0 [07:17:12:INFO] WebRTC.Video.DroppedFrames.Receiver 0 [07:17:12:INFO] [07:17:12:INFO] [001:376] [25603] (channel.cc:162): Destroyed channel: 1 [07:17:12:INFO] [001:377] [25603] (channel.cc:563): Channel disabled [07:17:12:INFO] [001:377] [25603] (audio_device_module_ios.mm:532): StopPlayout [07:17:12:INFO] [001:377] [25603] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout [07:17:12:INFO] [001:377] [25603] (audio_device_buffer.cc:139): StopPlayout [07:17:12:INFO] [001:377] [25603] (audio_device_buffer.cc:145): total playout time: 219 [07:17:12:INFO] [001:377] [25603] (audio_device_module_ios.mm:536): output: 0 [07:17:12:INFO] [001:377] [25603] (channel.cc:863): Changing voice state, recv=0 send=0 [07:17:12:INFO] [001:377] [25603] (webrtc_voice_engine.cc:1823): RemoveSendStream: 2766236256 [07:17:12:INFO] [001:377] [25603] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [07:17:12:INFO] [001:377] [25603] (rtp_transport_controller_send.cc:262): SignalNetworkState Up [07:17:12:INFO] [001:377] [25603] (audio_send_stream.cc:179): ~AudioSendStream: 2766236256 [07:17:12:INFO] [001:377] [25603] (webrtc_voice_engine.cc:1900): RemoveRecvStream: 2391384973 [07:17:12:INFO] [001:377] [25603] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [07:17:12:INFO] [001:382] [25603] (rtp_transport_controller_send.cc:262): SignalNetworkState Down [07:17:12:INFO] [001:382] [25603] (audio_receive_stream.cc:140): ~AudioReceiveStream: 2391384973 [07:17:12:INFO] [001:382] [9731] (paced_sender.cc:106): PacedSender paused. [07:17:12:INFO] [001:382] [9731] (control_handler.cc:77): Bitrate estimate state changed, BWE: 300 kbps. [07:17:12:INFO] [001:382] [25603] (channel.cc:162): Destroyed channel: 0 [07:17:12:INFO] [001:383] [25347] (openssl_stream_adapter.cc:896): Cleanup [07:17:12:INFO] [001:383] [25603] (paced_sender.cc:418): ProcessThreadAttached 0x0 [07:17:12:INFO] [001:383] [25603] (paced_sender.cc:418): ProcessThreadAttached 0x0 [07:17:12:INFO] [001:383] [25603] (rtc_event_log_impl.cc:211): Stopping WebRTC event log. [07:17:12:INFO] [001:383] [25603] (rtc_event_log_impl.cc:228): WebRTC event log successfully stopped. [07:17:12:INFO] [001:383] [38403] (peer_connection.cc:6776): Usage signature is 4088 [07:17:12:INFO] [001:384] [38403] (peer_connection.cc:915): Session: 6407474718852373401 is destroyed. [07:17:12:INFO] [001:384] [38403] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue. [07:17:12:INFO] [001:387] [25603] (webrtc_video_engine.cc:474): WebRtcVideoEngine::~WebRtcVideoEngine [07:17:12:INFO] [001:387] [25603] (webrtc_voice_engine.cc:205): WebRtcVoiceEngine::~WebRtcVoiceEngine [07:17:12:INFO] [001:387] [25603] (audio_device_module_ios.mm:532): StopPlayout [07:17:12:INFO] [001:387] [25603] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout [07:17:12:INFO] [001:387] [25603] (audio_device_module_ios.mm:536): output: 0 [07:17:12:INFO] [001:387] [25603] (audio_device_module_ios.mm:563): StopRecording [07:17:12:INFO] [001:387] [25603] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording [07:17:12:INFO] [001:387] [25603] (audio_device_module_ios.mm:567): output: 0 [07:17:12:INFO] [001:387] [25603] (audio_device_module_ios.mm:581): RegisterAudioCallback [07:17:12:INFO] [001:387] [25603] (audio_device_buffer.cc:81): RegisterAudioCallback [07:17:12:INFO] [001:387] [25603] (audio_device_module_ios.mm:92): Terminate [07:17:12:INFO] [001:387] [25603] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate [07:17:12:INFO] [001:387] [25603] (audio_device_ios.mm:249): AudioDeviceIOS::StopPlayout [07:17:13:INFO] [001:387] [25603] (audio_device_ios.mm:300): AudioDeviceIOS::StopRecording [07:17:13:INFO] [001:389] [25603] (audio_device_module_ios.mm:55): ~AudioDeviceModuleIOS [07:17:13:INFO] [001:391] [25603] (audio_device_buffer.cc:75): AudioDeviceBuffer::~dtor [07:17:13:INFO] [001:392] [25603] (audio_device_ios.mm:127): AudioDeviceIOS::~dtor<NSThread: 0x600001278c80>{number = 3, name = (null)} [07:17:13:INFO] [001:392] [25603] (audio_device_ios.mm:168): AudioDeviceIOS::Terminate [07:17:13:INFO] [001:392] [25603] (fine_audio_buffer.cc:46): ~FineAudioBuffer [07:17:13:INFO] [001:392] [25603] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:347 Uninitialize): Unintializing audio unit. [07:17:13:INFO] [001:392] [25603] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:354 Uninitialize): Uninitialized audio unit. [07:17:13:INFO] [001:392] [25603] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:459 DisposeAudioUnit): Disposing audio unit. [07:17:13:INFO] [07:17:13:INFO] [07:17:13:INFO] Test session results and logs: [07:17:13:INFO] /Users/chrome-bot/Library/Developer/Xcode/DerivedData/temporary-cysxxxcydjplqwhhkapbtbilmqwj/Logs/Test/Test-Transient Testing-2019.04.17_07-16-51--0700.xcresult [07:17:13:INFO] [07:17:13:INFO] 2019-04-17 07:17:12.937 xcodebuild[18734:512735] [MT] IDETestOperationsObserverDebug: 21.214 elapsed -- Testing started completed. [07:17:13:INFO] 2019-04-17 07:17:12.937 xcodebuild[18734:512735] [MT] IDETestOperationsObserverDebug: 0.000 sec, +0.000 sec -- start [07:17:13:INFO] 2019-04-17 07:17:12.937 xcodebuild[18734:512735] [MT] IDETestOperationsObserverDebug: 21.214 sec, +21.214 sec -- end [07:17:13:INFO] ** TEST EXECUTE SUCCEEDED ** [07:17:13:INFO] [07:17:19:DEBUG] Finished print_process_output. [07:17:19:INFO] Waiting for test process to terminate. [07:17:19:INFO] Test process terminated. [07:17:19:DEBUG] Setting sigterm handler. [07:17:19:DEBUG] Stdout flushed after test process. [07:17:19:DEBUG] Processing test results. [07:17:19:INFO] /b/s/w/ir/src/out/Debug-iphonesimulator/iossim returned 0 [07:17:19:DEBUG] Extracting test data. [07:17:19:DEBUG] Retrieving crash reports. [07:17:19:DEBUG] Retrieving derived data. [07:17:19:DEBUG] Making desktop screenshots. [07:17:19:DEBUG] Killing simulators. [07:17:24:DEBUG] Wiping simulator. Device wiped. [07:17:26:DEBUG] End of tear_down. 2019-04-17 07:18:01,246 - root: [DEBUG] Contents of task_output_dir: ['/b/s/w/ir/k/recipe_cleanup/0xm25_5/44447987e87a1210'] 2019-04-17 07:18:01,246 - root: [WARNING] Expected output.json file missing: set(['/b/s/w/ir/k/recipe_cleanup/0xm25_5/44447987e87a1210/output.json']) Found: [] Expected: ['/b/s/w/ir/k/recipe_cleanup/0xm25_5/44447987e87a1210/output.json'] 2019-04-17 07:18:01,247 - root: [WARNING] No shard json files found in task_output_dir: '/b/s/w/ir/k/recipe_cleanup/0xm25_5' Found ['/b/s/w/ir/k/recipe_cleanup/0xm25_5/44447987e87a1210'] 2019-04-17 07:18:01,247 - root: [DEBUG] Found shard_json_files: ['/b/s/w/ir/k/recipe_cleanup/0xm25_5/44447987e87a1210/output.json'] 2019-04-17 07:18:01,247 - 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/tmpYm4t4M.json --task-output-dir /b/s/w/ir/k/recipe_cleanup/0xm25_5 -o /b/s/w/ir/tmp/t/tmpqYi_g6.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/tmpYm4t4M.json', '--task-output-dir', '/b/s/w/ir/k/recipe_cleanup/0xm25_5', '-o', '/b/s/w/ir/tmp/t/tmpqYi_g6.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/tmpYm4t4M.json', '--task-output-dir', '/b/s/w/ir/k/recipe_cleanup/0xm25_5', '-o', '/b/s/w/ir/tmp/t/tmpqYi_g6.json'] returned exit code 0