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'C:\b\s\w\ir\cipd_bin_packages\python.exe' -u 'C:\b\s\w\ir\kitchen-checkout\build\scripts\tools\runit.py' --show-path --with-third-party-lib -- python 'C:\b\s\w\ir\kitchen-checkout\build\scripts\slave\recipe_modules\chromium_swarming\resources\collect_task.py' --verbose -o 'c:\b\s\w\ir\tmp\t\tmpfmmwqi.json' --task-output-dir 'c:\b\s\w\ir\tmp\t\tmpyl4hti' --merge-script 'C:\b\s\w\ir\cache\builder\src\testing\merge_scripts\standard_isolated_script_merge.py' --merge-script-stdout-file 'c:\b\s\w\ir\tmp\t\tmpuso6mqmerge_script_log' --merge-additional-args '[]' --summary-json-file 'c:\b\s\w\ir\tmp\t\tmpqa8tdp.json' -- swarming collect -server https://chromium-swarm.appspot.com -worker 50 -task-summary-python -task-output-stdout all -verbose -requests-json 'c:\b\s\w\ir\tmp\t\tmprmvyvk.json' in dir C:\b\s\w\ir\k: allow_subannotations: False cmd: ['C:\\b\\s\\w\\ir\\cipd_bin_packages\\python.exe', '-u', 'C:\\b\\s\\w\\ir\\kitchen-checkout\\build\\scripts\\tools\\runit.py', '--show-path', '--with-third-party-lib', '--', 'python', 'C:\\b\\s\\w\\ir\\kitchen-checkout\\build\\scripts\\slave\\recipe_modules\\chromium_swarming\\resources\\collect_task.py', '--verbose', '-o', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpfmmwqi.json', '--task-output-dir', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti', '--merge-script', 'C:\\b\\s\\w\\ir\\cache\\builder\\src\\testing\\merge_scripts\\standard_isolated_script_merge.py', '--merge-script-stdout-file', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpuso6mqmerge_script_log', '--merge-additional-args', '[]', '--summary-json-file', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpqa8tdp.json', '--', 'swarming', 'collect', '-server', 'https://chromium-swarm.appspot.com', '-worker', '50', '-task-summary-python', '-task-output-stdout', 'all', '-verbose', '-requests-json', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmprmvyvk.json'] env: {} env_prefixes: {'PATH': 'C:\\b\\s\\w\\ir\\cache\\swarming_client'} env_suffixes: {} infra_step: False name_tokens: ('peerconnection_unittests',) ok_ret: frozenset([0]) step_test_data: <lambda>(...) trigger_specs: () full environment: ALLUSERSPROFILE: C:\ProgramData APPDATA: C:\Users\chrome-bot\AppData\Roaming BOTO_CONFIG: C:\b\s\w\ir\tmp\gsutil_task\.boto BUILDBUCKET_EXPERIMENTAL: FALSE CIPD_CACHE_DIR: C:\b\s\cipd_cache\cache CIPD_EXE_SHIM: "C:\b\s\w\ir\cipd_bin_packages\vpython.exe" CIPD_PROTOCOL: v2 COMMONPROGRAMFILES: C:\Program Files\Common Files COMMONPROGRAMFILES(X86): C:\Program Files (x86)\Common Files COMMONPROGRAMW6432: C:\Program Files\Common Files COMPUTERNAME: WIN10-WEBRTC-89 COMSPEC: C:\Windows\system32\cmd.exe DOCKER_CONFIG: C:\b\s\w\ir\tmp\docker_cfg_task DOCKER_TMPDIR: C:\b\s\w\ir\tmp\docker_tmp_task FPS_BROWSER_APP_PROFILE_STRING: Internet Explorer FPS_BROWSER_USER_PROFILE_STRING: Default GIT_CONFIG_NOSYSTEM: 1 GIT_TERMINAL_PROMPT: 0 HOMEDRIVE: C: HOMEPATH: \Users\chrome-bot INFRA_GIT_WRAPPER_HOME: C:\b\s\w\ir\tmp\git_home_task LC_ALL: English_United States.UTF-8 LC_COLLATE: English_United States.UTF-8 LC_CTYPE: English_United States.UTF-8 LC_MONETARY: English_United States.UTF-8 LC_NUMERIC: English_United States.UTF-8 LC_TIME: English_United States.UTF-8 LOCALAPPDATA: C:\Users\chrome-bot\AppData\Local LOGDOG_COORDINATOR_HOST: logs.chromium.org LOGDOG_STREAM_PREFIX: buildbucket/cr-buildbucket.appspot.com/8918339764559667952 LOGDOG_STREAM_PROJECT: webrtc LOGDOG_STREAM_SERVER_PATH: net.pipe:LUCILogDogKitchen_1624 LOGONSERVER: \\WIN10-WEBRTC-89 LUCI_CONTEXT: C:\b\s\w\itiffp6g\luci_context.877528610 MAC_CHROMIUM_TMPDIR: C:\b\s\w\ir\tmp\t NO_GCE_CHECK: False NUMBER_OF_PROCESSORS: 8 ONEDRIVE: C:\Users\chrome-bot\OneDrive OS: Windows_NT PATH: C:\b\s\w\ir\cache\swarming_client;C:\b\s\w\ir\cipd_bin_packages;C:\b\s\w\ir\cipd_bin_packages\bin;C:\b\s\cipd_cache\bin;C:\Windows\system32;C:\Windows;C:\Windows\System32\Wbem;C:\Windows\System32\WindowsPowerShell\v1.0\;C:\Program Files\Puppet Labs\Puppet\bin PATHEXT: .COM;.EXE;.BAT;.CMD;.VBS;.VBE;.JS;.JSE;.WSF;.WSH;.MSC PROCESSOR_ARCHITECTURE: AMD64 PROCESSOR_IDENTIFIER: Intel64 Family 6 Model 63 Stepping 0, GenuineIntel PROCESSOR_LEVEL: 6 PROCESSOR_REVISION: 3f00 PROGRAMDATA: C:\ProgramData PROGRAMFILES: C:\Program Files PROGRAMFILES(X86): C:\Program Files (x86) PROGRAMW6432: C:\Program Files PROMPT: $P$G PSMODULEPATH: C:\Program Files\WindowsPowerShell\Modules;C:\Windows\system32\WindowsPowerShell\v1.0\Modules PUBLIC: C:\Users\Public PYTHONIOENCODING: UTF-8 PYTHONUNBUFFERED: 1 SESSIONNAME: Console SWARMING_BOT_ID: win10-webrtc-8983f7d1-us-central1-c-lk83 SWARMING_EXTERNAL_BOT_SETUP: 1 SWARMING_HEADLESS: 1 SWARMING_SERVER: https://chromium-swarm.appspot.com SWARMING_TASK_ID: 43bb13c6c7d8f211 SYSTEMDRIVE: C: SYSTEMROOT: C:\Windows TEMP: C:\b\s\w\ir\tmp\t TEMPDIR: C:\b\s\w\ir\tmp\t TMP: C:\b\s\w\ir\tmp\t TMPDIR: C:\b\s\w\ir\tmp\t USERDOMAIN: WIN10-WEBRTC-89 USERDOMAIN_ROAMINGPROFILE: WIN10-WEBRTC-89 USERNAME: chrome-bot USERPROFILE: C:\Users\chrome-bot VPYTHON_VIRTUALENV_ROOT: C:\b\s\w\ir\cache\vpython WINDIR: C:\Windows Set PYTHONPATH: C:\b\s\w\ir\kitchen-checkout\build\scripts;C:\b\s\w\ir\kitchen-checkout\build\site_config;C:\b\s\w\ir\kitchen-checkout\build\third_party;C:\b\s\w\ir\kitchen-checkout\build\third_party\buildbot_8_4p1;C:\b\s\w\ir\kitchen-checkout\build\third_party\buildbot_slave_8_4;C:\b\s\w\ir\kitchen-checkout\build\third_party\coverage-3.7.1;C:\b\s\w\ir\kitchen-checkout\build\third_party\decorator_3_3_1;C:\b\s\w\ir\kitchen-checkout\build\third_party\google_api_python_client;C:\b\s\w\ir\kitchen-checkout\build\third_party\httplib2\python2;C:\b\s\w\ir\kitchen-checkout\build\third_party\infra_libs;C:\b\s\w\ir\kitchen-checkout\build\third_party\jinja2;C:\b\s\w\ir\kitchen-checkout\build\third_party\markupsafe;C:\b\s\w\ir\kitchen-checkout\build\third_party\oauth2client;C:\b\s\w\ir\kitchen-checkout\build\third_party\pyasn1;C:\b\s\w\ir\kitchen-checkout\build\third_party\pyasn1-modules;C:\b\s\w\ir\kitchen-checkout\build\third_party\python-rsa;C:\b\s\w\ir\kitchen-checkout\build\third_party\requests_2_10_0;C:\b\s\w\ir\kitchen-checkout\build\third_party\setuptools-0.6c11;C:\b\s\w\ir\kitchen-checkout\build\third_party\sqlalchemy_0_7_1;C:\b\s\w\ir\kitchen-checkout\build\third_party\sqlalchemy_migrate_0_7_1;C:\b\s\w\ir\kitchen-checkout\build\third_party\tempita_0_5;C:\b\s\w\ir\kitchen-checkout\build\third_party\twisted_10_2;C:\b\s\w\ir\kitchen-checkout\build\third_party\uritemplate;C:\b\s\w\ir\kitchen-checkout\build\third_party\site-packages 2019-03-21 15:04:38,611 - root: [DEBUG] Using task_output_dir: 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti' 2019-03-21 15:04:38,611 - root: [WARNING] task_output_dir 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti' already exists! 2019-03-21 15:04:38,611 - root: [WARNING] task_output_dir existing content: [] 2019-03-21 15:04:38,611 - root: [INFO] collect_cmd: swarming collect -server https://chromium-swarm.appspot.com -worker 50 -task-summary-python -task-output-stdout all -verbose -requests-json c:\b\s\w\ir\tmp\t\tmprmvyvk.json -output-dir c:\b\s\w\ir\tmp\t\tmpyl4hti -task-summary-json c:\b\s\w\ir\tmp\t\tmpqa8tdp.json [D2019-03-21T15:04:38.626005-07:00 6764 0 auth.go:1265] Minting a new token {"key":"luci_ctx/593f414acf64edc38681756aa4e0202cad82491b83d18c5e777109218cd3ade0", "scopes":"https://www.googleapis.com/auth/userinfo.email"} [D2019-03-21T15:04:38.626005-07:00 6764 0 luci_ctx.go:138] POST http://127.0.0.1:50332/rpc/LuciLocalAuthService.GetOAuthToken {"key":"luci_ctx/593f414acf64edc38681756aa4e0202cad82491b83d18c5e777109218cd3ade0", "scopes":"https://www.googleapis.com/auth/userinfo.email"} [D2019-03-21T15:04:38.628151-07:00 6764 0 auth.go:1222] Token expires in 30m26.371849s {"key":"luci_ctx/593f414acf64edc38681756aa4e0202cad82491b83d18c5e777109218cd3ade0", "scopes":"https://www.googleapis.com/auth/userinfo.email"} 43bb18f2d5a80d10: exit 0 Additional test environment: CHROME_DEVEL_SANDBOX=/opt/chromium/chrome_sandbox CHROME_HEADLESS=1 LANG=en_US.UTF-8 Command: e:\b\s\w\ir\.swarming_module_cache\vpython\37d31f\Scripts\python.exe ..\..\tools_webrtc\gtest-parallel-wrapper.py --output_dir=e:\b\s\w\iom9yiye\test_logs --gtest_color=no --timeout=900 --retry_failed=3 .\peerconnection_unittests.exe --isolated-script-test-output=e:\b\s\w\iom9yiye\output.json --isolated-script-test-perf-output=e:\b\s\w\iom9yiye\perftest-output.json gtest-parallel-wrapper: Executing command e:\b\s\w\ir\.swarming_module_cache\vpython\37d31f\Scripts\python.exe e:\b\s\w\ir\third_party\gtest-parallel\gtest-parallel --shard_index=0 --shard_count=4 --output_dir=e:\b\s\w\iom9yiye\test_logs --retry_failed=3 --gtest_color=no --timeout=900 --dump_json_test_results=e:\b\s\w\iom9yiye\output.json .\peerconnection_unittests.exe -- --isolated-script-test-perf-output=e:\b\s\w\iom9yiye\perftest-output.json [0/452] Running tests... [1/452] PeerConnectionIceTest/PeerConnectionIceTest.LocalCandidatesNotRemovedWhenNetworkDownIfGatheringOnce/1 (1424 ms) [2/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.DisableAndEnableAudioRecording/1 (1481 ms) [3/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.StressTestUnorderedSctpDataChannel/1 (1685 ms) [4/452] PeerConnectionIceTest/PeerConnectionIceTest.NoIceCandidatesBeforeAnswerSetAsLocalDescription/1 (1346 ms) [5/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges/1 (1418 ms) [6/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.DisableAndEnableAudioPlayout/1 (1707 ms) [7/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaContinuesFlowingAfterIceRestart/1 (1440 ms) [8/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.CallTransferredForCaller/1 (4344 ms) [9/452] RTCStatsIntegrationTest.GetStatsFromCallee (1297 ms) [10/452] PeerConnectionIntegrationTestPlanB.MediaFlowsAfterEarlyWarmupWithCreateSender (1713 ms) [11/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.NewGetStatsManyAudioAndManyVideoStreams/1 (1476 ms) [12/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithDtls/1 (1557 ms) [13/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationInteropTest.OneAudioOneVideoLocalToOneAudioOneVideoRemote/1 (1177 ms) [14/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.SctpDataChannelToAudioVideoUpgrade/1 (1894 ms) [15/452] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.CallWithCustomCodec/1 (1211 ms) [16/452] RTCStatsIntegrationTest.GetStatsWithInvalidReceiverSelector (1042 ms) [17/452] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.Call/1 (1138 ms) [18/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.OneWayMediaCall/1 (963 ms) [19/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithSpecCompliantMaxBundleOffer/1 (1306 ms) [20/452] PeerConnectionIntegrationTestUnifiedPlan.OldGetStatsAssociatesTrackIdForManyMediaSections (2371 ms) [21/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithRtpDataChannel/1 (1070 ms) [22/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetBytesSentStatsWithOldStatsApi/1 (1108 ms) [23/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.CallerDtls12ToCalleeDtls10/1 (1138 ms) [24/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AnswererRejectsAudioSection/1 (1354 ms) [25/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AddAudioToVideoOnlyCall/1 (1719 ms) [26/452] RtpSenderReceiverTest.InsertDtmf (663 ms) [27/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.SSLCertificateVerifierFailureUsedForTurnConnectionsFailsConnection/1 (10368 ms) [28/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithoutSsrcOrMsidSignaling/1 (1064 ms) [29/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.TCPUsedForTurnConnections/1 (950 ms) [30/452] PeerConnectionIceTest/PeerConnectionIceTest.VerifyUfragPwdLength/1 (667 ms) [31/452] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate/1 (756 ms) [32/452] PeerConnectionJsepTest/RecycleMediaSectionTest.CurrentLocalAndCurrentRemoteRejected/3 (615 ms) [33/452] PeerConnectionJsepTest/RecycleMediaSectionTest.CurrentRemoteOnlyRejected/3 (507 ms) [34/452] PeerConnectionRtpTest/PeerConnectionRtpTest.RemoveTrackWithSharedStreamRemovesReceiver/1 (644 ms) [35/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.RotatedVideoWithCVOExtension/1 (657 ms) [36/452] PeerConnectionRtpTestUnifiedPlan.SetDirectionHoldCallsOnTrackTwice (493 ms) [37/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndConnectionTimeWithTurnTurnPair/1 (2534 ms) [38/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetAudioInputLevelStatsWithOldStatsApi/1 (734 ms) [39/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/19 (792 ms) [40/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/15 (636 ms) [41/452] PeerConnectionIceTest/PeerConnectionIceTest.AnswerContainsGatheredCandidates/1 (671 ms) [42/452] PeerConnectionIceTest/PeerConnectionIceUfragPwdAnswerTest.TestIncludedInAnswer/3 (607 ms) [43/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/19 (531 ms) [44/452] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.CloseDataChannelRemotelyWhileNotReferenced/1 (562 ms) [45/452] PeerConnectionRtpTest/PeerConnectionRtpTest.RemoveTrackWithoutStreamRemovesReceiver/1 (467 ms) [46/452] PeerConnectionIceTest/PeerConnectionIceTest.IceRestartOfferCandidateReplacesExistingCandidate/1 (493 ms) [47/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.TransportStatsReportedForDataChannelOnlyConnection/1 (733 ms) [48/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.RtpReceiverObserverOnFirstPacketReceived/1 (883 ms) [49/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/19 (402 ms) [50/452] PeerConnectionIntegrationTestUnifiedPlan.EndToEndCallWithoutSsrcSignaling (586 ms) [51/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/15 (716 ms) [52/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/7 (393 ms) [53/452] PeerConnectionRtpTestUnifiedPlan.ChangeMsidWhileReceiving (398 ms) [54/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/11 (444 ms) [55/452] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingLocalRejectedAndNoRemote/3 (498 ms) [56/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/15 (376 ms) [57/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/19 (618 ms) [58/452] PeerConnectionSignalingUnifiedPlanTest.DtlsTransportsMergeWhenBundled (434 ms) [59/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.PooledSessionsDiscardedAfterClose/1 (278 ms) [60/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/15 (722 ms) [61/452] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.CreateDataChannelWithDtlsDisabledSucceeds/1 (227 ms) [62/452] PeerConnectionBundleTest/PeerConnectionBundleTest.OneCandidateForEachTransportWhenNoBundleButRtcpMux/1 (546 ms) [63/452] WebRtcSdpTest.DeserializeSdpWithMultiSctpPort (305 ms) [64/452] PeerConnectionUsageHistogramTest.NotableUsageOnEventFiring (368 ms) [65/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AddRtpDataChannelInSubsequentOffer/1 (1210 ms) [66/452] PeerConnectionBundleTest/PeerConnectionBundleTest.ApplyDescriptionWithConflictedDemuxCriteriaFail/1 (487 ms) [67/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.Dtls10CipherStatsAndUmaMetrics/1 (1256 ms) [68/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/15 (445 ms) [69/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/19 (569 ms) [70/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.CalleeClosesSctpDataChannel/1 (904 ms) [71/452] PeerConnectionRtpTest/PeerConnectionRtpTest.VideoGetParametersHasHeaderExtensions/1 (398 ms) [72/452] PeerConnectionIceTest/PeerConnectionIceTest.CannotRemoveIceCandidatesWhenPeerConnectionClosed/1 (575 ms) [73/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/3 (449 ms) [74/452] PeerConnectionMediaTest/PeerConnectionMediaTest.MediaTransportNotPropagatedToVoiceEngine/1 (399 ms) [75/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/11 (365 ms) [76/452] WebRtcSdpTest.SerializeSessionDescriptionWithFingerprint (208 ms) [77/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportBidirectionalVideo/1 (10627 ms) Error: unrecognized flag --isolated-script-test-perf-output=e:\b\s\w\iom9yiye\perftest-output.json Note: Google Test filter = PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportBidirectionalVideo/1 [==========] Running 1 test from 1 test suite. [----------] Global test environment set-up. [----------] 1 test from PeerConnectionIntegrationTest/PeerConnectionIntegrationTest [ RUN ] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportBidirectionalVideo/1 [000:000] (loopback_media_transport.cc:156): LoopbackMediaTransport [000:000] (loopback_media_transport.cc:156): LoopbackMediaTransport [000:001] (audio_processing_impl.cc:421): Capture analyzer activated: 0 Capture post processor activated: 0 Render pre processor activated: 0 [000:002] (webrtc_video_engine.cc:460): WebRtcVideoEngine::WebRtcVideoEngine() [000:002] (webrtc_voice_engine.cc:194): WebRtcVoiceEngine::WebRtcVoiceEngine [000:003] (webrtc_voice_engine.cc:217): WebRtcVoiceEngine::Init [000:003] (webrtc_voice_engine.cc:224): Supported send codecs in order of preference: [000:004] (webrtc_voice_engine.cc:227): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:004] (webrtc_voice_engine.cc:227): ISAC/16000/1 (103) [000:004] (webrtc_voice_engine.cc:227): ISAC/32000/1 (104) [000:004] (webrtc_voice_engine.cc:227): G722/8000/1 (9) [000:004] (webrtc_voice_engine.cc:227): ILBC/8000/1 (102) [000:004] (webrtc_voice_engine.cc:227): PCMU/8000/1 (0) [000:005] (webrtc_voice_engine.cc:227): PCMA/8000/1 (8) [000:005] (webrtc_voice_engine.cc:227): CN/32000/1 (106) [000:005] (webrtc_voice_engine.cc:227): CN/16000/1 (105) [000:005] (webrtc_voice_engine.cc:227): CN/8000/1 (13) [000:005] (webrtc_voice_engine.cc:227): telephone-event/48000/1 (110) [000:005] (webrtc_voice_engine.cc:227): telephone-event/32000/1 (112) [000:005] (webrtc_voice_engine.cc:227): telephone-event/16000/1 (113) [000:005] (webrtc_voice_engine.cc:227): telephone-event/8000/1 (126) [000:005] (webrtc_voice_engine.cc:230): Supported recv codecs in order of preference: [000:006] (webrtc_voice_engine.cc:233): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:006] (webrtc_voice_engine.cc:233): ISAC/16000/1 (103) [000:006] (webrtc_voice_engine.cc:233): ISAC/32000/1 (104) [000:006] (webrtc_voice_engine.cc:233): G722/8000/1 (9) [000:006] (webrtc_voice_engine.cc:233): ILBC/8000/1 (102) [000:006] (webrtc_voice_engine.cc:233): PCMU/8000/1 (0) [000:006] (webrtc_voice_engine.cc:233): PCMA/8000/1 (8) [000:006] (webrtc_voice_engine.cc:233): CN/32000/1 (106) [000:006] (webrtc_voice_engine.cc:233): CN/16000/1 (105) [000:006] (webrtc_voice_engine.cc:233): CN/8000/1 (13) [000:006] (webrtc_voice_engine.cc:233): telephone-event/48000/1 (110) [000:006] (webrtc_voice_engine.cc:233): telephone-event/32000/1 (112) [000:006] (webrtc_voice_engine.cc:233): telephone-event/16000/1 (113) [000:007] (webrtc_voice_engine.cc:233): telephone-event/8000/1 (126) [000:008] (webrtc_voice_engine.cc:310): 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, } [000:009] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [000:009] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [000:010] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [000:011] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [000:011] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [000:011] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [000:015] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:017] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:017] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:017] (apm_helpers.cc:90): Echo control set to 1 with mode 0 [000:021] (apm_helpers.cc:79): AGC set to 1 with mode 0 [000:021] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:021] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:021] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:021] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:022] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:022] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:022] (apm_helpers.cc:104): NS set to 1 [000:022] (webrtc_voice_engine.cc:466): Stereo swapping enabled? 0 [000:022] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:022] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:022] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:022] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:022] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:022] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:022] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:022] (webrtc_voice_engine.cc:536): Typing detection is enabled? 1 [000:022] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:022] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:022] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:023] (rtc_event_log_impl.cc:72): Creating legacy encoder for RTC event log. [000:023] (peer_connection_factory.cc:395): Using default network controller factory [000:025] (bitrate_prober.cc:62): Bandwidth probing enabled, set to inactive [000:025] (paced_sender.cc:427): ProcessThreadAttached 0x2088a60 [000:025] (cpu_info.cc:49): Available number of cores: 8 [000:025] (aimd_rate_control.cc:101): Using aimd rate control with back off factor 0.85 [000:025] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating. [000:029] (audio_processing_impl.cc:421): Capture analyzer activated: 0 Capture post processor activated: 0 Render pre processor activated: 0 [000:030] (webrtc_video_engine.cc:460): WebRtcVideoEngine::WebRtcVideoEngine() [000:030] (webrtc_voice_engine.cc:194): WebRtcVoiceEngine::WebRtcVoiceEngine [000:031] (webrtc_voice_engine.cc:217): WebRtcVoiceEngine::Init [000:031] (webrtc_voice_engine.cc:224): Supported send codecs in order of preference: [000:033] (webrtc_voice_engine.cc:227): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:033] (webrtc_voice_engine.cc:227): ISAC/16000/1 (103) [000:033] (webrtc_voice_engine.cc:227): ISAC/32000/1 (104) [000:033] (webrtc_voice_engine.cc:227): G722/8000/1 (9) [000:033] (webrtc_voice_engine.cc:227): ILBC/8000/1 (102) [000:033] (webrtc_voice_engine.cc:227): PCMU/8000/1 (0) [000:033] (webrtc_voice_engine.cc:227): PCMA/8000/1 (8) [000:033] (webrtc_voice_engine.cc:227): CN/32000/1 (106) [000:033] (webrtc_voice_engine.cc:227): CN/16000/1 (105) [000:033] (webrtc_voice_engine.cc:227): CN/8000/1 (13) [000:033] (webrtc_voice_engine.cc:227): telephone-event/48000/1 (110) [000:033] (webrtc_voice_engine.cc:227): telephone-event/32000/1 (112) [000:033] (webrtc_voice_engine.cc:227): telephone-event/16000/1 (113) [000:033] (webrtc_voice_engine.cc:227): telephone-event/8000/1 (126) [000:033] (webrtc_voice_engine.cc:230): Supported recv codecs in order of preference: [000:035] (webrtc_voice_engine.cc:233): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:035] (webrtc_voice_engine.cc:233): ISAC/16000/1 (103) [000:035] (webrtc_voice_engine.cc:233): ISAC/32000/1 (104) [000:035] (webrtc_voice_engine.cc:233): G722/8000/1 (9) [000:035] (webrtc_voice_engine.cc:233): ILBC/8000/1 (102) [000:035] (webrtc_voice_engine.cc:233): PCMU/8000/1 (0) [000:035] (webrtc_voice_engine.cc:233): PCMA/8000/1 (8) [000:035] (webrtc_voice_engine.cc:233): CN/32000/1 (106) [000:035] (webrtc_voice_engine.cc:233): CN/16000/1 (105) [000:035] (webrtc_voice_engine.cc:233): CN/8000/1 (13) [000:035] (webrtc_voice_engine.cc:233): telephone-event/48000/1 (110) [000:035] (webrtc_voice_engine.cc:233): telephone-event/32000/1 (112) [000:035] (webrtc_voice_engine.cc:233): telephone-event/16000/1 (113) [000:035] (webrtc_voice_engine.cc:233): telephone-event/8000/1 (126) [000:036] (webrtc_voice_engine.cc:310): 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, } [000:037] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [000:038] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [000:038] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [000:038] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [000:038] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [000:038] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [000:039] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:039] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:039] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:039] (apm_helpers.cc:90): Echo control set to 1 with mode 0 [000:039] (apm_helpers.cc:79): AGC set to 1 with mode 0 [000:039] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:040] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:040] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:040] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:040] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:040] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:040] (apm_helpers.cc:104): NS set to 1 [000:040] (webrtc_voice_engine.cc:466): Stereo swapping enabled? 0 [000:040] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:041] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:041] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:041] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:041] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:041] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:041] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:041] (webrtc_voice_engine.cc:536): Typing detection is enabled? 1 [000:041] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:041] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:041] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:043] (rtc_event_log_impl.cc:72): Creating legacy encoder for RTC event log. [000:043] (peer_connection_factory.cc:395): Using default network controller factory [000:043] (bitrate_prober.cc:62): Bandwidth probing enabled, set to inactive [000:044] (paced_sender.cc:427): ProcessThreadAttached 0x21090b0 [000:044] (aimd_rate_control.cc:101): Using aimd rate control with back off factor 0.85 [000:044] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating. [000:047] (peer_connection.cc:1381): Adding video transceiver in response to a call to AddTrack. [000:048] (peer_connection.cc:1381): Adding video transceiver in response to a call to AddTrack. [000:049] (jsep_transport_controller.cc:1518): Generating media transport offer! [000:049] (media_session.cc:335): Duplicate id found. Reassigning from 102 to 127 [000:049] (media_session.cc:335): Duplicate id found. Reassigning from 104 to 125 [000:050] (media_session.cc:335): Duplicate id found. Reassigning from 103 to 124 [000:050] (media_session.cc:335): Duplicate id found. Reassigning from 1 to 14 [000:050] (media_session.cc:335): Duplicate id found. Reassigning from 2 to 13 [000:050] (media_session.cc:335): Duplicate id found. Reassigning from 3 to 12 [000:050] (media_session.cc:335): Duplicate id found. Reassigning from 5 to 11 [000:052] (peer_connection_integrationtest.cc:753): Caller: SetLocalDescriptionAndSendSdpMessage [000:053] (jsep_transport_controller.cc:975): Offered media transport has now been activated. [000:053] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [000:053] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [000:053] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [000:053] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [000:053] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [000:053] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [000:054] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [000:054] (p2p_transport_channel.cc:432): Set ICE ufrag: utsH pwd: ZgXn9pLIQFEdoug2CAsDVjz8 on transport 0 [000:054] (webrtc_video_engine.cc:472): CreateMediaChannel. Options: VideoOptions {} [000:055] (channel.cc:144): Created channel for 0 [000:058] (channel.cc:212): BaseChannel::Init_w, media_transport=1 [000:058] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:058] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:058] (peer_connection.cc:4097): Session: 1372623304754600989 Old state: kStable New state: kHaveLocalOffer [000:059] (channel.cc:1009): Setting local video description [000:059] (webrtc_video_engine.cc:945): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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}]} [000:060] (webrtc_video_engine.cc:960): Changing recv codecs from {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} to {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} [000:061] (webrtc_video_engine.cc:1061): AddSendStream: {id:e992882a-daeb-4b03-958f-3d8a26127e33;ssrcs:[355574507,2742550061];ssrc_groups:{semantics:FID;ssrcs:[355574507,2742550061]};cname:+Z459UzLBTKeyDmf;stream_ids:;} [000:061] (webrtc_video_engine.cc:1100): SetLocalSsrc on all the receive streams because we added a send stream. [000:061] (channel.cc:676): Add send stream ssrc: 355574507 [000:061] (channel.cc:996): Changing video state, send=0 [000:062] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 355574507, options: VideoOptions {is_screencast : false, }, source = (source)) [000:063] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [000:063] (basic_port_allocator.cc:851): Network manager has started [000:063] (basic_port_allocator.cc:766): Allocate ports on 1 networks [000:063] (peer_connection_integrationtest.cc:683): Callee: HandleIncomingOffer [000:063] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Udp [000:063] (port.cc:318): Port[2f065a0::1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port created with network cost 50 [000:063] (basic_port_allocator.cc:875): Adding allocated port for 0 [000:063] (basic_port_allocator.cc:894): Port[2f065a0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Added port to allocator [000:064] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:064] (basic_port_allocator.cc:912): Port[2f065a0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Gathered candidate: Cand[:2214029314:1:udp:2122260223:192.168.1.1:49152:local::0:utsH:ZgXn9pLIQFEdoug2CAsDVjz8:1:50:0] [000:064] (basic_port_allocator.cc:940): Port[2f065a0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port ready. [000:064] (basic_port_allocator.cc:1017): Port[2f065a0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port completed gathering candidates. [000:067] (peer_connection_integrationtest.cc:771): Callee: SetRemoteDescription [000:068] (jsep_transport_controller.cc:996): Returning new, client media transport. [000:068] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [000:068] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [000:068] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [000:068] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [000:068] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [000:068] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [000:069] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [000:069] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=utsH, renomination disabled [000:069] (webrtc_video_engine.cc:472): CreateMediaChannel. Options: VideoOptions {} [000:070] (channel.cc:144): Created channel for 0 [000:070] (channel.cc:212): BaseChannel::Init_w, media_transport=1 [000:071] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:071] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:071] (peer_connection.cc:4097): Session: 1908916124702233608 Old state: kStable New state: kHaveRemoteOffer [000:071] (channel.cc:1061): Setting remote video description [000:071] (webrtc_video_engine.cc:670): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 0} [000:072] (webrtc_video_engine.cc:679): Using codec: VideoCodec[96:VP8] [000:072] (webrtc_video_engine.cc:747): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. [000:072] (webrtc_video_engine.cc:1163): AddRecvStream: {id:e992882a-daeb-4b03-958f-3d8a26127e33;ssrcs:[355574507,2742550061];ssrc_groups:{semantics:FID;ssrcs:[355574507,2742550061]};cname:+Z459UzLBTKeyDmf;stream_ids:;} [000:074] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 355574507, 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: 104, red_type: 102, rtx_ssrc: 2742550061, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 103 (pt) -> 102 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, target_delay_ms: 0} [000:074] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:074] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:074] (bitrate_allocator.cc:115): Current BWE 300000 [000:076] (channel.cc:719): Add remote ssrc: 355574507 [000:078] (channel.cc:996): Changing video state, send=0 [000:078] (peer_connection.cc:2687): Processing the MSIDs for MID=0 (streams=[]). [000:078] (peer_connection.cc:2698): Processing the addition of a remote track for MID=0. [000:078] (webrtc_video_engine.cc:1293): SetSink: ssrc:355574507 (ptr) [000:079] (peer_connection_integrationtest.cc:919): Caller: OnIceCandidate [000:079] (peer_connection_integrationtest.cc:867): Callee: ReceiveIceMessage [000:079] (peer_connection.cc:6141): 0 is not ready to use the remote candidate because the local or remote description is not set. [000:081] (peer_connection.cc:5449): Local and Remote descriptions must be applied to get the SSL Role of the session. [000:085] (peer_connection_integrationtest.cc:753): Callee: SetLocalDescriptionAndSendSdpMessage [000:094] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [000:094] (p2p_transport_channel.cc:432): Set ICE ufrag: FoCF pwd: 9S6vGzfJBbQBRnVzd8GRGLNT on transport 0 [000:095] (dtls_transport.cc:253): DtlsTransport[0|1|__]: Other side didn't support DTLS. [000:095] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv. [000:096] (channel.cc:553): Channel enabled [000:100] (channel.cc:996): Changing video state, send=0 [000:100] (peer_connection.cc:4097): Session: 1908916124702233608 Old state: kHaveRemoteOffer New state: kStable [000:100] (channel.cc:1009): Setting local video description [000:100] (webrtc_video_engine.cc:945): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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}]} [000:102] (webrtc_video_engine.cc:960): Changing recv codecs from {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} to {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} [000:102] (webrtc_video_engine.cc:2493): MaybeRecreateWebRtcFlexfecStream (recv) because of SetRecvParameters [000:102] (webrtc_video_engine.cc:2498): RecreateWebRtcVideoStream (recv) because of SetRecvParameters [000:102] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:102] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:102] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 355574507, 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: 104, red_type: 102, rtx_ssrc: 2742550061, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 103 (pt) -> 102 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, target_delay_ms: 0} [000:102] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [000:103] (receive_statistics_proxy.cc:487): Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [000:105] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 355574507, 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: 125, red_type: 127, rtx_ssrc: 2742550061, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [000:105] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:105] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:106] (webrtc_video_engine.cc:1061): AddSendStream: {id:c78dbf91-bfa3-4536-934f-82d436845126;ssrcs:[3813599382,3652285399];ssrc_groups:{semantics:FID;ssrcs:[3813599382,3652285399]};cname:EK4BB4b9lRgWu1rc;stream_ids:;} [000:106] (webrtc_video_engine.cc:1859): RecreateWebRtcStream (send) because of SetCodec. [000:107] (alr_experiment.cc:66): 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 [000:110] (video_send_stream_impl.cc:249): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [3813599382], 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: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [3652285399], payload_type: 97}, c_name: EK4BB4b9lRgWu1rc}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [000:110] (video_send_stream_impl.cc:275): ERROR: Initial encoder max bitrate = -1 which is <= 0! [000:110] (video_send_stream_impl.cc:277): Using default encoder max bitrate = 10 Mbps [000:110] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:110] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:110] (video_send_stream.cc:160): VideoSendStream::Stop [000:110] (webrtc_video_engine.cc:1100): SetLocalSsrc on all the receive streams because we added a send stream. [000:110] (webrtc_video_engine.cc:2434): RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc=3813599382 [000:110] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:110] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:110] (video_stream_encoder.cc:579): ConfigureEncoder requested. [000:110] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 355574507, 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: 125, red_type: 127, rtx_ssrc: 2742550061, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [000:111] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [000:111] (video_send_stream_impl.cc:415): VideoSendStream::Stop [000:112] (receive_statistics_proxy.cc:487): Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [000:113] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 355574507, local_ssrc: 3813599382, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 2742550061, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [000:113] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:114] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:114] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Relay [000:115] (channel.cc:676): Add send stream ssrc: 3813599382 [000:116] (video_send_stream.cc:160): VideoSendStream::Stop [000:116] (channel.cc:996): Changing video state, send=0 [000:118] (video_send_stream_impl.cc:415): VideoSendStream::Stop [000:119] (peer_connection.cc:4033): Changing IceConnectionState 0 => 1 [000:119] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 3813599382, options: VideoOptions {is_screencast : false, }, source = (source)) [000:120] (video_stream_encoder.cc:579): ConfigureEncoder requested. [000:121] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [000:121] (basic_port_allocator.cc:851): Network manager has started [000:121] (basic_port_allocator.cc:766): Allocate ports on 1 networks [000:121] (peer_connection_integrationtest.cc:703): Caller: HandleIncomingAnswer [000:121] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Udp [000:121] (port.cc:318): Port[2f37a90::1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port created with network cost 50 [000:121] (basic_port_allocator.cc:875): Adding allocated port for 0 [000:121] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:121] (basic_port_allocator.cc:894): Port[2f37a90:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Added port to allocator [000:122] (basic_port_allocator.cc:912): Port[2f37a90:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Gathered candidate: Cand[:2214029314:1:udp:2122260223:192.168.1.1:49153:local::0:FoCF:9S6vGzfJBbQBRnVzd8GRGLNT:1:50:0] [000:122] (basic_port_allocator.cc:940): Port[2f37a90:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port ready. [000:122] (port.cc:1126): Conn[2fc57e0:0:Net[test0:192.168.1.0/24:Unknown:id=1]:+x32CLhv:1:0:local:udp:192.168.1.1:49153->fMN2yCYo:1:2122260223:local:udp:192.168.1.1:49152|C--W|-|0|0|9115038255631187454|-]: Connection created [000:122] (p2p_transport_channel.cc:1264): Channel[0|1|__]: Created connection with origin: 2, total: 1 [000:122] (p2p_transport_channel.cc:1914): Channel[0|1|__]: Transport channel state changed from 0 to 2 [000:122] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:122] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:122] (p2p_transport_channel.cc:1476): Channel[0|1|__]: Have a pingable connection for the first time; starting to ping. [000:123] (basic_port_allocator.cc:1017): Port[2f37a90:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port completed gathering candidates. [000:123] (port.cc:1756): Conn[2fc57e0:0:Net[test0:192.168.1.0/24:Unknown:id=1]:+x32CLhv:1:0:local:udp:192.168.1.1:49153->fMN2yCYo:1:2122260223:local:udp:192.168.1.1:49152|C--W|-|0|0|9115038255631187454|-]: Sent STUN ping, id=2b346a633668557a4c4d544c, use_candidate=0, nomination=0 [000:123] (port.cc:528): Received STUN ping id=2b346a633668557a4c4d544c from unknown address 192.168.1.1:49153 [000:124] (port.cc:1126): Conn[2fc8f40:0:Net[test0:192.168.1.0/24:Unknown:id=1]:okNDQ5Us:1:0:local:udp:192.168.1.1:49152->fWmEUmVj:1:1853824767:prflx:udp:192.168.1.1:49153|C--W|-|0|0|7962116751024340479|-]: Connection created [000:124] (p2p_transport_channel.cc:988): Adding connection from peer reflexive candidate: Cand[:1291304870:1:udp:1853824767:192.168.1.1:49153:prflx::0:FoCF::1:50:0] [000:124] (port.cc:853): Port[2f065a0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Sent STUN ping response, to=192.168.1.1:49153, id=2b346a633668557a4c4d544c [000:124] (p2p_transport_channel.cc:1914): Channel[0|1|R_]: Transport channel state changed from 0 to 2 [000:124] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:124] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:125] (port.cc:1704): Conn[2fc57e0:0:Net[test0:192.168.1.0/24:Unknown:id=1]:+x32CLhv:1:0:local:udp:192.168.1.1:49153->fMN2yCYo:1:2122260223:local:udp:192.168.1.1:49152|C--I|-|0|0|9115038255631187454|-]: Received STUN ping response, id=2b346a633668557a4c4d544c, code=0, rtt=2, pings_since_last_response=2b346a633668557a4c4d544c [000:125] (peer_connection_integrationtest.cc:771): Caller: SetRemoteDescription [000:125] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed [000:126] (p2p_transport_channel.cc:1852): Channel[0|1|__]: New selected connection: Conn[2fc57e0:0:Net[test0:192.168.1.0/24:Unknown:id=1]:+x32CLhv:1:0:local:udp:192.168.1.1:49153->fMN2yCYo:1:2122260223:local:udp:192.168.1.1:49152|CRWS|S|0|0|9115038255631187454|2] [000:126] (jsep_transport_controller.cc:1214): Transport 0 writability changed to 1. [000:126] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:126] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:126] (channel.cc:583): Channel writable (0) for the first time [000:126] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:126] (basic_port_allocator.cc:1119): All candidates gathered for 0:1:0 [000:126] (p2p_transport_channel.cc:854): P2PTransportChannel: 0, component 1 gathering complete [000:126] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:126] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:127] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps [000:127] (video_stream_encoder.cc:1477): Video suspend state changed to: not suspended [000:127] (channel.cc:996): Changing video state, send=1 [000:128] (webrtc_video_engine.cc:1338): Call stats: 2321582, {send_bw_bps: 0, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1} [000:129] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [000:130] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=FoCF, renomination disabled [000:130] (dtls_transport.cc:253): DtlsTransport[0|1|R_]: Other side didn't support DTLS. [000:130] (p2p_transport_channel.cc:1476): Channel[0|1|R_]: Have a pingable connection for the first time; starting to ping. [000:130] (p2p_transport_channel.cc:2440): Selecting connection for triggered check: Conn[2fc8f40:0:Net[test0:192.168.1.0/24:Unknown:id=1]:okNDQ5Us:1:0:local:udp:192.168.1.1:49152->fWmEUmVj:1:1853824767:prflx:udp:192.168.1.1:49153|CR-W|-|0|0|7962116751024340479|-] [000:130] (channel.cc:553): Channel enabled [000:130] (port.cc:1756): Conn[2fc8f40:0:Net[test0:192.168.1.0/24:Unknown:id=1]:okNDQ5Us:1:0:local:udp:192.168.1.1:49152->fWmEUmVj:1:1853824767:prflx:udp:192.168.1.1:49153|CR-W|-|0|0|7962116751024340479|-]: Sent STUN ping, id=513474457369725a61633046, use_candidate=1, nomination=0 [000:131] (channel.cc:996): Changing video state, send=0 [000:131] (peer_connection.cc:4097): Session: 1372623304754600989 Old state: kHaveLocalOffer New state: kStable [000:131] (channel.cc:1061): Setting remote video description [000:131] (port.cc:1704): Conn[2fc8f40:0:Net[test0:192.168.1.0/24:Unknown:id=1]:okNDQ5Us:1:0:local:udp:192.168.1.1:49152->fWmEUmVj:1:1853824767:prflx:udp:192.168.1.1:49153|CR-I|-|0|0|7962116751024340479|-]: Received STUN ping response, id=513474457369725a61633046, code=0, rtt=1, pings_since_last_response=513474457369725a61633046 [000:131] (basic_port_allocator.cc:1119): All candidates gathered for 0:1:0 [000:131] (p2p_transport_channel.cc:854): P2PTransportChannel: 0, component 1 gathering complete [000:132] (webrtc_video_engine.cc:670): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 0} [000:132] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed [000:132] (p2p_transport_channel.cc:1852): Channel[0|1|R_]: New selected connection: Conn[2fc8f40:0:Net[test0:192.168.1.0/24:Unknown:id=1]:okNDQ5Us:1:0:local:udp:192.168.1.1:49152->fWmEUmVj:1:1853824767:prflx:udp:192.168.1.1:49153|CRWS|S|0|0|7962116751024340479|1] [000:132] (jsep_transport_controller.cc:1214): Transport 0 writability changed to 1. [000:132] (channel.cc:583): Channel writable (0) for the first time [000:132] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:133] (webrtc_video_engine.cc:679): Using codec: VideoCodec[96:VP8] [000:134] (webrtc_video_engine.cc:1859): RecreateWebRtcStream (send) because of SetCodec. [000:134] (alr_experiment.cc:66): 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 [000:154] (video_stream_encoder.cc:1051): Video frame parameters changed: dimensions=640x480, texture=0. [000:154] (video_stream_encoder.cc:233): Set max framerate: 60 [000:164] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 30000bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 1700000bps [000:159] (bitrate_allocator.cc:115): Current BWE 300000 [000:187] (video_send_stream_impl.cc:249): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [355574507], 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: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [2742550061], payload_type: 97}, c_name: +Z459UzLBTKeyDmf}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [000:187] (video_send_stream_impl.cc:275): ERROR: Initial encoder max bitrate = -1 which is <= 0! [000:187] (video_send_stream_impl.cc:277): Using default encoder max bitrate = 10 Mbps [000:189] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:189] (video_stream_encoder.cc:579): ConfigureEncoder requested. [000:189] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:190] (video_send_stream.cc:160): VideoSendStream::Stop [000:190] (webrtc_video_engine.cc:747): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. [000:190] (webrtc_video_engine.cc:1163): AddRecvStream: {id:c78dbf91-bfa3-4536-934f-82d436845126;ssrcs:[3813599382,3652285399];ssrc_groups:{semantics:FID;ssrcs:[3813599382,3652285399]};cname:EK4BB4b9lRgWu1rc;stream_ids:;} [000:191] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 3813599382, local_ssrc: 355574507, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 3652285399, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [000:192] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:192] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:193] (channel.cc:719): Add remote ssrc: 3813599382 [000:193] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:194] (video_send_stream_impl.cc:415): VideoSendStream::Stop [000:194] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:195] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps [000:195] (channel.cc:996): Changing video state, send=1 [000:195] (message_queue.cc:518): Message took 64ms to dispatch. Posted from: cricket::BaseChannel::SetRemoteContent@../../pc/channel.cc:302 [000:195] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:195] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:195] (peer_connection.cc:4033): Changing IceConnectionState 0 => 1 [000:195] (peer_connection.cc:2687): Processing the MSIDs for MID=0 (streams=[]). [000:195] (peer_connection.cc:2698): Processing the addition of a remote track for MID=0. [000:195] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:195] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv. [000:195] (video_stream_encoder.cc:1477): Video suspend state changed to: not suspended [000:195] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:195] (channel.cc:996): Changing video state, send=1 [000:195] (webrtc_video_engine.cc:1293): SetSink: ssrc:3813599382 (ptr) [000:196] (peer_connection_integrationtest.cc:919): Callee: OnIceCandidate [000:197] (peer_connection_integrationtest.cc:867): Caller: ReceiveIceMessage [000:202] (peer_connection.cc:5981): Changing to ICE completed state because all transports are complete. [000:202] (peer_connection.cc:4033): Changing IceConnectionState 1 => 2 [000:202] (peer_connection.cc:4033): Changing IceConnectionState 2 => 3 [000:204] (peer_connection.cc:5975): Changing to ICE connected state because all transports are writable. [000:204] (peer_connection.cc:4033): Changing IceConnectionState 1 => 2 [000:255] (video_stream_encoder.cc:1051): Video frame parameters changed: dimensions=640x480, texture=0. [000:256] (video_stream_encoder.cc:233): Set max framerate: 60 [000:273] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 30000bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 1700000bps [000:294] (decoder_database.cc:140): Initializing decoder with payload type '96'. ../../pc/peer_connection_integrationtest.cc(1502): error: Value of: caller()->audio_frames_received() >= total_caller_audio_frames_expected && caller()->min_video_frames_received_per_track() >= total_caller_video_frames_expected && callee()->audio_frames_received() >= total_callee_audio_frames_expected && callee()->min_video_frames_received_per_track() >= total_callee_video_frames_expected Actual: false Expected: true Stack trace: 000000013FD36B10: testing::DefaultValue<rtc::AsyncResolverInterface * __ptr64>::Exists 000000013FD85D4D: webrtc::VP9EncoderImpl::PopulateCodecSpecific 000000014063625C: testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void> 00000001406360FB: testing::internal::HandleExceptionsInMethodIfSupported<testing::Test,void> 00000001406595DA: testing::Test::Run 0000000140659725: testing::TestInfo::Run ... Google Test internal frames ... ../../pc/peer_connection_integrationtest.cc(1518): error: Expected: (caller()->min_video_frames_received_per_track()) >= (total_caller_video_frames_expected), actual: 0 vs 3 Stack trace: 000000013FD36D66: testing::DefaultValue<rtc::AsyncResolverInterface * __ptr64>::Exists 000000013FD85D4D: webrtc::VP9EncoderImpl::PopulateCodecSpecific 000000014063625C: testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void> 00000001406360FB: testing::internal::HandleExceptionsInMethodIfSupported<testing::Test,void> 00000001406595DA: testing::Test::Run 0000000140659725: testing::TestInfo::Run ... Google Test internal frames ... ../../pc/peer_connection_integrationtest.cc(3718): error: Value of: ExpectNewFrames(media_expectations) Actual: false Expected: true Stack trace: 000000013FD85E02: webrtc::VP9EncoderImpl::PopulateCodecSpecific 000000014063625C: testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void> 00000001406360FB: testing::internal::HandleExceptionsInMethodIfSupported<testing::Test,void> 00000001406595DA: testing::Test::Run 0000000140659725: testing::TestInfo::Run ... Google Test internal frames ... [010:286] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 355574507, options: nullptr, source = nullptr) [010:286] (webrtc_video_engine.cc:1293): SetSink: ssrc:3813599382 nullptr [010:287] (channel.cc:563): Channel disabled [010:287] (video_send_stream.cc:160): VideoSendStream::Stop [010:287] (channel.cc:996): Changing video state, send=0 [010:287] (video_send_stream_impl.cc:415): VideoSendStream::Stop [010:287] (rtp_rtcp_impl.cc:359): Failed to send RTCP BYE [010:287] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps [010:287] (video_send_stream.cc:160): VideoSendStream::Stop [010:287] (video_send_stream_impl.cc:415): VideoSendStream::Stop [010:288] (video_stream_encoder.cc:1477): Video suspend state changed to: suspended [010:292] (video_send_stream_impl.cc:415): VideoSendStream::Stop [010:292] (video_send_stream_impl.cc:337): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [355574507], 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: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [2742550061], payload_type: 97}, c_name: +Z459UzLBTKeyDmf}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [010:293] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [010:293] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [010:293] (send_statistics_proxy.cc:654): Frames encoded 101 WebRTC.Video.DroppedFrames.Capturer 0 WebRTC.Video.DroppedFrames.EncoderQueue 0 WebRTC.Video.DroppedFrames.Encoder 0 WebRTC.Video.DroppedFrames.Ratelimiter 0 [010:293] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [010:294] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [010:294] (paced_sender.cc:115): PacedSender paused. [010:294] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 3813599382, local_ssrc: 355574507, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 3652285399, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [010:295] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [010:306] (receive_statistics_proxy.cc:487): WebRTC.Video.ReceiveStreamLifetimeInSeconds 10 Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [010:306] (channel.cc:162): Destroyed channel: 0 [010:306] (peer_connection.cc:901): Session: 1372623304754600989 is destroyed. [010:308] (paced_sender.cc:427): ProcessThreadAttached 0x0 [010:308] (paced_sender.cc:427): ProcessThreadAttached 0x0 [010:308] (rtc_event_log_impl.cc:212): Stopping WebRTC event log. [010:309] (rtc_event_log_impl.cc:229): WebRTC event log successfully stopped. [010:324] (webrtc_video_engine.cc:464): WebRtcVideoEngine::~WebRtcVideoEngine [010:324] (webrtc_voice_engine.cc:203): WebRtcVoiceEngine::~WebRtcVoiceEngine [010:333] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 3813599382, options: nullptr, source = nullptr) [010:333] (webrtc_video_engine.cc:1293): SetSink: ssrc:355574507 nullptr [010:334] (channel.cc:563): Channel disabled [010:335] (video_send_stream.cc:160): VideoSendStream::Stop [010:335] (channel.cc:996): Changing video state, send=0 [010:335] (video_send_stream_impl.cc:415): VideoSendStream::Stop [010:335] (rtp_rtcp_impl.cc:359): Failed to send RTCP BYE [010:335] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps [010:335] (video_send_stream.cc:160): VideoSendStream::Stop [010:335] (video_stream_encoder.cc:1477): Video suspend state changed to: suspended [010:336] (video_send_stream_impl.cc:415): VideoSendStream::Stop [010:340] (video_send_stream_impl.cc:415): VideoSendStream::Stop [010:340] (video_send_stream_impl.cc:337): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [3813599382], 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: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [3652285399], payload_type: 97}, c_name: EK4BB4b9lRgWu1rc}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [010:341] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [010:341] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [010:341] (send_statistics_proxy.cc:654): Frames encoded 102 WebRTC.Video.DroppedFrames.Capturer 0 WebRTC.Video.DroppedFrames.EncoderQueue 0 WebRTC.Video.DroppedFrames.Encoder 0 WebRTC.Video.DroppedFrames.Ratelimiter 0 [010:341] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [010:341] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [010:342] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 355574507, local_ssrc: 3813599382, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 2742550061, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [010:343] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [010:343] (paced_sender.cc:115): PacedSender paused. [010:344] (receive_statistics_proxy.cc:487): WebRTC.Video.ReceiveStreamLifetimeInSeconds 10 Frames decoded 101 WebRTC.Video.DroppedFrames.Receiver -101 WebRTC.Video.InterframeDelay95PercentileInMs 113 WebRTC.Video.InterframeDelay95PercentileInMs.S0 113 [010:344] (video_quality_observer.cc:135): WebRTC.Video.MeanTimeBetweenFreezesMs 9965 WebRTC.Video.TimeInHdPercentage 0 WebRTC.Video.TimeInBlockyVideoPercentage 0 WebRTC.Video.NumberResolutionDownswitchesPerMinute 0 WebRTC.Video.NumberFreezesPerMinute 0 WebRTC.Video.HarmonicFrameRate 10 [010:344] (channel.cc:162): Destroyed channel: 0 [010:344] (peer_connection.cc:901): Session: 1908916124702233608 is destroyed. [010:346] (paced_sender.cc:427): ProcessThreadAttached 0x0 [010:346] (paced_sender.cc:427): ProcessThreadAttached 0x0 [010:346] (rtc_event_log_impl.cc:212): Stopping WebRTC event log. [010:346] (rtc_event_log_impl.cc:229): WebRTC event log successfully stopped. [010:352] (webrtc_video_engine.cc:464): WebRtcVideoEngine::~WebRtcVideoEngine [010:352] (webrtc_voice_engine.cc:203): WebRtcVoiceEngine::~WebRtcVoiceEngine [010:356] (loopback_media_transport.cc:160): ~LoopbackMediaTransport [010:356] (loopback_media_transport.cc:160): ~LoopbackMediaTransport [ FAILED ] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportBidirectionalVideo/1, where GetParam() = 4-byte object <01-00 00-00> (10360 ms) [----------] 1 test from PeerConnectionIntegrationTest/PeerConnectionIntegrationTest (10360 ms total) [----------] Global test environment tear-down [==========] 1 test from 1 test suite ran. (10360 ms total) [ PASSED ] 0 tests. [ FAILED ] 1 test, listed below: [ FAILED ] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportBidirectionalVideo/1, where GetParam() = 4-byte object <01-00 00-00> 1 FAILED TEST [77/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportBidirectionalVideo/1 returned/aborted with exit code 1 (10627 ms) [78/452] WebRtcSdpTest.DeserializeSdpWithRtpDataChannels (211 ms) [79/452] PeerConnectionMediaTestUnifiedPlan.StoppedRemoteTransceiversRemovesMediaChannels (447 ms) [80/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetRemoteAnswerWithNoFingerprintWhenDtlsOn/1 (420 ms) [81/452] PeerConnectionSimulcastMetricsTests.IncomingSimulcastIsLogged (444 ms) [82/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/1 (11445 ms) Error: unrecognized flag --isolated-script-test-perf-output=e:\b\s\w\iom9yiye\perftest-output.json Note: Google Test filter = PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/1 [==========] Running 1 test from 1 test suite. [----------] Global test environment set-up. [----------] 1 test from PeerConnectionIntegrationTest/PeerConnectionIntegrationTest [ RUN ] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/1 [000:000] (loopback_media_transport.cc:156): LoopbackMediaTransport [000:000] (loopback_media_transport.cc:156): LoopbackMediaTransport [000:002] (audio_processing_impl.cc:421): Capture analyzer activated: 0 Capture post processor activated: 0 Render pre processor activated: 0 [000:003] (webrtc_video_engine.cc:460): WebRtcVideoEngine::WebRtcVideoEngine() [000:003] (webrtc_voice_engine.cc:194): WebRtcVoiceEngine::WebRtcVoiceEngine [000:006] (webrtc_voice_engine.cc:217): WebRtcVoiceEngine::Init [000:007] (webrtc_voice_engine.cc:224): Supported send codecs in order of preference: [000:009] (webrtc_voice_engine.cc:227): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:009] (webrtc_voice_engine.cc:227): ISAC/16000/1 (103) [000:009] (webrtc_voice_engine.cc:227): ISAC/32000/1 (104) [000:009] (webrtc_voice_engine.cc:227): G722/8000/1 (9) [000:009] (webrtc_voice_engine.cc:227): ILBC/8000/1 (102) [000:009] (webrtc_voice_engine.cc:227): PCMU/8000/1 (0) [000:009] (webrtc_voice_engine.cc:227): PCMA/8000/1 (8) [000:010] (webrtc_voice_engine.cc:227): CN/32000/1 (106) [000:010] (webrtc_voice_engine.cc:227): CN/16000/1 (105) [000:010] (webrtc_voice_engine.cc:227): CN/8000/1 (13) [000:010] (webrtc_voice_engine.cc:227): telephone-event/48000/1 (110) [000:010] (webrtc_voice_engine.cc:227): telephone-event/32000/1 (112) [000:010] (webrtc_voice_engine.cc:227): telephone-event/16000/1 (113) [000:010] (webrtc_voice_engine.cc:227): telephone-event/8000/1 (126) [000:010] (webrtc_voice_engine.cc:230): Supported recv codecs in order of preference: [000:012] (webrtc_voice_engine.cc:233): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:012] (webrtc_voice_engine.cc:233): ISAC/16000/1 (103) [000:012] (webrtc_voice_engine.cc:233): ISAC/32000/1 (104) [000:012] (webrtc_voice_engine.cc:233): G722/8000/1 (9) [000:012] (webrtc_voice_engine.cc:233): ILBC/8000/1 (102) [000:012] (webrtc_voice_engine.cc:233): PCMU/8000/1 (0) [000:012] (webrtc_voice_engine.cc:233): PCMA/8000/1 (8) [000:012] (webrtc_voice_engine.cc:233): CN/32000/1 (106) [000:012] (webrtc_voice_engine.cc:233): CN/16000/1 (105) [000:012] (webrtc_voice_engine.cc:233): CN/8000/1 (13) [000:013] (webrtc_voice_engine.cc:233): telephone-event/48000/1 (110) [000:013] (webrtc_voice_engine.cc:233): telephone-event/32000/1 (112) [000:013] (webrtc_voice_engine.cc:233): telephone-event/16000/1 (113) [000:013] (webrtc_voice_engine.cc:233): telephone-event/8000/1 (126) [000:013] (webrtc_voice_engine.cc:310): 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, } [000:016] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [000:016] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [000:016] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [000:016] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [000:016] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [000:017] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [000:018] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:018] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:018] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:018] (apm_helpers.cc:90): Echo control set to 1 with mode 0 [000:018] (apm_helpers.cc:79): AGC set to 1 with mode 0 [000:018] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:019] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:019] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:019] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:019] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:019] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:019] (apm_helpers.cc:104): NS set to 1 [000:019] (webrtc_voice_engine.cc:466): Stereo swapping enabled? 0 [000:019] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:019] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:020] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:020] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:020] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:020] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:020] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:020] (webrtc_voice_engine.cc:536): Typing detection is enabled? 1 [000:020] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:020] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:020] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:025] (rtc_event_log_impl.cc:72): Creating legacy encoder for RTC event log. [000:025] (peer_connection_factory.cc:395): Using default network controller factory [000:067] (bitrate_prober.cc:62): Bandwidth probing enabled, set to inactive [000:075] (paced_sender.cc:427): ProcessThreadAttached 0x1f69dd0 [000:075] (cpu_info.cc:49): Available number of cores: 8 [000:075] (aimd_rate_control.cc:101): Using aimd rate control with back off factor 0.85 [000:076] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating. [000:076] (message_queue.cc:518): Message took 51ms to dispatch. Posted from: webrtc::PeerConnectionFactory::CreatePeerConnection@../../pc/peer_connection_factory.cc:309 [000:084] (audio_processing_impl.cc:421): Capture analyzer activated: 0 Capture post processor activated: 0 Render pre processor activated: 0 [000:084] (webrtc_video_engine.cc:460): WebRtcVideoEngine::WebRtcVideoEngine() [000:085] (webrtc_voice_engine.cc:194): WebRtcVoiceEngine::WebRtcVoiceEngine [000:086] (webrtc_voice_engine.cc:217): WebRtcVoiceEngine::Init [000:087] (webrtc_voice_engine.cc:224): Supported send codecs in order of preference: [000:090] (webrtc_voice_engine.cc:227): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:090] (webrtc_voice_engine.cc:227): ISAC/16000/1 (103) [000:090] (webrtc_voice_engine.cc:227): ISAC/32000/1 (104) [000:090] (webrtc_voice_engine.cc:227): G722/8000/1 (9) [000:090] (webrtc_voice_engine.cc:227): ILBC/8000/1 (102) [000:090] (webrtc_voice_engine.cc:227): PCMU/8000/1 (0) [000:090] (webrtc_voice_engine.cc:227): PCMA/8000/1 (8) [000:090] (webrtc_voice_engine.cc:227): CN/32000/1 (106) [000:090] (webrtc_voice_engine.cc:227): CN/16000/1 (105) [000:091] (webrtc_voice_engine.cc:227): CN/8000/1 (13) [000:091] (webrtc_voice_engine.cc:227): telephone-event/48000/1 (110) [000:091] (webrtc_voice_engine.cc:227): telephone-event/32000/1 (112) [000:091] (webrtc_voice_engine.cc:227): telephone-event/16000/1 (113) [000:091] (webrtc_voice_engine.cc:227): telephone-event/8000/1 (126) [000:091] (webrtc_voice_engine.cc:230): Supported recv codecs in order of preference: [000:093] (webrtc_voice_engine.cc:233): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:093] (webrtc_voice_engine.cc:233): ISAC/16000/1 (103) [000:093] (webrtc_voice_engine.cc:233): ISAC/32000/1 (104) [000:093] (webrtc_voice_engine.cc:233): G722/8000/1 (9) [000:093] (webrtc_voice_engine.cc:233): ILBC/8000/1 (102) [000:093] (webrtc_voice_engine.cc:233): PCMU/8000/1 (0) [000:093] (webrtc_voice_engine.cc:233): PCMA/8000/1 (8) [000:093] (webrtc_voice_engine.cc:233): CN/32000/1 (106) [000:093] (webrtc_voice_engine.cc:233): CN/16000/1 (105) [000:094] (webrtc_voice_engine.cc:233): CN/8000/1 (13) [000:094] (webrtc_voice_engine.cc:233): telephone-event/48000/1 (110) [000:094] (webrtc_voice_engine.cc:233): telephone-event/32000/1 (112) [000:094] (webrtc_voice_engine.cc:233): telephone-event/16000/1 (113) [000:094] (webrtc_voice_engine.cc:233): telephone-event/8000/1 (126) [000:094] (webrtc_voice_engine.cc:310): 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, } [000:096] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [000:096] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [000:096] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [000:096] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [000:096] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [000:096] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [000:098] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:099] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:099] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:099] (apm_helpers.cc:90): Echo control set to 1 with mode 0 [000:099] (apm_helpers.cc:79): AGC set to 1 with mode 0 [000:099] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:099] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:099] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:099] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:100] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:100] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:100] (apm_helpers.cc:104): NS set to 1 [000:100] (webrtc_voice_engine.cc:466): Stereo swapping enabled? 0 [000:100] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:100] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:100] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:100] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:100] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:100] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:100] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:100] (webrtc_voice_engine.cc:536): Typing detection is enabled? 1 [000:100] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:101] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:101] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:114] (rtc_event_log_impl.cc:72): Creating legacy encoder for RTC event log. [000:114] (peer_connection_factory.cc:395): Using default network controller factory [000:114] (bitrate_prober.cc:62): Bandwidth probing enabled, set to inactive [000:115] (paced_sender.cc:427): ProcessThreadAttached 0x1fe4ee0 [000:115] (aimd_rate_control.cc:101): Using aimd rate control with back off factor 0.85 [000:116] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating. [000:128] (peer_connection.cc:1381): Adding video transceiver in response to a call to AddTrack. [000:130] (peer_connection.cc:1381): Adding video transceiver in response to a call to AddTrack. [000:130] (jsep_transport_controller.cc:1518): Generating media transport offer! [000:131] (media_session.cc:335): Duplicate id found. Reassigning from 102 to 127 [000:131] (media_session.cc:335): Duplicate id found. Reassigning from 104 to 125 [000:132] (media_session.cc:335): Duplicate id found. Reassigning from 103 to 124 [000:132] (media_session.cc:335): Duplicate id found. Reassigning from 1 to 14 [000:132] (media_session.cc:335): Duplicate id found. Reassigning from 2 to 13 [000:132] (media_session.cc:335): Duplicate id found. Reassigning from 3 to 12 [000:132] (media_session.cc:335): Duplicate id found. Reassigning from 5 to 11 [000:138] (peer_connection_integrationtest.cc:753): Caller: SetLocalDescriptionAndSendSdpMessage [000:164] (jsep_transport_controller.cc:975): Offered media transport has now been activated. [000:164] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [000:164] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [000:164] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [000:164] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [000:164] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [000:164] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [000:164] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [000:165] (p2p_transport_channel.cc:432): Set ICE ufrag: m0fl pwd: OMrArGHfpKVwcBVciMMu3RBa on transport 0 [000:177] (webrtc_video_engine.cc:472): CreateMediaChannel. Options: VideoOptions {} [000:179] (channel.cc:144): Created channel for 0 [000:190] (channel.cc:212): BaseChannel::Init_w, media_transport=1 [000:191] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:191] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:194] (peer_connection.cc:4097): Session: 8554892447944362861 Old state: kStable New state: kHaveLocalOffer [000:196] (channel.cc:1009): Setting local video description [000:204] (webrtc_video_engine.cc:945): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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}]} [000:205] (webrtc_video_engine.cc:960): Changing recv codecs from {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} to {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} [000:210] (webrtc_video_engine.cc:1061): AddSendStream: {id:9e9097d0-16bb-4981-9096-3c6144c18bf3;ssrcs:[1001470961,2241372548];ssrc_groups:{semantics:FID;ssrcs:[1001470961,2241372548]};cname:Gn2fMaPs9n2FSqz2;stream_ids:;} [000:210] (webrtc_video_engine.cc:1100): SetLocalSsrc on all the receive streams because we added a send stream. [000:210] (channel.cc:676): Add send stream ssrc: 1001470961 [000:215] (channel.cc:996): Changing video state, send=0 [000:216] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 1001470961, options: VideoOptions {is_screencast : false, }, source = (source)) [000:218] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [000:218] (basic_port_allocator.cc:851): Network manager has started [000:218] (basic_port_allocator.cc:766): Allocate ports on 1 networks [000:218] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Udp [000:218] (port.cc:318): Port[2e2fa00::1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port created with network cost 50 [000:218] (peer_connection_integrationtest.cc:683): Callee: HandleIncomingOffer [000:218] (basic_port_allocator.cc:875): Adding allocated port for 0 [000:218] (basic_port_allocator.cc:894): Port[2e2fa00:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Added port to allocator [000:219] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:219] (basic_port_allocator.cc:912): Port[2e2fa00:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Gathered candidate: Cand[:2214029314:1:udp:2122260223:192.168.1.1:49152:local::0:m0fl:OMrArGHfpKVwcBVciMMu3RBa:1:50:0] [000:219] (basic_port_allocator.cc:940): Port[2e2fa00:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port ready. [000:219] (basic_port_allocator.cc:1017): Port[2e2fa00:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port completed gathering candidates. [000:223] (peer_connection_integrationtest.cc:771): Callee: SetRemoteDescription [000:228] (jsep_transport_controller.cc:996): Returning new, client media transport. [000:228] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [000:228] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [000:228] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [000:228] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [000:228] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [000:228] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [000:229] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [000:229] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=m0fl, renomination disabled [000:237] (webrtc_video_engine.cc:472): CreateMediaChannel. Options: VideoOptions {} [000:239] (channel.cc:144): Created channel for 0 [000:243] (channel.cc:212): BaseChannel::Init_w, media_transport=1 [000:243] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:243] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:243] (peer_connection.cc:4097): Session: 2858609001489612395 Old state: kStable New state: kHaveRemoteOffer [000:243] (channel.cc:1061): Setting remote video description [000:244] (webrtc_video_engine.cc:670): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 0} [000:246] (webrtc_video_engine.cc:679): Using codec: VideoCodec[96:VP8] [000:246] (webrtc_video_engine.cc:747): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. [000:246] (webrtc_video_engine.cc:1163): AddRecvStream: {id:9e9097d0-16bb-4981-9096-3c6144c18bf3;ssrcs:[1001470961,2241372548];ssrc_groups:{semantics:FID;ssrcs:[1001470961,2241372548]};cname:Gn2fMaPs9n2FSqz2;stream_ids:;} [000:249] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 1001470961, 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: 104, red_type: 102, rtx_ssrc: 2241372548, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 103 (pt) -> 102 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, target_delay_ms: 0} [000:249] (bitrate_allocator.cc:115): Current BWE 300000 [000:249] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:249] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:255] (channel.cc:719): Add remote ssrc: 1001470961 [000:258] (channel.cc:996): Changing video state, send=0 [000:259] (peer_connection.cc:2687): Processing the MSIDs for MID=0 (streams=[]). [000:259] (peer_connection.cc:2698): Processing the addition of a remote track for MID=0. [000:259] (webrtc_video_engine.cc:1293): SetSink: ssrc:1001470961 (ptr) [000:261] (peer_connection_integrationtest.cc:919): Caller: OnIceCandidate [000:261] (peer_connection_integrationtest.cc:867): Callee: ReceiveIceMessage [000:262] (peer_connection.cc:6141): 0 is not ready to use the remote candidate because the local or remote description is not set. [000:265] (peer_connection.cc:5449): Local and Remote descriptions must be applied to get the SSL Role of the session. [000:268] (peer_connection_integrationtest.cc:753): Callee: SetLocalDescriptionAndSendSdpMessage [000:278] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [000:278] (p2p_transport_channel.cc:432): Set ICE ufrag: rokK pwd: LK+joh6nMEK+wfEigp+bpUTi on transport 0 [000:278] (dtls_transport.cc:253): DtlsTransport[0|1|__]: Other side didn't support DTLS. [000:279] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv. [000:279] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Relay [000:284] (channel.cc:553): Channel enabled [000:284] (channel.cc:996): Changing video state, send=0 [000:285] (peer_connection.cc:4097): Session: 2858609001489612395 Old state: kHaveRemoteOffer New state: kStable [000:285] (channel.cc:1009): Setting local video description [000:285] (webrtc_video_engine.cc:945): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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}]} [000:287] (webrtc_video_engine.cc:960): Changing recv codecs from {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} to {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} [000:287] (webrtc_video_engine.cc:2493): MaybeRecreateWebRtcFlexfecStream (recv) because of SetRecvParameters [000:287] (webrtc_video_engine.cc:2498): RecreateWebRtcVideoStream (recv) because of SetRecvParameters [000:287] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:287] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:287] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 1001470961, 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: 104, red_type: 102, rtx_ssrc: 2241372548, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 103 (pt) -> 102 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, target_delay_ms: 0} [000:294] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [000:295] (receive_statistics_proxy.cc:487): Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [000:296] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 1001470961, 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: 125, red_type: 127, rtx_ssrc: 2241372548, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [000:296] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:296] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:305] (webrtc_video_engine.cc:1061): AddSendStream: {id:59705b04-7769-42f0-9519-4c1fbd77bf10;ssrcs:[2603169483,2793006978];ssrc_groups:{semantics:FID;ssrcs:[2603169483,2793006978]};cname:okt4qP815bV4aTVL;stream_ids:;} [000:305] (webrtc_video_engine.cc:1859): RecreateWebRtcStream (send) because of SetCodec. [000:306] (alr_experiment.cc:66): 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 [000:307] (video_send_stream_impl.cc:249): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [2603169483], 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: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [2793006978], payload_type: 97}, c_name: okt4qP815bV4aTVL}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [000:307] (video_send_stream_impl.cc:275): ERROR: Initial encoder max bitrate = -1 which is <= 0! [000:307] (video_send_stream_impl.cc:277): Using default encoder max bitrate = 10 Mbps [000:312] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:312] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:312] (video_send_stream.cc:160): VideoSendStream::Stop [000:312] (video_send_stream_impl.cc:415): VideoSendStream::Stop [000:312] (webrtc_video_engine.cc:1100): SetLocalSsrc on all the receive streams because we added a send stream. [000:312] (webrtc_video_engine.cc:2434): RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc=2603169483 [000:312] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:312] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:312] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 1001470961, 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: 125, red_type: 127, rtx_ssrc: 2241372548, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [000:313] (video_stream_encoder.cc:579): ConfigureEncoder requested. [000:313] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [000:314] (receive_statistics_proxy.cc:487): Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [000:315] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 1001470961, local_ssrc: 2603169483, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 2241372548, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [000:316] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:316] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:317] (channel.cc:676): Add send stream ssrc: 2603169483 [000:317] (video_send_stream.cc:160): VideoSendStream::Stop [000:317] (video_send_stream_impl.cc:415): VideoSendStream::Stop [000:317] (channel.cc:996): Changing video state, send=0 [000:318] (peer_connection.cc:4033): Changing IceConnectionState 0 => 1 [000:323] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 2603169483, options: VideoOptions {is_screencast : false, }, source = (source)) [000:324] (video_stream_encoder.cc:579): ConfigureEncoder requested. [000:324] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [000:324] (basic_port_allocator.cc:851): Network manager has started [000:325] (basic_port_allocator.cc:766): Allocate ports on 1 networks [000:325] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Udp [000:325] (port.cc:318): Port[2da9850::1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port created with network cost 50 [000:325] (basic_port_allocator.cc:875): Adding allocated port for 0 [000:325] (basic_port_allocator.cc:894): Port[2da9850:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Added port to allocator [000:325] (peer_connection_integrationtest.cc:703): Caller: HandleIncomingAnswer [000:325] (basic_port_allocator.cc:912): Port[2da9850:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Gathered candidate: Cand[:2214029314:1:udp:2122260223:192.168.1.1:49153:local::0:rokK:LK+joh6nMEK+wfEigp+bpUTi:1:50:0] [000:325] (basic_port_allocator.cc:940): Port[2da9850:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port ready. [000:325] (port.cc:1126): Conn[2e75f30:0:Net[test0:192.168.1.0/24:Unknown:id=1]:bj9dwOiS:1:0:local:udp:192.168.1.1:49153->oWkzGwZn:1:2122260223:local:udp:192.168.1.1:49152|C--W|-|0|0|9115038255631187454|-]: Connection created [000:326] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:326] (p2p_transport_channel.cc:1264): Channel[0|1|__]: Created connection with origin: 2, total: 1 [000:326] (p2p_transport_channel.cc:1914): Channel[0|1|__]: Transport channel state changed from 0 to 2 [000:326] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:326] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:326] (p2p_transport_channel.cc:1476): Channel[0|1|__]: Have a pingable connection for the first time; starting to ping. [000:326] (basic_port_allocator.cc:1017): Port[2da9850:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port completed gathering candidates. [000:327] (port.cc:1756): Conn[2e75f30:0:Net[test0:192.168.1.0/24:Unknown:id=1]:bj9dwOiS:1:0:local:udp:192.168.1.1:49153->oWkzGwZn:1:2122260223:local:udp:192.168.1.1:49152|C--W|-|0|0|9115038255631187454|-]: Sent STUN ping, id=586e61563559367858516341, use_candidate=0, nomination=0 [000:327] (port.cc:528): Received STUN ping id=586e61563559367858516341 from unknown address 192.168.1.1:49153 [000:328] (port.cc:1126): Conn[2e76c40:0:Net[test0:192.168.1.0/24:Unknown:id=1]:+1BRXVrT:1:0:local:udp:192.168.1.1:49152->jmKgFMBq:1:1853824767:prflx:udp:192.168.1.1:49153|C--W|-|0|0|7962116751024340479|-]: Connection created [000:328] (p2p_transport_channel.cc:988): Adding connection from peer reflexive candidate: Cand[:2744478354:1:udp:1853824767:192.168.1.1:49153:prflx::0:rokK::1:50:0] [000:329] (port.cc:853): Port[2e2fa00:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Sent STUN ping response, to=192.168.1.1:49153, id=586e61563559367858516341 [000:329] (p2p_transport_channel.cc:1914): Channel[0|1|R_]: Transport channel state changed from 0 to 2 [000:329] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:329] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:329] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Tcp [000:330] (port.cc:318): Port[2e7a1a0::1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port created with network cost 50 [000:330] (basic_port_allocator.cc:875): Adding allocated port for 0 [000:330] (basic_port_allocator.cc:894): Port[2e7a1a0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Added port to allocator [000:330] (basic_port_allocator.cc:912): Port[2e7a1a0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Gathered candidate: Cand[:3446969586:1:tcp:1518280447:192.168.1.1:49154:local::0:m0fl:OMrArGHfpKVwcBVciMMu3RBa:1:50:0] [000:330] (basic_port_allocator.cc:940): Port[2e7a1a0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port ready. [000:330] (basic_port_allocator.cc:1017): Port[2e7a1a0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port completed gathering candidates. [000:330] (basic_port_allocator.cc:1119): All candidates gathered for 0:1:0 [000:330] (p2p_transport_channel.cc:854): P2PTransportChannel: 0, component 1 gathering complete [000:331] (port.cc:1704): Conn[2e75f30:0:Net[test0:192.168.1.0/24:Unknown:id=1]:bj9dwOiS:1:0:local:udp:192.168.1.1:49153->oWkzGwZn:1:2122260223:local:udp:192.168.1.1:49152|C--I|-|0|0|9115038255631187454|-]: Received STUN ping response, id=586e61563559367858516341, code=0, rtt=4, pings_since_last_response=586e61563559367858516341 [000:331] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed [000:331] (p2p_transport_channel.cc:1852): Channel[0|1|__]: New selected connection: Conn[2e75f30:0:Net[test0:192.168.1.0/24:Unknown:id=1]:bj9dwOiS:1:0:local:udp:192.168.1.1:49153->oWkzGwZn:1:2122260223:local:udp:192.168.1.1:49152|CRWS|S|0|0|9115038255631187454|4] [000:331] (jsep_transport_controller.cc:1214): Transport 0 writability changed to 1. [000:331] (channel.cc:583): Channel writable (0) for the first time [000:332] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:332] (basic_port_allocator.cc:1119): All candidates gathered for 0:1:0 [000:332] (p2p_transport_channel.cc:854): P2PTransportChannel: 0, component 1 gathering complete [000:333] (peer_connection_integrationtest.cc:771): Caller: SetRemoteDescription [000:334] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:334] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:334] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:335] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:335] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps [000:335] (video_stream_encoder.cc:1477): Video suspend state changed to: not suspended [000:335] (channel.cc:996): Changing video state, send=1 [000:336] (video_stream_encoder.cc:1051): Video frame parameters changed: dimensions=640x480, texture=0. [000:337] (video_stream_encoder.cc:233): Set max framerate: 60 [000:347] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 30000bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 1700000bps [000:378] (port.cc:1285): Conn[2e76c40:0:Net[test0:192.168.1.0/24:Unknown:id=1]:+1BRXVrT:1:0:local:udp:192.168.1.1:49152->jmKgFMBq:1:1853824767:prflx:udp:192.168.1.1:49153|CR-W|-|0|0|7962116751024340479|-]: Received STUN ping, id=717a5379677842775166786b [000:378] (port.cc:853): Port[2e2fa00:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Sent STUN ping response, to=192.168.1.1:49153, id=717a5379677842775166786b [000:382] (webrtc_video_engine.cc:1338): Call stats: 2322226, {send_bw_bps: 0, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1} [000:389] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [000:389] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=rokK, renomination disabled [000:389] (dtls_transport.cc:253): DtlsTransport[0|1|R_]: Other side didn't support DTLS. [000:389] (p2p_transport_channel.cc:1476): Channel[0|1|R_]: Have a pingable connection for the first time; starting to ping. [000:389] (p2p_transport_channel.cc:2440): Selecting connection for triggered check: Conn[2e76c40:0:Net[test0:192.168.1.0/24:Unknown:id=1]:+1BRXVrT:1:0:local:udp:192.168.1.1:49152->jmKgFMBq:1:1853824767:prflx:udp:192.168.1.1:49153|CR-W|-|0|0|7962116751024340479|-] [000:390] (port.cc:1756): Conn[2e76c40:0:Net[test0:192.168.1.0/24:Unknown:id=1]:+1BRXVrT:1:0:local:udp:192.168.1.1:49152->jmKgFMBq:1:1853824767:prflx:udp:192.168.1.1:49153|CR-W|-|0|0|7962116751024340479|-]: Sent STUN ping, id=79594a7249454a6f79473542, use_candidate=1, nomination=0 [000:391] (port.cc:1704): Conn[2e76c40:0:Net[test0:192.168.1.0/24:Unknown:id=1]:+1BRXVrT:1:0:local:udp:192.168.1.1:49152->jmKgFMBq:1:1853824767:prflx:udp:192.168.1.1:49153|CR-I|-|0|0|7962116751024340479|-]: Received STUN ping response, id=79594a7249454a6f79473542, code=0, rtt=1, pings_since_last_response=79594a7249454a6f79473542 [000:391] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed [000:392] (p2p_transport_channel.cc:1852): Channel[0|1|R_]: New selected connection: Conn[2e76c40:0:Net[test0:192.168.1.0/24:Unknown:id=1]:+1BRXVrT:1:0:local:udp:192.168.1.1:49152->jmKgFMBq:1:1853824767:prflx:udp:192.168.1.1:49153|CRWS|S|0|0|7962116751024340479|1] [000:392] (jsep_transport_controller.cc:1214): Transport 0 writability changed to 1. [000:392] (channel.cc:583): Channel writable (0) for the first time [000:392] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:392] (channel.cc:553): Channel enabled [000:393] (channel.cc:996): Changing video state, send=0 [000:393] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:393] (peer_connection.cc:4097): Session: 8554892447944362861 Old state: kHaveLocalOffer New state: kStable [000:393] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:393] (channel.cc:1061): Setting remote video description [000:394] (webrtc_video_engine.cc:670): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 0} [000:396] (webrtc_video_engine.cc:679): Using codec: VideoCodec[96:VP8] [000:396] (webrtc_video_engine.cc:1859): RecreateWebRtcStream (send) because of SetCodec. [000:397] (alr_experiment.cc:66): 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 [000:402] (bitrate_allocator.cc:115): Current BWE 300000 [000:408] (video_send_stream_impl.cc:249): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [1001470961], 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: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [2241372548], payload_type: 97}, c_name: Gn2fMaPs9n2FSqz2}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [000:408] (video_send_stream_impl.cc:275): ERROR: Initial encoder max bitrate = -1 which is <= 0! [000:408] (video_send_stream_impl.cc:277): Using default encoder max bitrate = 10 Mbps [000:409] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:409] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:409] (video_send_stream.cc:160): VideoSendStream::Stop [000:409] (webrtc_video_engine.cc:747): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. [000:410] (webrtc_video_engine.cc:1163): AddRecvStream: {id:59705b04-7769-42f0-9519-4c1fbd77bf10;ssrcs:[2603169483,2793006978];ssrc_groups:{semantics:FID;ssrcs:[2603169483,2793006978]};cname:okt4qP815bV4aTVL;stream_ids:;} [000:411] (video_stream_encoder.cc:579): ConfigureEncoder requested. [000:411] (video_send_stream_impl.cc:415): VideoSendStream::Stop [000:411] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 2603169483, local_ssrc: 1001470961, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 2793006978, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [000:411] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:411] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:417] (channel.cc:719): Add remote ssrc: 2603169483 [000:418] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:418] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:418] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps [000:418] (video_stream_encoder.cc:1477): Video suspend state changed to: not suspended [000:418] (channel.cc:996): Changing video state, send=1 [000:418] (peer_connection.cc:4033): Changing IceConnectionState 0 => 1 [000:418] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:419] (peer_connection.cc:2687): Processing the MSIDs for MID=0 (streams=[]). [000:419] (peer_connection.cc:2698): Processing the addition of a remote track for MID=0. [000:419] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv. [000:423] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:424] (video_stream_encoder.cc:1051): Video frame parameters changed: dimensions=640x480, texture=0. [000:424] (video_stream_encoder.cc:233): Set max framerate: 60 [000:433] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 30000bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 1700000bps [000:433] (channel.cc:996): Changing video state, send=1 [000:433] (webrtc_video_engine.cc:1293): SetSink: ssrc:2603169483 (ptr) [000:436] (peer_connection_integrationtest.cc:919): Callee: OnIceCandidate [000:436] (peer_connection_integrationtest.cc:867): Caller: ReceiveIceMessage [000:440] (peer_connection_integrationtest.cc:919): Caller: OnIceCandidate [000:440] (peer_connection_integrationtest.cc:867): Callee: ReceiveIceMessage [000:446] (peer_connection.cc:1381): Adding audio transceiver in response to a call to AddTrack. [000:447] (peer_connection.cc:1381): Adding audio transceiver in response to a call to AddTrack. [000:454] (decoder_database.cc:140): Initializing decoder with payload type '96'. [000:459] (jsep_transport_controller.cc:1513): Not regenerating media transport for the new offer in existing session. [000:463] (peer_connection.cc:5981): Changing to ICE completed state because all transports are complete. [000:463] (peer_connection.cc:4033): Changing IceConnectionState 1 => 2 [000:463] (peer_connection.cc:4033): Changing IceConnectionState 2 => 3 [000:470] (peer_connection.cc:5975): Changing to ICE connected state because all transports are writable. [000:470] (peer_connection.cc:4033): Changing IceConnectionState 1 => 2 [000:471] (peer_connection_integrationtest.cc:753): Callee: SetLocalDescriptionAndSendSdpMessage [000:480] (webrtc_video_engine.cc:2205): VideoSendStream stats: 2322325, {input_fps: 14, encode_fps: 14, encode_ms: 0, encode_usage_perc: 0, target_bps: 300000, media_bps: 49840, suspended: false, bw_adapted: false} {ssrc: 2603169483, width: 640, height: 480, 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} [000:480] (webrtc_video_engine.cc:2685): VideoReceiveStream stats: 2322325, {ssrc: 1001470961, total_bps: 0, width: 640, height: 480, key: 1, delta: 0, network_fps: 1, 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, discarded: 0, sync_offset_ms: 2147483647, cum_loss: 0, max_ext_seq: 0, nack: 0, fir: 0, pli: 0} [000:482] (webrtc_video_engine.cc:1338): Call stats: 2322325, {send_bw_bps: 300000, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1} [000:484] (p2p_transport_channel.cc:432): Set ICE ufrag: rokK pwd: LK+joh6nMEK+wfEigp+bpUTi on transport 0 [000:485] (webrtc_voice_engine.cc:1488): 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, } [000:485] (webrtc_voice_engine.cc:310): 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, } [000:485] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:485] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:485] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:485] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:485] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:485] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:485] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:485] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:485] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:485] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:486] (webrtc_voice_engine.cc:1506): 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, } [000:486] (channel.cc:144): Created channel for 1 [000:486] (channel.cc:583): Channel writable (1) for the first time [000:486] (channel.cc:212): BaseChannel::Init_w, media_transport=1 [000:486] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:487] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:487] (peer_connection.cc:4097): Session: 2858609001489612395 Old state: kStable New state: kHaveLocalOffer [000:487] (channel.cc:863): Changing voice state, recv=0 send=0 [000:487] (channel.cc:1009): Setting local video description [000:491] (webrtc_video_engine.cc:945): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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}]} [000:494] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:494] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:496] (channel.cc:996): Changing video state, send=1 [000:497] (channel.cc:871): Setting local voice description [000:498] (webrtc_voice_engine.cc:1327): 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}]} [000:498] (webrtc_voice_engine.cc:1516): Setting receive voice codecs. [000:501] (webrtc_voice_engine.cc:1787): AddSendStream: {id:5281ac43-3ad2-4c87-83be-6df3b1d5a1f9;ssrcs:[3120027264];ssrc_groups:;cname:okt4qP815bV4aTVL;stream_ids:;} [000:503] (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 [000:503] (audio_coding_module.cc:333): Created [000:504] (channel_send.cc:704): Setting media_transport_ rate observers. [000:504] (audio_send_stream.cc:142): AudioSendStream: 3120027264 [000:504] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 3120027264, extmap-allow-mixed: false, extensions: [], c_name: okt4qP815bV4aTVL}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: <unset>} [000:504] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:504] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:504] (channel.cc:676): Add send stream ssrc: 3120027264 [000:504] (channel_send.cc:816): Trying to call unset encoder. [000:504] (channel.cc:863): Changing voice state, recv=0 send=0 [000:510] (webrtc_voice_engine.cc:1488): Setting voice channel options: AudioOptions {hf: 0, } [000:510] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {hf: 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, } [000:510] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:510] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:510] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:510] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:510] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:510] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:510] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:510] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:511] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:511] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:511] (webrtc_voice_engine.cc:1506): Set voice channel options. Current options: AudioOptions {hf: 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, } [000:511] (peer_connection_integrationtest.cc:683): Caller: HandleIncomingOffer [000:512] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:519] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:527] (peer_connection_integrationtest.cc:771): Caller: SetRemoteDescription [000:538] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=rokK, renomination disabled [000:541] (webrtc_voice_engine.cc:1488): 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, } [000:541] (webrtc_voice_engine.cc:310): 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, } [000:541] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:541] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:541] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:541] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:541] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:541] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:541] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:541] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:542] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:542] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:542] (webrtc_voice_engine.cc:1506): 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, } [000:542] (channel.cc:144): Created channel for 1 [000:543] (channel.cc:583): Channel writable (1) for the first time [000:543] (channel.cc:212): BaseChannel::Init_w, media_transport=1 [000:543] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:543] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:544] (peer_connection.cc:4097): Session: 8554892447944362861 Old state: kStable New state: kHaveRemoteOffer [000:544] (channel.cc:863): Changing voice state, recv=0 send=0 [000:544] (channel.cc:1061): Setting remote video description [000:545] (webrtc_video_engine.cc:670): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 0} [000:546] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:546] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:546] (channel.cc:996): Changing video state, send=1 [000:551] (channel.cc:923): Setting remote voice description [000:552] (webrtc_voice_engine.cc:1282): 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: 1, options: AudioOptions {}} [000:552] (webrtc_voice_engine.cc:1705): Recreate all the receive streams because the send codec has changed. [000:552] (webrtc_voice_engine.cc:2166): WebRtcVoiceMediaChannel::SetMaxSendBitrate. [000:552] (webrtc_voice_engine.cc:1488): Setting voice channel options: AudioOptions {} [000:553] (webrtc_voice_engine.cc:310): 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, } [000:553] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:553] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:553] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:553] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:553] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:553] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:553] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:553] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:553] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:553] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:554] (webrtc_voice_engine.cc:1506): 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, } [000:554] (webrtc_voice_engine.cc:1852): AddRecvStream: {id:5281ac43-3ad2-4c87-83be-6df3b1d5a1f9;ssrcs:[3120027264];ssrc_groups:;cname:okt4qP815bV4aTVL;stream_ids:;} [000:554] (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 [000:554] (audio_coding_module.cc:333): Created [000:554] (audio_receive_stream.cc:117): AudioReceiveStream: 3120027264 [000:555] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3120027264, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: (Transport)} [000:555] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:555] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:555] (channel.cc:719): Add remote ssrc: 3120027264 [000:556] (channel.cc:863): Changing voice state, recv=0 send=0 [000:558] (peer_connection.cc:2687): Processing the MSIDs for MID=0 (streams=[]). [000:558] (peer_connection.cc:2687): Processing the MSIDs for MID=1 (streams=[]). [000:558] (peer_connection.cc:2698): Processing the addition of a remote track for MID=1. [000:558] (webrtc_voice_engine.cc:1965): SetOutputVolume() to 1 for recv stream with ssrc 3120027264 [000:576] (peer_connection_integrationtest.cc:753): Caller: SetLocalDescriptionAndSendSdpMessage [000:578] (srtp_filter.cc:194): Applying the same SRTP send parameters again. No-op. [000:578] (srtp_filter.cc:225): Applying the same SRTP recv parameters again. No-op. [000:578] (srtp_transport.cc:309): SRTP updated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [000:578] (p2p_transport_channel.cc:432): Set ICE ufrag: m0fl pwd: OMrArGHfpKVwcBVciMMu3RBa on transport 0 [000:578] (dtls_transport.cc:253): DtlsTransport[0|1|RW]: Other side didn't support DTLS. [000:578] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from kSendRecv to kSendRecv. [000:578] (rtp_transceiver.cc:169): Changing transceiver (MID=1) current direction from <not set> to kSendRecv. [000:578] (channel.cc:553): Channel enabled [000:579] (channel.cc:863): Changing voice state, recv=0 send=0 [000:581] (peer_connection.cc:4097): Session: 8554892447944362861 Old state: kHaveRemoteOffer New state: kStable [000:582] (channel.cc:1009): Setting local video description [000:583] (webrtc_video_engine.cc:945): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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}]} [000:586] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:586] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:591] (channel.cc:996): Changing video state, send=1 [000:594] (channel.cc:871): Setting local voice description [000:595] (webrtc_voice_engine.cc:1327): 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}]} [000:595] (webrtc_voice_engine.cc:1516): Setting receive voice codecs. [000:595] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3120027264, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: (Transport)} [000:596] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:596] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:596] (audio_receive_stream.cc:140): ~AudioReceiveStream: 3120027264 [000:597] (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 [000:597] (audio_coding_module.cc:333): Created [000:597] (audio_receive_stream.cc:117): AudioReceiveStream: 3120027264 [000:597] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3120027264, 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: (Transport)} [000:598] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:598] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:598] (webrtc_voice_engine.cc:1787): AddSendStream: {id:4bc44b0c-a4df-483e-a67d-4b15ec15326e;ssrcs:[423644137];ssrc_groups:;cname:Gn2fMaPs9n2FSqz2;stream_ids:;} [000:599] (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 [000:599] (audio_coding_module.cc:333): Created [000:599] (channel_send.cc:704): Setting media_transport_ rate observers. [000:600] (audio_send_stream.cc:142): AudioSendStream: 423644137 [000:600] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 423644137, 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: Gn2fMaPs9n2FSqz2}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), 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}}}} [000:600] (audio_encoder_opus.cc:772): Set Opus bitrate to 32000 bps. [000:600] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:600] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:601] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3120027264, local_ssrc: 423644137, 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: (Transport)} [000:601] (channel.cc:676): Add send stream ssrc: 423644137 [000:605] (webrtc_voice_engine.cc:310): 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, } [000:605] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:605] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:605] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:605] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:605] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:605] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:605] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:605] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:605] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:605] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:605] (channel.cc:863): Changing voice state, recv=1 send=1 [000:606] (webrtc_voice_engine.cc:1488): Setting voice channel options: AudioOptions {hf: 0, } [000:606] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {hf: 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, } [000:606] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:606] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:606] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:606] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:606] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:607] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:607] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:607] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:607] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:607] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:607] (webrtc_voice_engine.cc:1506): Set voice channel options. Current options: AudioOptions {hf: 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, } [000:608] (peer_connection_integrationtest.cc:703): Callee: HandleIncomingAnswer [000:608] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:612] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [000:612] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [000:612] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [000:612] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [000:612] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [000:612] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [000:612] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:615] (peer_connection_integrationtest.cc:771): Callee: SetRemoteDescription [000:615] (render_delay_buffer.cc:306): Receiving a first externally reported audio buffer delay of 0 ms. [000:615] (agc_manager_direct.cc:383): [agc] Initial GetMicVolume()=0 [000:615] (agc_manager_direct.cc:388): [agc] Initial volume too low, raising to 12 [000:621] (render_delay_buffer.cc:334): Applying total delay of 1 blocks. [000:621] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 1: 2 blocks [000:621] (echo_remover.cc:213): Gain change detected at block 1 [000:622] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 2: 3 blocks [000:622] (render_delay_buffer.cc:255): Render buffer underrun detected at block 2 [000:622] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 3: 4 blocks [000:622] (render_delay_buffer.cc:255): Render buffer underrun detected at block 3 [000:624] (srtp_filter.cc:194): Applying the same SRTP send parameters again. No-op. [000:624] (srtp_filter.cc:225): Applying the same SRTP recv parameters again. No-op. [000:625] (srtp_transport.cc:309): SRTP updated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [000:625] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=m0fl, renomination disabled [000:625] (dtls_transport.cc:253): DtlsTransport[0|1|RW]: Other side didn't support DTLS. [000:653] (channel.cc:553): Channel enabled [000:653] (channel.cc:863): Changing voice state, recv=1 send=0 [000:654] (peer_connection.cc:4097): Session: 2858609001489612395 Old state: kHaveLocalOffer New state: kStable [000:655] (channel.cc:1061): Setting remote video description [000:656] (webrtc_video_engine.cc:670): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 0} [000:665] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:665] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:665] (channel.cc:996): Changing video state, send=1 [000:666] (channel.cc:923): Setting remote voice description [000:667] (webrtc_voice_engine.cc:1282): 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: 1, options: AudioOptions {}} [000:667] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 3120027264, extmap-allow-mixed: false, extensions: [], c_name: okt4qP815bV4aTVL}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), 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}}}} [000:668] (audio_encoder_opus.cc:772): Set Opus bitrate to 32000 bps. [000:668] (webrtc_voice_engine.cc:1705): Recreate all the receive streams because the send codec has changed. [000:668] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 3120027264, 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: okt4qP815bV4aTVL}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), 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}}}} [000:669] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 3120027264, 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: okt4qP815bV4aTVL}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), 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}}}} [000:669] (webrtc_voice_engine.cc:2166): WebRtcVoiceMediaChannel::SetMaxSendBitrate. [000:669] (webrtc_voice_engine.cc:1488): Setting voice channel options: AudioOptions {} [000:669] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {hf: 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, } [000:669] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:669] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:669] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:669] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:669] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:669] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:669] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:669] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:670] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:670] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:670] (webrtc_voice_engine.cc:1506): Set voice channel options. Current options: AudioOptions {hf: 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, } [000:670] (webrtc_voice_engine.cc:1852): AddRecvStream: {id:4bc44b0c-a4df-483e-a67d-4b15ec15326e;ssrcs:[423644137];ssrc_groups:;cname:Gn2fMaPs9n2FSqz2;stream_ids:;} [000:670] (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 [000:671] (audio_coding_module.cc:333): Created [000:671] (audio_receive_stream.cc:117): AudioReceiveStream: 423644137 [000:671] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 423644137, local_ssrc: 3120027264, 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: (Transport)} [000:672] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:672] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:676] (channel.cc:719): Add remote ssrc: 423644137 [000:683] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {hf: 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, } [000:683] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:684] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:684] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:684] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:684] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:685] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:685] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:685] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:685] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:685] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:685] (channel.cc:863): Changing voice state, recv=1 send=1 [000:691] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [000:692] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [000:692] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [000:692] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [000:692] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [000:692] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [000:694] (peer_connection.cc:2687): Processing the MSIDs for MID=0 (streams=[]). [000:694] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from kSendRecv to kSendRecv. [000:694] (peer_connection.cc:2687): Processing the MSIDs for MID=1 (streams=[]). [000:694] (peer_connection.cc:2698): Processing the addition of a remote track for MID=1. [000:694] (rtp_transceiver.cc:169): Changing transceiver (MID=1) current direction from <not set> to kSendRecv. [000:695] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [000:695] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [000:695] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [000:695] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [000:695] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [000:695] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [000:697] (render_delay_buffer.cc:306): Receiving a first externally reported audio buffer delay of 0 ms. [000:697] (agc_manager_direct.cc:383): [agc] Initial GetMicVolume()=0 [000:697] (agc_manager_direct.cc:388): [agc] Initial volume too low, raising to 12 [000:699] (render_delay_buffer.cc:334): Applying total delay of 1 blocks. [000:699] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 1: 2 blocks [000:700] (echo_remover.cc:213): Gain change detected at block 1 [000:700] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 2: 3 blocks [000:700] (render_delay_buffer.cc:255): Render buffer underrun detected at block 2 [000:701] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 3: 4 blocks [000:701] (render_delay_buffer.cc:255): Render buffer underrun detected at block 3 [000:703] (webrtc_voice_engine.cc:1965): SetOutputVolume() to 1 for recv stream with ssrc 423644137 [000:711] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [000:711] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [000:711] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [000:711] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [000:711] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [000:711] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [000:716] (render_delay_buffer.cc:306): Receiving a first externally reported audio buffer delay of 0 ms. [000:716] (render_delay_buffer.cc:334): Applying total delay of 1 blocks. [000:716] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 1: 2 blocks [000:717] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 2: 3 blocks [000:717] (render_delay_buffer.cc:255): Render buffer underrun detected at block 2 [000:718] (agc_manager_direct.cc:383): [agc] Initial GetMicVolume()=12 ../../pc/peer_connection_integrationtest.cc(1502): error: Value of: caller()->audio_frames_received() >= total_caller_audio_frames_expected && caller()->min_video_frames_received_per_track() >= total_caller_video_frames_expected && callee()->audio_frames_received() >= total_callee_audio_frames_expected && callee()->min_video_frames_received_per_track() >= total_callee_video_frames_expected Actual: false Expected: true Stack trace: 000000013FD36B10: testing::DefaultValue<rtc::AsyncResolverInterface * __ptr64>::Exists 000000013FD8AAB9: webrtc::VP9EncoderImpl::PopulateCodecSpecific 000000014063625C: testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void> 00000001406360FB: testing::internal::HandleExceptionsInMethodIfSupported<testing::Test,void> 00000001406595DA: testing::Test::Run 0000000140659725: testing::TestInfo::Run ... Google Test internal frames ... ../../pc/peer_connection_integrationtest.cc(1518): error: Expected: (caller()->min_video_frames_received_per_track()) >= (total_caller_video_frames_expected), actual: 0 vs 3 Stack trace: 000000013FD36D66: testing::DefaultValue<rtc::AsyncResolverInterface * __ptr64>::Exists 000000013FD8AAB9: webrtc::VP9EncoderImpl::PopulateCodecSpecific 000000014063625C: testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void> 00000001406360FB: testing::internal::HandleExceptionsInMethodIfSupported<testing::Test,void> 00000001406595DA: testing::Test::Run 0000000140659725: testing::TestInfo::Run ... Google Test internal frames ... ../../pc/peer_connection_integrationtest.cc(3652): error: Value of: ExpectNewFrames(media_expectations) Actual: false Expected: true Stack trace: 000000013FD8AB6E: webrtc::VP9EncoderImpl::PopulateCodecSpecific 000000014063625C: testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void> 00000001406360FB: testing::internal::HandleExceptionsInMethodIfSupported<testing::Test,void> 00000001406595DA: testing::Test::Run 0000000140659725: testing::TestInfo::Run ... Google Test internal frames ... [010:812] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 1001470961, options: nullptr, source = nullptr) [010:816] (webrtc_video_engine.cc:1293): SetSink: ssrc:2603169483 nullptr [010:874] (webrtc_voice_engine.cc:1965): SetOutputVolume() to 0 for recv stream with ssrc 3120027264 [010:886] (channel.cc:563): Channel disabled [010:891] (video_send_stream.cc:160): VideoSendStream::Stop [010:891] (channel.cc:996): Changing video state, send=0 [010:891] (video_send_stream_impl.cc:415): VideoSendStream::Stop [010:891] (rtp_rtcp_impl.cc:359): Failed to send RTCP BYE [010:891] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps [010:891] (video_stream_encoder.cc:1477): Video suspend state changed to: suspended [010:892] (video_send_stream.cc:160): VideoSendStream::Stop [010:892] (video_send_stream_impl.cc:415): VideoSendStream::Stop [010:913] (video_send_stream_impl.cc:415): VideoSendStream::Stop [010:913] (video_send_stream_impl.cc:337): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [1001470961], 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: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [2241372548], payload_type: 97}, c_name: Gn2fMaPs9n2FSqz2}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [010:914] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [010:914] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [010:914] (send_statistics_proxy.cc:654): Frames encoded 104 WebRTC.Video.DroppedFrames.Capturer 0 WebRTC.Video.DroppedFrames.EncoderQueue 0 WebRTC.Video.DroppedFrames.Encoder 0 WebRTC.Video.DroppedFrames.Ratelimiter 0 [010:915] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [010:915] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [010:915] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 2603169483, local_ssrc: 1001470961, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 2793006978, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [010:915] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [010:956] (receive_statistics_proxy.cc:487): WebRTC.Video.ReceiveStreamLifetimeInSeconds 10 Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [010:957] (channel.cc:162): Destroyed channel: 0 [010:957] (message_queue.cc:518): Message took 71ms to dispatch. Posted from: cricket::ChannelManager::DestroyVideoChannel@../../pc/channel_manager.cc:276 [010:974] (channel.cc:563): Channel disabled [010:975] (channel.cc:863): Changing voice state, recv=0 send=0 [010:976] (webrtc_voice_engine.cc:1826): RemoveSendStream: 423644137 [010:976] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [010:976] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [010:976] (audio_send_stream.cc:179): ~AudioSendStream: 423644137 [010:978] (webrtc_voice_engine.cc:1903): RemoveRecvStream: 3120027264 [010:978] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [010:979] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [010:979] (audio_receive_stream.cc:140): ~AudioReceiveStream: 3120027264 [010:979] (paced_sender.cc:115): PacedSender paused. [010:979] (channel.cc:162): Destroyed channel: 1 [010:983] (peer_connection.cc:901): Session: 8554892447944362861 is destroyed. [011:005] (paced_sender.cc:427): ProcessThreadAttached 0x0 [011:005] (paced_sender.cc:427): ProcessThreadAttached 0x0 [011:006] (rtc_event_log_impl.cc:212): Stopping WebRTC event log. [011:006] (rtc_event_log_impl.cc:229): WebRTC event log successfully stopped. [011:008] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue. [011:016] (webrtc_video_engine.cc:464): WebRtcVideoEngine::~WebRtcVideoEngine [011:017] (webrtc_voice_engine.cc:203): WebRtcVoiceEngine::~WebRtcVoiceEngine [011:024] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 2603169483, options: nullptr, source = nullptr) [011:024] (webrtc_video_engine.cc:1293): SetSink: ssrc:1001470961 nullptr [011:026] (webrtc_voice_engine.cc:1965): SetOutputVolume() to 0 for recv stream with ssrc 423644137 [011:026] (channel.cc:563): Channel disabled [011:027] (video_send_stream.cc:160): VideoSendStream::Stop [011:027] (channel.cc:996): Changing video state, send=0 [011:027] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:027] (rtp_rtcp_impl.cc:359): Failed to send RTCP BYE [011:027] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps [011:027] (video_send_stream.cc:160): VideoSendStream::Stop [011:027] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:027] (video_stream_encoder.cc:1477): Video suspend state changed to: suspended [011:035] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:036] (video_send_stream_impl.cc:337): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [2603169483], 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: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [2793006978], payload_type: 97}, c_name: okt4qP815bV4aTVL}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [011:036] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:036] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:038] (send_statistics_proxy.cc:654): Frames encoded 107 WebRTC.Video.DroppedFrames.Capturer 0 WebRTC.Video.DroppedFrames.EncoderQueue 0 WebRTC.Video.DroppedFrames.Encoder 0 WebRTC.Video.DroppedFrames.Ratelimiter 0 [011:038] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:038] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:039] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 1001470961, local_ssrc: 2603169483, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 2241372548, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [011:040] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [011:041] (receive_statistics_proxy.cc:487): WebRTC.Video.ReceiveStreamLifetimeInSeconds 10 Frames decoded 104 WebRTC.Video.DroppedFrames.Receiver -104 WebRTC.Video.InterframeDelay95PercentileInMs 106 WebRTC.Video.InterframeDelay95PercentileInMs.S0 106 WebRTC.Video.MediaBitrateReceivedInKbps.S0 7 [011:041] (video_quality_observer.cc:135): WebRTC.Video.MeanTimeBetweenFreezesMs 10269 WebRTC.Video.TimeInHdPercentage 0 WebRTC.Video.TimeInBlockyVideoPercentage 0 WebRTC.Video.NumberResolutionDownswitchesPerMinute 0 WebRTC.Video.NumberFreezesPerMinute 0 WebRTC.Video.HarmonicFrameRate 10 [011:041] (channel.cc:162): Destroyed channel: 0 [011:045] (channel.cc:563): Channel disabled [011:047] (channel.cc:863): Changing voice state, recv=0 send=0 [011:048] (webrtc_voice_engine.cc:1826): RemoveSendStream: 3120027264 [011:048] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:048] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:048] (audio_send_stream.cc:179): ~AudioSendStream: 3120027264 [011:050] (webrtc_voice_engine.cc:1903): RemoveRecvStream: 423644137 [011:050] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [011:050] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [011:050] (audio_receive_stream.cc:140): ~AudioReceiveStream: 423644137 [011:051] (channel.cc:162): Destroyed channel: 1 [011:051] (paced_sender.cc:115): PacedSender paused. [011:051] (peer_connection.cc:901): Session: 2858609001489612395 is destroyed. [011:053] (paced_sender.cc:427): ProcessThreadAttached 0x0 [011:053] (paced_sender.cc:427): ProcessThreadAttached 0x0 [011:056] (rtc_event_log_impl.cc:212): Stopping WebRTC event log. [011:060] (rtc_event_log_impl.cc:229): WebRTC event log successfully stopped. [011:064] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue. [011:065] (webrtc_video_engine.cc:464): WebRtcVideoEngine::~WebRtcVideoEngine [011:065] (webrtc_voice_engine.cc:203): WebRtcVoiceEngine::~WebRtcVoiceEngine [011:075] (loopback_media_transport.cc:160): ~LoopbackMediaTransport [011:075] (loopback_media_transport.cc:160): ~LoopbackMediaTransport [ FAILED ] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/1, where GetParam() = 4-byte object <01-00 00-00> (11080 ms) [----------] 1 test from PeerConnectionIntegrationTest/PeerConnectionIntegrationTest (11081 ms total) [----------] Global test environment tear-down [==========] 1 test from 1 test suite ran. (11081 ms total) [ PASSED ] 0 tests. [ FAILED ] 1 test, listed below: [ FAILED ] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/1, where GetParam() = 4-byte object <01-00 00-00> 1 FAILED TEST [82/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/1 returned/aborted with exit code 1 (11445 ms) [83/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.IceStateSetupFailure/3 (21618 ms) [84/452] PeerConnectionBundleTest/PeerConnectionBundleTest.IgnoreCandidatesForUnusedTransportWhenBundling/1 (588 ms) [85/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportBidirectionalVideo/1 (800 ms) [86/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.Aes128Sha1_32_CipherUsedWhenSupported/1 (805 ms) [87/452] PeerConnectionJsepTest.LegacyNoMidAudioOnlyAnswer (391 ms) [88/452] PeerConnectionIceTest/PeerConnectionIceTest.IceCredentialsCreateAnswer/1 (297 ms) [89/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/15 (437 ms) [90/452] PeerConnectionMsidSignalingTest.UnifiedPlanToPlanBAnswer (500 ms) [91/452] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.DataChannelFromOpenWorksAfterPreviousChannelClosed/1 (593 ms) [92/452] PeerConnectionBundleTest/PeerConnectionBundleMatrixTest.VerifyTransportsBeforeAndAfterSettingRemoteAnswer/9 (526 ms) [93/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/1 (1011 ms) [94/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.IceStatesReachCompletion/1 (1088 ms) [95/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/3 (323 ms) [96/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.Aes128Sha1_32_CipherNotUsedWhenOnlyCallerSupported/1 (881 ms) [97/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.RtcEventLogOutputWriteCalled/1 (817 ms) [98/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetStatsForUnsignaledStreamWithOldStatsApi/1 (1016 ms) [99/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/3 (404 ms) [100/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/11 (605 ms) [101/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.ExchangeOfferAnswerWhenDtlsCertificateInConfig/1 (483 ms) [102/452] PeerConnectionRtpTestUnifiedPlan.RemoveTrackWithNullSenderTrackIsNoOp (286 ms) [103/452] PeerConnectionSimulcastTests.RidsAreAutogeneratedIfNotProvided (252 ms) [104/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/19 (468 ms) [105/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.ChangeSsrcOnTrackInLocalSessionDescription/1 (345 ms) [106/452] PeerConnectionBundleTest/PeerConnectionBundleTest.CreateRtcpTransportOnlyBeforeAnswerWithRtcpMuxNegotiate/1 (499 ms) [107/452] PeerConnectionBundleTest/PeerConnectionBundleTest.RejectBundleGroupWithNonExistingMid/1 (332 ms) [108/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationReturnsRangeErrorForBadCandidatePoolSize/1 (279 ms) [109/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/19 (554 ms) [110/452] PeerConnectionIceTest/PeerConnectionIceTest.RemoveCandidateRemovesFromRemoteDescription/1 (505 ms) [111/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/11 (331 ms) [112/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.IceRenominationOptionInOfferAndAnswer/1 (355 ms) [113/452] WebRtcSdpTest.ParseConnectionDataIPv6 (243 ms) [114/452] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingRemoteRejectedAndNotRejectedLocal/3 (516 ms) [115/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/7 (516 ms) [116/452] WebRtcSdpTest.ParseMediaTransportSkipRepeatedTransport (249 ms) [117/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/7 (361 ms) [118/452] PeerConnectionIceTest/PeerConnectionIceTest.OfferFromLiteIceControllingAndAnswerFromLiteIceControlled/1 (567 ms) [119/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/3 (450 ms) [120/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.VerifyBestConnection/5 (814 ms) [121/452] PeerConnectionIceTest/PeerConnectionIceTest.OfferAnswerWithNoTransportsDoesNotChangeIceConnectionState/1 (339 ms) [122/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.ExchangeOfferAnswerWhenDtlsOn/1 (565 ms) [123/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/3 (381 ms) [124/452] PeerConnectionInterfaceTestPlanB.Renegotiate (546 ms) [125/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationInteropTest.OneAudioLocalToNoMediaRemote/1 (595 ms) [126/452] PeerConnectionMediaTestPlanB.SimulcastAnswer (315 ms) [127/452] PeerConnectionJsepTest/RecycleMediaSectionTest.CurrentLocalOnlyRejected/3 (661 ms) [128/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.NonGcmCipherUsedWhenOnlyCalleeSupportsGcm/1 (965 ms) [129/452] PeerConnectionBundleTest/PeerConnectionBundleTest.RemovingContentAndRejectBundleGroup/1 (361 ms) [130/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetSourcesAudio/1 (714 ms) [131/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.ExtmapAllowMixedIsConfigurable/1 (306 ms) [132/452] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetRemoteAnswer/3 (417 ms) [133/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GcmCipherUsedWhenGcmSupported/1 (846 ms) [134/452] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetRemoteAnswer/7 (335 ms) [135/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CantChangePoolSizeAfterSetLocalDescription/1 (279 ms) [136/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/11 (368 ms) [137/452] PeerConnectionJsepTest.SetRemoteOfferReusesTransceiversOfBothTypes (365 ms) [138/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.OffersAndAnswersHaveTrickleIceOption/1 (332 ms) [139/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/7 (422 ms) [140/452] PeerConnectionSimulcastMetricsTests.SimulcastDisabledIsNotLogged (381 ms) [141/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInAnswerWhenDtlsEnabled/1 (342 ms) [142/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/3 (264 ms) [143/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInAnswerWhenEncryptionDisabled/1 (322 ms) [144/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/27 (361 ms) [145/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CanSetSdesGcmRemoteOfferAndLocalAnswer/1 (481 ms) [146/452] PeerConnectionRtpTestUnifiedPlan.AddRemoveAddTrackOffersWorksVideo (334 ms) [147/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/15 (358 ms) [148/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferReceiveAnswer/1 (420 ms) [149/452] PeerConnectionJsepTest.SetRemoteOfferWithTwoTracksSameStreamFiresOnAddTrack (374 ms) [150/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.IceCandidates/1 (338 ms) [151/452] PeerConnectionFactoryTest.CreatePCUsingTurnUrlWithTransportParam (233 ms) [152/452] PeerConnectionJsepTest.RtpReceiverCreatedBySetRemoteDescriptionHasSignaledTrackId (377 ms) [153/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.TestRejectRtpDataChannelInAnswer/1 (257 ms) [154/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/7 (309 ms) [155/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetRemoteAudioSSLCertificateReturnsExchangedCertificate/1 (481 ms) [156/452] PeerConnectionJsepTest.SetRemoteOfferDoesNotReuseTransceiverIfDirectionSendOnly (401 ms) [157/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInAnswerWhenDtlsDisabled/1 (310 ms) [158/452] PeerConnectionBundleTest/PeerConnectionBundleTest.FailToSetRemoteOfferWithNoBundleWhenBundlePolicyMaxBundle/1 (496 ms) [159/452] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.DataChannelIdAssignment/1 (584 ms) [160/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/11 (268 ms) [161/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.VerifyBestConnection/1 (875 ms) [162/452] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetLocalAnswer/7 (307 ms) [163/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.ReceiveOfferCreateAnswer/1 (457 ms) [164/452] PeerConnectionSignalingTest/PeerConnectionSignalingTest.SessionVersionIncrementedInSubsequentDifferentAnswer/1 (319 ms) [165/452] PeerConnectionRtpTest/PeerConnectionRtpTest.RemoveTrackWithStreamFiresOnRemoveTrack/1 (367 ms) [166/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/7 (361 ms) [167/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/3 (416 ms) [168/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/31 (332 ms) [169/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.DataChannelOnlyOfferWithMaxBundlePolicy/1 (319 ms) [170/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/23 (262 ms) [171/452] PeerConnectionRtpTest/PeerConnectionRtpTest.AddTrackWithoutStreamAddsReceiver/1 (374 ms) [172/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/11 (362 ms) [173/452] SimulcastSdpSerializerTest.Deserialize_PausedStreams (267 ms) [174/452] PeerConnectionJsepTest.TwoVideoPlanBToUnifiedPlanFails (302 ms) [175/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/15 (497 ms) [176/452] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.NoSctpTransportCreatedIfMediaTransportDataChannelsEnabled/1 (268 ms) [177/452] PeerConnectionMediaTest/PeerConnectionMediaTest.AnswerHasDifferentDirectionsForAudioVideo/1 (257 ms) [178/452] PeerConnectionMediaTest/PeerConnectionMediaTest.CreateAnswerWithNoStreamsAndDefaultOptionsDoesNotReject/1 (311 ms) [179/452] PeerConnectionIceTest/PeerConnectionIceTest.SetLocalDescriptionFailsIfNoIceCredentials/1 (350 ms) [180/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/3 (332 ms) [181/452] NumberOfSendEncodings/PeerConnectionSimulcastMetricsTests.NumberOfSendEncodingsIsLogged/3 (278 ms) [182/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/11 (379 ms) [183/452] PeerConnectionSimulcastTests.SimulcastAppearsInSessionDescription (355 ms) [184/452] PeerConnectionBundleTest/PeerConnectionBundleMatrixTest.VerifyTransportsBeforeAndAfterSettingRemoteAnswer/1 (467 ms) [185/452] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingLocalRejectedAndNotRejectedRemote/3 (470 ms) [186/452] PeerConnectionJsepTest.SetLocalAnswerUpdatesCurrentDirection (421 ms) [187/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AnswererRejectsAudioAndVideoSections/1 (657 ms) [188/452] PeerConnectionBundleTest/PeerConnectionBundleTest.MaxBundleNotSupportedInAnswer/1 (480 ms) [189/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.GetStatsForSpecificTrack/1 (549 ms) [190/452] PeerConnectionJsepTest.DataChannelDoesNotRecycleMediaSection (486 ms) [191/452] PeerConnectionIceTest/PeerConnectionIceTest.CreateAnswerHasNewUfragPwdForOnlyMediaSectionWhichRestarted/1 (750 ms) [192/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationCausingPartialIceRestart/1 (537 ms) [193/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreatePeerConnectionWithDifferentConfigurations/1 (365 ms) [194/452] PeerConnectionIceTest/PeerConnectionIceTest.CannotAddCandidateWhenPeerConnectionClosed/1 (634 ms) [195/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/3 (578 ms) [196/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AddMediaToConnectedBundleDoesNotRestartIce/1 (805 ms) [197/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/23 (306 ms) [198/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CnameGenerationInOffer/1 (312 ms) [199/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInOfferWithSdesAndGcm/1 (325 ms) [200/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/19 (371 ms) [201/452] PeerConnectionInterfaceTestPlanB.RemoveTrackAfterAddStream (328 ms) [202/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/39 (342 ms) [203/452] PeerConnectionMediaTest/PeerConnectionMediaTest.CreateAnswerWithNoVoiceActivityDetectionIncludesNoComfortNoiseCodecs/1 (342 ms) [204/452] PeerConnectionInterfaceTestPlanB.AddedStreamsPresentInOffer (355 ms) [205/452] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.CreateOfferWithNoDataChannelsGivesNoDataSection/1 (255 ms) [206/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportDataChannelConfigSentToOtherSide/1 (456 ms) [207/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetLocalAnswerWithNoCryptosWhenSdesOn/1 (433 ms) [208/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetRemoteOfferWithNoFingerprintWhenDtlsOn/1 (371 ms) [209/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationChangesCandidateFilter/1 (332 ms) [210/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferWithRtpMux/1 (301 ms) [211/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/23 (282 ms) [212/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/15 (291 ms) [213/452] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/16 (300 ms) [214/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferFailsWithInvalidOfferToReceiveVideo/1 (296 ms) [215/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferWithDefaultOfferAnswerOptions/1 (303 ms) [216/452] RtpTransceiverDirectionTest/EnumerateAllDirectionsTest.TestReversedSwapped/3 (269 ms) [217/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/7 (309 ms) [218/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/11 (337 ms) [219/452] RTCStatsCollectorTest.GetStatsWithReceiverSelector (285 ms) [220/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateNegativeMinFails/1 (262 ms) [221/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.RejectMediaContent/1 (388 ms) [222/452] WebRtcSdpTest.DeserializeSessionDescriptionWithoutRtpmap (233 ms) [223/452] PeerConnectionRtpTestUnifiedPlan.AddTransceiverWithTrackCreatesSenderWithTrack (322 ms) [224/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/3 (320 ms) [225/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateMaxLessThanMinFails/1 (288 ms) [226/452] PeerConnectionSimulcastTests.NegotiationDoesNotInvalidateParameterTransactions (571 ms) [227/452] PeerConnectionJsepTest.StoppingTransceiverInOfferStopsTransceiverOnRemoteSide (454 ms) [228/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CloseAndGetStats/1 (435 ms) [229/452] PeerConnectionInterfaceTestPlanB.RemoveTrackThenRejectMediaContent (439 ms) [230/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/19 (551 ms) [231/452] PeerConnectionInterfaceTestPlanB.SignalSameTracksInSeparateMediaStream (418 ms) [232/452] PeerConnectionJsepTest.CreateOfferGeneratesUniqueMidForDataSectionIfAlreadyTaken (519 ms) [233/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.ReceiveUpdatedAudioOfferWithBadCodecs/1 (401 ms) [234/452] PeerConnectionBundleTest/PeerConnectionBundleMatrixTest.VerifyTransportsBeforeAndAfterSettingRemoteAnswer/5 (606 ms) [235/452] PeerConnectionRtpTest/PeerConnectionRtpTest.AddTrackWithStreamFiresOnAddTrack/1 (381 ms) [236/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/15 (500 ms) [237/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/7 (578 ms) [238/452] PeerConnectionSignalingTest/PeerConnectionSignalingTest.CreateAnswerFailsIfStable/1 (382 ms) [239/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/7 (294 ms) [240/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/7 (505 ms) [241/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferWithAudioOnlyOptions/1 (344 ms) [242/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/19 (338 ms) [243/452] PeerConnectionSimulcastTests.RejectedSimulcastLayersAreDeactivated (496 ms) [244/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.SessionErrorIfFingerprintInvalid/1 (452 ms) [245/452] PeerConnectionJsepTest.MultipleDataChannelsCreateOnlyOneDataSection (359 ms) [246/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateCurrentLessThanImplicitMin/1 (322 ms) [247/452] IceServerParsingTest.ParseHostnameAndPort (251 ms) [248/452] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.DataSectionRejectedIfCalleeHasSctpDisabled/1 (344 ms) [249/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/7 (394 ms) [250/452] SctpDataChannelTest.SendDataSsrc (275 ms) [251/452] MediaStreamTest.ChangeVideoTrack (216 ms) [252/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.GetStatsForInvalidTrack/1 (598 ms) [253/452] PeerConnectionInterfaceTestPlanB.SdpWithoutMsidAndStreamsCreatesDefaultStream (349 ms) [254/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetIceRegatherIntervalRangeWithoutContinualGatheringFails/1 (279 ms) [255/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.AddTrackBeforeConnecting/1 (358 ms) [256/452] PeerConnectionRtpTestUnifiedPlan.AddAudioTransceiverCreatesAudioSenderAndReceiver (388 ms) [257/452] PeerConnectionMediaTest/PeerConnectionMediaTest.CombinedAudioVideoBweConfigPropagatedToMediaEngine/1 (371 ms) [258/452] HasStream/StatsCollectorTrackTest.GetStatsFromRemoteStream/0 (232 ms) [259/452] PeerConnectionMediaTest/PeerConnectionMediaOfferDirectionTest.VerifyDirection/3 (320 ms) [260/452] PeerConnectionSignalingTest/PeerConnectionSignalingTest.FailToSetNullLocalDescription/1 (284 ms) [261/452] PeerConnectionRtpTestUnifiedPlan.AddAudioTrackCreatesAudioSender (289 ms) [262/452] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/0 (239 ms) [263/452] PeerConnectionIceTest/PeerConnectionIceTest.LocalDescriptionUpdatedWhenContinualGathering/1 (383 ms) [264/452] PeerConnectionMediaTest/PeerConnectionMediaTest.FailToSetLocalDescriptionIfCreateMediaChannelFails/1 (317 ms) [265/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.TestSendOnlyDataChannel/1 (394 ms) [266/452] PeerConnectionMediaTest/PeerConnectionMediaTest.NewStreamInRemoteOfferAddsRecvStreams/1 (417 ms) [267/452] TrackMediaInfoMapTest.SingleSenderReceiverPerTrackWithSsrcNotUnique (240 ms) [268/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.VerifyIceStates/5 (35193 ms) [269/452] RtpTransceiverDirectionTest/EnumerateAllDirectionsTest.TestIdentity/3 (203 ms) [270/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/43 (250 ms) [271/452] PeerConnectionIceTest/PeerConnectionIceTest.TrickledSingleCandidateAddedToRemoteDescription/1 (365 ms) [272/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportDataChannelEndToEnd/1 (395 ms) [273/452] HasStream/StatsCollectorTrackTest.FilterOutNegativeInitialValues/0 (205 ms) [274/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/31 (239 ms) [275/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationReturnsInvalidParameterIfCredentialsMissing/1 (283 ms) [276/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.TestDataChannel/1 (332 ms) [277/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.ReceiveFireFoxOffer/1 (353 ms) [278/452] PeerConnectionInterfaceTestPlanB.SdpWithMsidWithoutSsrcCreatesDefaultStream (359 ms) [279/452] PeerConnectionJsepTest.NoMsidInOfferIfTransceiverDirectionHasNoSend (289 ms) [280/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.DataBufferedUntilRtpDataChannelObserverRegistered/1 (490 ms) [281/452] SdpFormatReceivedTest.SimplePlanBIsReportedAsSimple (328 ms) [282/452] VideoTrackTest.RenderVideo (200 ms) [283/452] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingRemoteRejectedAndNoLocal/3 (345 ms) [284/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.RenegotiationNeededForNewRtpDataChannel/1 (276 ms) [285/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/27 (334 ms) [286/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/27 (379 ms) [287/452] PeerConnectionFactoryTest.CheckRtpReceiverDataCapabilities (221 ms) [288/452] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetLocalAnswer/3 (341 ms) [289/452] PeerConnectionSignalingTest/PeerConnectionSignalingTest.CreateOffersAndShutdown/1 (302 ms) [290/452] SignalingProxyTest.Method0 (207 ms) [291/452] PeerConnectionRtpTestPlanB.MultipleRemoteSendersWithoutStreamParamIdAddsMultipleReceivers (345 ms) [292/452] PeerConnectionRtpTestUnifiedPlan.RenegotiationNeededAfterTransceiverSetDirection (317 ms) [293/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/19 (256 ms) [294/452] IceServerParsingTest.ParseMultipleUrls (246 ms) [295/452] PeerConnectionJsepTest.EmptyInitialOffer (270 ms) [296/452] PeerConnectionMediaTest/PeerConnectionMediaTest.FailToApplyDescriptionIfVideoEncoderHasEverFailed/1 (352 ms) [297/452] PeerConnectionRtpTestUnifiedPlan.AddTrackChangesDirectionFromInactiveToSendOnly (255 ms) [298/452] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.RtpDataChannelCreatedEvenIfSctpAvailable/1 (286 ms) [299/452] PeerConnectionRtpTestUnifiedPlan.AddTrackErrorIfTrackAlreadyHasSender (299 ms) [300/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateReliableRtpDataChannelShouldFail/1 (332 ms) [301/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateCurrentNegativeFails/1 (272 ms) [302/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.StartAndStopLoggingToFileAfterPeerConnectionClosed/1 (315 ms) [303/452] RTCStatsCollectorTest.MultipleCallbacksWithInvalidatedCacheInBetween (263 ms) [304/452] Encrypted/WebRtcSdpExtmapTest.DeserializeSessionDescriptionWithInvalidExtmap/0 (308 ms) [305/452] RtpParametersConversionTest.ToCricketStreamParamsVecSimple (198 ms) [306/452] RTCStatsCollectorTest.CollectRTCOutboundRTPStreamStats_Video (268 ms) [307/452] SimulcastSdpSerializerTest.Deserialize_WithSomeAlternatives (220 ms) [308/452] WebRtcSdpTest.DeserializeSessionDescriptionWithSendOnlyContent (250 ms) [309/452] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.CannotEnableBothMediaTransportAndRtpDataChannels/1 (275 ms) [310/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.GetConfigurationAfterCreatePeerConnection/1 (263 ms) [311/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationChangesIceCheckInterval/1 (284 ms) [312/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/11 (267 ms) [313/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationFailsAfterClose/1 (340 ms) [314/452] SctpDataChannelTest.StateTransition (199 ms) [315/452] NumberOfSendEncodings/PeerConnectionSimulcastMetricsTests.NumberOfSendEncodingsIsLogged/7 (291 ms) [316/452] SctpDataChannelTest.VerifyMessagesAndBytesSent (289 ms) [317/452] RtpParametersConversionTest.ToAudioRtpCodecParameters (213 ms) [318/452] RtpParametersConversionTest.ToRtpParameters (194 ms) [319/452] DtmfSenderTest.InsertDtmfWhileSenderIsDeleted (252 ms) [320/452] RidDescriptionSdpSerializerTest.Serialize_FormatListSingleFormat (198 ms) [321/452] RtpSenderReceiverTest.AudioSenderHasDtmfSender (280 ms) [322/452] WebRtcSdpTest.SerializeCandidates (202 ms) [323/452] RtpSenderReceiverTest.VideoSenderCantSetUnimplementedRtpParameters (367 ms) [324/452] WebRtcSdpTest.DeserializeSessionDescription (252 ms) [325/452] RidDescriptionSdpSerializerTest.Deserialize_SimpleSendCase (291 ms) [326/452] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithSendSet/7 (235 ms) [327/452] SimulcastDeserializationErrors/SimulcastSdpSerializerTest.SimulcastDeserializationFailed/3 (226 ms) [328/452] WebRtcSdpTest.DeserializeSerializeRtcpFbWildcard (307 ms) [329/452] SimulcastDeserializationErrors/SimulcastSdpSerializerTest.SimulcastDeserializationFailed/7 (245 ms) [330/452] SctpDataChannelTest.SendUnorderedAfterReceivesOpenAck (213 ms) [331/452] PeerConnectionFactoryTest.CheckRtpSenderVideoCapabilities (318 ms) [332/452] RTCStatsCollectorTest.CollectRemoteRTCMediaStreamStatsAndRTCMediaStreamTrackStats_Video (213 ms) [333/452] PeerConnectionUsageHistogramTest.UsageFingerprintHistogramFromTimeout (316 ms) [334/452] RTCStatsCollectorTest.CollectRTCCertificateStatsChain (279 ms) [335/452] RTCStatsTraversalTest.CyclicGraph (219 ms) [336/452] RTCStatsCollectorTest.DoNotCrashOnSsrcChange (290 ms) [337/452] RtpParametersConversionTest.ToAudioCodecInvalidParameters (242 ms) [338/452] RtpParametersConversionTest.ToCricketCodecs (203 ms) [339/452] RtpParametersConversionTest.ToStreamParamsWithPrimarySsrcSetAndRtxUnset (213 ms) [340/452] RtpSenderReceiverTest.RemoteVideoTrackDisable (274 ms) [341/452] RtpSenderReceiverTest.AudioSenderEarlyWarmupTrackThenSsrc (238 ms) [342/452] RtpParametersConversionTest.ToAudioRtpCodecCapability (298 ms) [343/452] RtpSenderReceiverTest.VideoSenderSsrcSetToZero (220 ms) [344/452] RtpSenderReceiverTest.VideoSenderSsrcChanged (206 ms) [345/452] RtpSenderReceiverTest.AudioSenderMustCallGetParametersBeforeSetParametersBeforeNegotiation (250 ms) [346/452] RtpSenderReceiverTest.SetAudioMaxSendBitrate (201 ms) [347/452] RtpSenderReceiverTest.AudioSenderCheckTransactionIdRefresh (229 ms) [348/452] RtpSenderReceiverTest.VideoSenderInitParametersMovedAfterNegotiation (263 ms) [349/452] RtpSenderReceiverTest.VideoSenderCantSetUnimplementedEncodingParametersWithSimulcast (225 ms) [350/452] RtpSenderReceiverTest.VideoSenderSetParametersInvalidatesTransactionId (261 ms) [351/452] WebRtcSdpTest.DeserializeSessionDescriptionWithBandwidth (229 ms) [352/452] TrackMediaInfoMapTest.SingleSenderReceiverPerTrackWithAudioAndVideoUseSameSsrc (266 ms) [353/452] WebRtcSdpTest.DeserializeMediaContentDescriptionWithExtmapAllowMixed (220 ms) [354/452] WebRtcSdpTest.DeserializeSdpWithRtpDataChannelsAndBandwidth (207 ms) [355/452] WebRtcSdpTest.DeserializeSessionDescriptionSpecialMsid (220 ms) [356/452] WebRtcSdpTest.DeserializePlanBSessionDescription (286 ms) [357/452] WebRtcSdpTest.DeserializeMsidAttributeWithMissingTrackId (208 ms) [358/452] WebRtcSdpTest.DeserializeUnifiedPlanSessionDescriptionNoSsrcSignaling (253 ms) [359/452] WebRtcSdpTest.TestDeserializeIgnoresMalformedRidLines (216 ms) [360/452] WebRtcSdpTest.PlanBHasSsrcAttributeMsidSignaling (267 ms) [361/452] WebRtcSdpTest.ParseNoMid (217 ms) [362/452] JsepSessionDescriptionTest/EnumerateAllSdpTypesTest.TestIdentity/2 (225 ms) [363/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/35 (192 ms) [364/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/31 (240 ms) [365/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/15 (277 ms) [366/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/43 (213 ms) [367/452] TrackMediaInfoMapDeathTest.MultipleOneSsrcReceiversPerTrack (5832 ms) [368/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/35 (213 ms) [369/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateSubsequentInactiveOffer/1 (390 ms) [370/452] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithSendSet/3 (200 ms) [371/452] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/8 (197 ms) [372/452] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithRecvSet/3 (236 ms) [373/452] HasStream/StatsCollectorTrackTest.RemoteSsrcInfoIsPresent/0 (208 ms) [374/452] HasStream/StatsCollectorTrackTest.LocalAndRemoteTracksWithSameSsrc/0 (230 ms) [375/452] PeerConnectionSignalingTest/PeerConnectionSignalingTest.SetLocalOfferTwiceWorks/1 (286 ms) [376/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/3 (332 ms) [377/452] PeerConnectionMediaConfigTest.TestDefaults (273 ms) [378/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetLocalOfferWithNoCryptosWhenSdesOn/1 (271 ms) [379/452] PeerConnectionJsepTest.SetLocalEmptyOfferCreatesNoTransceivers (286 ms) [380/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateSctpDataChannelWithInvalidIdShouldFail/1 (315 ms) [381/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreatePeerConnectionWithPooledCandidates/1 (272 ms) [382/452] PeerConnectionMediaTest/PeerConnectionMediaOfferDirectionTest.VerifyDirection/7 (233 ms) [383/452] PeerConnectionRtpTest/PeerConnectionRtpTest.LegacyObserverNotCalledIfPeerConnectionDereferenced/1 (256 ms) [384/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/39 (289 ms) [385/452] PeerConnectionRtpTestUnifiedPlan.TwoAddTrackCreatesTwoTransceivers (224 ms) [386/452] PeerConnectionRtpTestUnifiedPlan.CheckForInvalidEncodingParameters (234 ms) [387/452] LocalAudioSourceTest.InitWithAudioOptions (263 ms) [388/452] WebRtcSdpTest.DeserializeAndSerializeSdpWithIceLite (265 ms) [389/452] PeerConnectionMediaTest/PeerConnectionMediaTest.ReOfferHasSameMidsAsFirstOffer/1 (312 ms) [390/452] PeerConnectionMediaTest/PeerConnectionMediaOfferDirectionTest.VerifyDirection/11 (242 ms) [391/452] RtpSenderReceiverTest.AudioReceiverCanSetFrameDecryptor (221 ms) [392/452] PeerConnectionUsageHistogramTest.FingerprintStunTurn (259 ms) [393/452] SctpDataChannelTest.ClosedOnTransportError (217 ms) [394/452] RtpTransceiverTest.CanUnsetChannelOnStoppedTransceiver (288 ms) [395/452] SctpDataChannelTest.TransportDestroyedWhileDataBuffered (196 ms) [396/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationCausingIceRestart/1 (405 ms) [397/452] JsepSessionDescriptionTest.AddAndRemoveCandidatesWithMid (206 ms) [398/452] DtmfSenderTest.CanInsertDtmf (273 ms) [399/452] PlayoutLatencyTest.Caching (230 ms) [400/452] RTCStatsTraversalTest.SelfReference (207 ms) [401/452] RTCStatsCollectorTest.CollectRTCPeerConnectionStats (221 ms) [402/452] RtpSenderReceiverTest.SetVideoMinMaxSendBitrateSimulcast (230 ms) [403/452] RtpSenderReceiverTest.VideoReceiverCanGetParametersWithSimulcast (226 ms) [404/452] SimulcastSdpSerializerTest.Serialize_ComplexSerialization (240 ms) [405/452] WebRtcSdpTest.SerializeSessionDescriptionWithInactiveContent (200 ms) [406/452] FakeAdmTest.RecordTest (250 ms) [407/452] StatsCollectorTest.ChainlessCertificateReportsCreated (299 ms) [408/452] WebRtcSdpTest.SerializeSessionDescriptionWithRtpDataChannel (212 ms) [409/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateSctpDataChannel/1 (239 ms) [410/452] HasStream/StatsCollectorTrackTest.AudioBandwidthEstimationInfoIsReported/0 (213 ms) [411/452] DisableSimulcastLayersInSender/RtpSenderReceiverTest.DisableSimulcastLayersWithoutMediaEngine/3 (332 ms) [412/452] HasStream/StatsCollectorTrackTest.TransportObjectLinkedFromSsrcObject/0 (226 ms) [413/452] SctpDataChannelTest.VerifyMessagesAndBytesReceived (217 ms) [414/452] DtmfSenderTest.InsertDtmfWithInvalidDurationOrGap (227 ms) [415/452] OwnedProxyTest.BasicTest (319 ms) [416/452] RtpSenderReceiverTest.VideoReceiverCanSetFrameDecryptor (237 ms) [417/452] WebRtcSdpTest.DeserializeSessionDescriptionWithRejectedAudioVideo (231 ms) [418/452] WebRtcSdpTest.DeserializeCandidateWithUfragPwd (207 ms) [419/452] WebRtcSdpTest.BandwidthLimitOfNegativeOneIgnored (194 ms) [420/452] WebRtcSdpTest.MediaContentOrderMaintainedRoundTrip (207 ms) [421/452] WebRtcSdpTest.TestDeserializeSimulcastAttribute (276 ms) [422/452] PeerConnectionMediaTest/PeerConnectionMediaTest.TestAVOfferWithVideoOnlyAnswer/1 (198 ms) [423/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/47 (230 ms) [424/452] RtpTransceiverDirectionTest/EnumerateAllDirectionsTest.TestReversedIdentity/3 (211 ms) [425/452] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/12 (204 ms) [426/452] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/4 (229 ms) [427/452] DisableSimulcastLayersInSender/RtpSenderReceiverTest.DisableSimulcastLayersWithMediaEngine/3 (256 ms) [428/452] JsepSessionDescriptionTest.AddHostnameCandidate (201 ms) [429/452] Encrypted/WebRtcSdpExtmapTest.DeserializeSessionDescriptionWithSessionLevelExtmap/0 (242 ms) [430/452] JsepSessionDescriptionTest.SerializeSessionDescriptionWithBothIPFamilies (225 ms) [431/452] ProxyTest.ConstMethod0 (233 ms) [432/452] SctpUtilsTest.WriteParseAckMessage (210 ms) [433/452] RtpSenderReceiverTest.AddAndDestroyVideoRtpReceiver (240 ms) [434/452] RtpParametersConversionTest.ToCricketFeedbackParamErrors (265 ms) [435/452] RidDescriptionSdpSerializerTest.Deserialize_WithRestriction (210 ms) [436/452] StatsCollectorTest.SessionObjectExists (208 ms) [437/452] WebRtcSdpTest.SerializeSessionDescriptionWithExtmapAllowMixed (210 ms) [438/452] WebRtcSdpTest.SerializeSessionDescriptionWithBandwidth (268 ms) [439/452] WebRtcSdpTest.SerializeMediaTransportSettings (207 ms) [440/452] WebRtcSdpTest.DeserializeSdpWithInvalidAttributeValue (236 ms) [441/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/47 (206 ms) [442/452] WebRtcSdpTest.SerializeAndDeserializeWithHostnameConnectionAddress (269 ms) [443/452] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithRecvSet/7 (252 ms) [444/452] HasStream/StatsCollectorTrackTest.VerifyVideoSendSsrcStats/0 (200 ms) [445/452] PeerConnectionMediaTest/PeerConnectionMediaTest.MediaTransportOnlyForDataChannels/1 (250 ms) [446/452] WebRtcSdpTest.SerializeAudioFmtpWithUnknownParameter (216 ms) [447/452] RidDescriptionSdpSerializerTest.Deserialize_RestrictionHasNoValue (200 ms) [448/452] RtpSenderReceiverTest.LocalAudioTrackDisable (211 ms) [449/452] SignalingProxyTest.Method2 (275 ms) [450/452] WebRtcSdpTest.SerializeBothMediaSectionAndSsrcAttributeMsid (246 ms) [451/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/11 (233 ms) [452/452] RtpSenderReceiverTest.RemoteVideoTrackLatency (199 ms) [453/452] HasStream/StatsCollectorTrackTest.TrackObjectExistsWithoutUpdateStats/0 (218 ms) [454/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.VerifyIceStates/1 (45485 ms) FAILED TESTS (2/452): 10627 ms: .\peerconnection_unittests.exe PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportBidirectionalVideo/1 (try #1) 11445 ms: .\peerconnection_unittests.exe PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/1 (try #1) 43bb18f97cdb7610: exit 0 Additional test environment: CHROME_DEVEL_SANDBOX=/opt/chromium/chrome_sandbox CHROME_HEADLESS=1 LANG=en_US.UTF-8 Command: e:\b\s\w\ir\.swarming_module_cache\vpython\37d31f\Scripts\python.exe ..\..\tools_webrtc\gtest-parallel-wrapper.py --output_dir=e:\b\s\w\ioshjw9j\test_logs --gtest_color=no --timeout=900 --retry_failed=3 .\peerconnection_unittests.exe --isolated-script-test-output=e:\b\s\w\ioshjw9j\output.json --isolated-script-test-perf-output=e:\b\s\w\ioshjw9j\perftest-output.json gtest-parallel-wrapper: Executing command e:\b\s\w\ir\.swarming_module_cache\vpython\37d31f\Scripts\python.exe e:\b\s\w\ir\third_party\gtest-parallel\gtest-parallel --shard_index=1 --shard_count=4 --output_dir=e:\b\s\w\ioshjw9j\test_logs --retry_failed=3 --gtest_color=no --timeout=900 --dump_json_test_results=e:\b\s\w\ioshjw9j\output.json .\peerconnection_unittests.exe -- --isolated-script-test-perf-output=e:\b\s\w\ioshjw9j\perftest-output.json [0/452] Running tests... [1/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.ClosingConnectionStopsPacketFlow/0 (2021 ms) [2/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AudioToVideoUpgrade/0 (1491 ms) [3/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.VideoFlowsAfterMediaSectionIsRejectedAndRecycled/0 (1301 ms) [4/452] TrackMediaInfoMapDeathTest.MultipleMultiSsrcReceiversPerTrack (5038 ms) [5/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithDtls10/0 (1095 ms) [6/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithSdes/0 (1332 ms) [7/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetCaptureStartNtpTimeWithOldStatsApi/0 (6164 ms) [8/452] RTCStatsIntegrationTest.GetStatsWithSenderSelector (997 ms) [9/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.BundlingEnabledWhileIceRestartOccurs/0 (1074 ms) [10/452] PeerConnectionIceTest/PeerConnectionIceTest.NoIceCandidatesBeforeSetLocalDescription/0 (1259 ms) [11/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.CallTransferredForCallee/0 (7927 ms) [12/452] RTCStatsIntegrationTest.GetsStatsWhileClosingPeerConnection (1197 ms) [13/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.CallerDtls10ToCalleeDtls12/0 (1269 ms) [14/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgrade/0 (1258 ms) [15/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaFlowsWhenCandidatesSetOnlyInSdp/0 (1585 ms) [16/452] PeerConnectionIntegrationTestPlanB.CanSendRemoteVideoTrack (1226 ms) [17/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.CodecNamesAreCaseInsensitive/0 (1062 ms) [18/452] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.CallWithSdesKeyNegotiation/0 (1272 ms) [19/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.DtmfSenderObserver/0 (1235 ms) [20/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.VideoRejectedInSubsequentOffer/0 (1212 ms) [21/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetBytesReceivedStatsWithOldStatsApi/0 (1016 ms) [22/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithSctpDataChannel/0 (1075 ms) [23/452] PeerConnectionIntegrationTestUnifiedPlan.MediaFlowsAfterEarlyWarmupWithAddTransceiver (907 ms) [24/452] PeerConnectionIntegrationTestUnifiedPlan.EndToEndCallWithTwoVideoTracksAndNoSignaledSsrc (946 ms) [25/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithGcmCipher/0 (1000 ms) [26/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationInteropTest.ReverseRolesOneAudioLocalToOneVideoRemote/0 (1565 ms) [27/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithTwoVideoTracks/0 (753 ms) [28/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportDataChannelUsesRtpBidirectionalVideo/0 (882 ms) [29/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithIceRenomination/0 (1039 ms) [30/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.Send1280By720ResolutionAndReceive16To9AspectRatio/0 (735 ms) [31/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.DisableAudioPlayoutStillGeneratesAudioStats/0 (713 ms) [32/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AddSctpDataChannelInSubsequentOffer/0 (1253 ms) [33/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationInteropTest.OneAudioOneVideoToNoMediaRemote/0 (1091 ms) [34/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.RtpDataChannelSignaledClosedInCalleeOffer/0 (764 ms) [35/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.Aes128Sha1_32_CipherNotUsedWhenOnlyCalleeSupported/0 (726 ms) [36/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.IceStatesReachCompletionWithRemoteHostname/0 (862 ms) [37/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.NonGcmCipherUsedWhenGcmNotSupported/0 (962 ms) [38/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.SctpDataChannelConfigSentToOtherSide/0 (831 ms) [39/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.NonGcmCipherUsedWhenOnlyCallerSupportsGcm/0 (770 ms) [40/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetAudioOutputLevelStatsWithOldStatsApi/0 (671 ms) [41/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.Dtls12CipherStatsAndUmaMetrics/0 (1068 ms) [42/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetMediaStatsForUnsignaledStreamWithNewStatsApi/0 (603 ms) [43/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/8 (547 ms) [44/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.VerifyBestConnection/2 (582 ms) [45/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.TurnCustomizerUsedForTurnConnections/0 (642 ms) [46/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.RtpDataChannelsRejectedByCallee/0 (850 ms) [47/452] PeerConnectionBundleTest/PeerConnectionBundleTest.TwoCandidatesForEachTransportWhenNoBundleNoRtcpMux/0 (575 ms) [48/452] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetRemoteAnswer/0 (352 ms) [49/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.SSLCertificateVerifierUsedForTurnConnections/0 (799 ms) [50/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportBidirectionalAudio/0 (445 ms) [51/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CreateAnswerWithDifferentSslRoles/0 (633 ms) [52/452] PeerConnectionIceTest/PeerConnectionIceTest.LaterAnswerHasSameIceCredentialsIfNoIceRestart/0 (422 ms) [53/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/8 (529 ms) [54/452] PeerConnectionJsepTest.AudioOnlyInitialOffer (258 ms) [55/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AnswererRejectsVideoSection/0 (564 ms) [56/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/4 (538 ms) [57/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.RotatedVideoWithoutCVOExtension/0 (628 ms) [58/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetStatsForUnsignaledStreamWithNewStatsApi/0 (508 ms) [59/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/4 (442 ms) [60/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.DataChannelWorksWhenSpecCompliantSctpOfferReceived/0 (481 ms) [61/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetSourcesVideo/0 (530 ms) [62/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/4 (464 ms) [63/452] PeerConnectionJsepTest.SetRemoteAnswerUpdatesCurrentDirection (380 ms) [64/452] PeerConnectionRtpTest/PeerConnectionRtpTest.RemoveTrackWithSharedStreamFiresOnRemoveTrack/0 (454 ms) [65/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/12 (483 ms) [66/452] PeerConnectionRtpTestUnifiedPlan.CreateOfferSameTrackLabel (256 ms) [67/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/8 (402 ms) [68/452] PeerConnectionIceTest/PeerConnectionIceUfragPwdAnswerTest.TestIncludedInAnswer/0 (517 ms) [69/452] PeerConnectionIceTest/PeerConnectionIceTest.DuplicateIceCandidateIgnoredWhenAdded/0 (389 ms) [70/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/24 (256 ms) [71/452] PeerConnectionBundleTest/PeerConnectionBundleMatrixTest.VerifyTransportsBeforeAndAfterSettingRemoteAnswer/2 (440 ms) [72/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.IceStateSetupFailure/4 (19292 ms) [73/452] PeerConnectionSignalingTest/PeerConnectionSignalingTest.InitiatorFlagSetOnCallerAndNotOnCallee/0 (412 ms) [74/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/16 (480 ms) [75/452] PeerConnectionBundleTest/PeerConnectionBundleMatrixTest.VerifyTransportsBeforeAndAfterSettingRemoteAnswer/6 (384 ms) [76/452] PeerConnectionBundleTest/PeerConnectionBundleTest.OneCandidateOnlyOnFirstTransportWhenBundleAndRtcpMux/0 (437 ms) [77/452] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.MediaTransportDataChannelCreatedEvenIfSctpAvailable/0 (264 ms) [78/452] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.CreateDataChannelBeforeNegotiate/0 (405 ms) [79/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/16 (387 ms) [80/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/8 (427 ms) [81/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/4 (279 ms) [82/452] PeerConnectionIceTest/PeerConnectionIceTest.CandidateInSubsequentOfferIsAddedToRemoteDescription/0 (447 ms) [83/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.IceEventsGeneratedAndLoggedInRtcEventLog/0 (425 ms) [84/452] PeerConnectionUsageHistogramTest.FingerprintAudioVideo (407 ms) [85/452] PeerConnectionBundleTest/PeerConnectionBundleTest.RemoveContentFromBundleGroup/0 (418 ms) [86/452] PeerConnectionJsepTest.CalleeDoesReoffer (440 ms) [87/452] PeerConnectionIceTest/PeerConnectionIceUfragPwdAnswerTest.TestIncludedInAnswer/4 (421 ms) [88/452] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingLocalRejectedAndNotRejectedRemote/0 (348 ms) [89/452] PeerConnectionMediaTest/PeerConnectionMediaTest.TestAVOfferWithAudioOnlyAnswer/0 (377 ms) [90/452] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.DataChannelFromOpenWorksWhilePreviousChannelClosing/0 (401 ms) [91/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/4 (378 ms) [92/452] PeerConnectionJsepTest/RecycleMediaSectionTest.CurrentLocalAndCurrentRemoteRejected/0 (389 ms) [93/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/0 (345 ms) [94/452] PeerConnectionIceTest/PeerConnectionIceTest.TwoTrickledCandidatesAddedToRemoteDescription/0 (420 ms) [95/452] PeerConnectionRtpTestUnifiedPlan.AddTransceiverCallsOnTrack (384 ms) [96/452] PeerConnectionIceTest/PeerConnectionIceTest.AddRemoveCandidateWithEmptyTransportDoesNotCrash/0 (398 ms) [97/452] PeerConnectionJsepTest.MediaSectionsInInitialOfferOrderedCorrectly (252 ms) [98/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/4 (435 ms) [99/452] PeerConnectionRtpTest/PeerConnectionRtpTest.RemoveTrackWithStreamRemovesReceiver/0 (375 ms) [100/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/8 (424 ms) [101/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.IceStateSetupFailure/0 (21420 ms) [102/452] PeerConnectionSignalingUnifiedPlanTest.DtlsTransportsAreSeparateeWhenUnbundled (516 ms) [103/452] PeerConnectionMediaTest/PeerConnectionMediaTest.CreateOfferWithNoVoiceActivityDetectionIncludesNoComfortNoiseCodecs/0 (246 ms) [104/452] PeerConnectionIceTest/PeerConnectionIceTest.OfferFromLiteIceControlledAndAnswerFromFullIceControlling/0 (417 ms) [105/452] PeerConnectionMediaTest/PeerConnectionMediaOfferDirectionTest.VerifyDirection/0 (252 ms) [106/452] PeerConnectionBundleTest/PeerConnectionBundleTest.NeverCreateRtcpTransportWithRtcpMuxRequired/0 (411 ms) [107/452] PeerConnectionJsepTest.AudioTrackAddedAfterDataSectionInReoffer (342 ms) [108/452] PeerConnectionIceTest/PeerConnectionIceTest.IceRestartOfferClearsExistingCandidate/0 (398 ms) [109/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/0 (251 ms) [110/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInAnswerWithSdesAndGcm/0 (377 ms) [111/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/4 (404 ms) [112/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/0 (369 ms) [113/452] PeerConnectionIceTest/PeerConnectionIceTest.CandidatesGeneratedForEachLocalInterface/0 (407 ms) [114/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/12 (388 ms) [115/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/8 (404 ms) [116/452] NumberOfSendEncodings/PeerConnectionSimulcastMetricsTests.NumberOfSendEncodingsIsLogged/0 (253 ms) [117/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.ExchangeOfferAnswerWhenSdesOn/0 (400 ms) [118/452] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.MessageTransferBetweenTwoPairsOfDataChannels/0 (443 ms) [119/452] PeerConnectionUsageHistogramTest.NoNotableUsageOnEventFiringAfterClose (363 ms) [120/452] PeerConnectionBundleTest/PeerConnectionBundleTest.TransportsSameForMaxBundleWithBundleInRemoteOffer/0 (405 ms) [121/452] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingRemoteRejectedAndNotRejectedLocal/0 (354 ms) [122/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.MediaSectionCountEnforcedForSubsequentOffer/0 (386 ms) [123/452] PeerConnectionJsepTest/RecycleMediaSectionTest.CurrentLocalOnlyRejected/0 (384 ms) [124/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/12 (375 ms) [125/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.VerifyIceStates/2 (22875 ms) [126/452] PeerConnectionMediaTest/PeerConnectionMediaTest.AnswerHasSameMidsAsOffer/0 (382 ms) [127/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/24 (314 ms) [128/452] PeerConnectionFactoryTest.CreatePCUsingIceServers (253 ms) [129/452] PeerConnectionBundleTest/PeerConnectionBundleTest.BundleOnFirstMidInAnswer/0 (430 ms) [130/452] PeerConnectionIceTest/PeerConnectionIceTest.OfferContainsGatheredCandidates/0 (500 ms) [131/452] PeerConnectionJsepTest.CreateAnswerHasSameMidsAsOffer (437 ms) [132/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInOfferWhenDtlsDisabled/0 (221 ms) [133/452] PeerConnectionJsepTest/RecycleMediaSectionTest.CurrentRemoteOnlyRejected/0 (435 ms) [134/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.ExchangeOfferAnswerWhenNoEncryption/0 (371 ms) [135/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/12 (371 ms) [136/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/4 (355 ms) [137/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/16 (483 ms) [138/452] PeerConnectionJsepTest.LegacyNoMidAudioVideoAnswer (417 ms) [139/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportDataChannelCalleeCloses/0 (448 ms) [140/452] PeerConnectionMediaTest/PeerConnectionMediaTest.FailToSetRemoteDescriptionIfCreateMediaChannelFails/0 (427 ms) [141/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetRemoteAnswerWithNoCryptosWhenSdesOn/0 (426 ms) [142/452] PeerConnectionSimulcastTests.SimulcastLayersAreSetInSender (441 ms) [143/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/8 (483 ms) [144/452] PeerConnectionSimulcastMetricsTests.RejectedSimulcastIsLogged (608 ms) [145/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CloseAndTestStreamsAndStates/0 (467 ms) [146/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CnameGenerationInAnswer/0 (594 ms) [147/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/16 (371 ms) [148/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetLocalAnswerWithNoFingerprintWhenDtlsOn/0 (417 ms) [149/452] PeerConnectionJsepTest.CreateOfferDoesNotRecycleMediaSectionIfFirstStopped (472 ms) [150/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/8 (654 ms) [151/452] PeerConnectionRtpTestUnifiedPlan.ApplyTwoRemoteOffersWithNoAnswerResultsInOneAddTrackEvent (347 ms) [152/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.GetRTCStatsBeforeAndAfterCalling/0 (505 ms) [153/452] PeerConnectionInterfaceTestPlanB.SdpWithEmptyMsidAndSsrcCreatesDefaultStreamId (427 ms) [154/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/20 (299 ms) [155/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/12 (490 ms) [156/452] SdpFormatReceivedTest.ComplexUnifiedIsReportedAsComplexUnifiedPlan (566 ms) [157/452] PeerConnectionSignalingTest/PeerConnectionSignalingTest.SetRemoteOfferTwiceWorks/0 (330 ms) [158/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/12 (362 ms) [159/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/16 (324 ms) [160/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/16 (325 ms) [161/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/0 (418 ms) [162/452] PeerConnectionJsepTest.NoMsidInAnswerIfNoRespondingTracks (388 ms) [163/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/8 (449 ms) [164/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/4 (434 ms) [165/452] PeerConnectionMediaTestPlanB.EmptyLocalAnswerRemovesSendStreams (496 ms) [166/452] PeerConnectionBundleTest/PeerConnectionBundleTest.RejectDescriptionChangingBundleTag/0 (392 ms) [167/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationNotCausingIceRestart/0 (394 ms) [168/452] PeerConnectionRtpTest/PeerConnectionRtpTest.RemoveTrackWithoutStreamFiresOnRemoveTrack/0 (456 ms) [169/452] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingLocalRejectedAndNoRemote/0 (408 ms) [170/452] PeerConnectionRtpTest/PeerConnectionRtpTest.AddTrackWithStreamAddsReceiver/0 (458 ms) [171/452] PeerConnectionMediaTestUnifiedPlan.StoppedLocalTransceiversRemovesMediaChannels (490 ms) [172/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.GetStatsForVideoTrack/0 (617 ms) [173/452] PeerConnectionMediaTest/PeerConnectionMediaTest.SetRemoteDescriptionFailsWithDuplicateMids/0 (347 ms) [174/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateSubsequentRecvOnlyOffer/0 (426 ms) [175/452] PeerConnectionJsepTest.SetRemoteOfferWithOneTrackTwoStreamFiresOnAddTrack (329 ms) [176/452] PeerConnectionRtpTest/PeerConnectionRtpTest.AudioGetParametersHasHeaderExtensions/0 (440 ms) [177/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/16 (440 ms) [178/452] PeerConnectionIceTest/PeerConnectionIceTest.CanSetRemoteSessionDescriptionWithRemoteCandidates/0 (378 ms) [179/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/36 (332 ms) [180/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/0 (378 ms) [181/452] PeerConnectionJsepTest.RtpReceiverCreatedBeforeSetRemoteDescriptionKeepsId (318 ms) [182/452] PeerConnectionMsidSignalingTest.PureUnifiedPlanToUs (506 ms) [183/452] PeerConnectionInterfaceTestPlanB.RenegotiateAudioOnly (523 ms) [184/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferReceivePrAnswerAndAnswer/0 (476 ms) [185/452] PeerConnectionIceTest/PeerConnectionIceTest.IceCredentialsCreateOffer/0 (333 ms) [186/452] PeerConnectionSimulcastTests.ServerSendsOfferToReceiveSimulcast (346 ms) [187/452] PeerConnectionBundleTest/PeerConnectionBundleTest.AddContentToBundleGroupInAnswerNotSupported/0 (361 ms) [188/452] PeerConnectionMediaTest/PeerConnectionMediaTest.NewStreamInLocalAnswerAddsSendStreams/0 (503 ms) [189/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.DtlsSdesFallbackNotSupported/0 (303 ms) [190/452] PeerConnectionRtpTest/PeerConnectionRtpTest.LegacyObserverOnSuccess/0 (305 ms) [191/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.ReceiveOfferCreatePrAnswerAndAnswer/0 (415 ms) [192/452] PeerConnectionSignalingTest/PeerConnectionSignalingTest.SessionVersionIncrementedInSubsequentDifferentOffer/0 (446 ms) [193/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/0 (414 ms) [194/452] PeerConnectionJsepTest.SetLocalOfferSetsTransceiverMid (320 ms) [195/452] PeerConnectionJsepTest.OneVideoUnifiedPlanToTwoVideoPlanBFails (370 ms) [196/452] PeerConnectionSimulcastTests.NegotiationDoesNotHaveRidExtension (555 ms) [197/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/16 (475 ms) [198/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/24 (444 ms) [199/452] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingRemoteRejectedAndNoLocal/0 (475 ms) [200/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/8 (299 ms) [201/452] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.SctpContentAndTransportNameSetCorrectly/0 (600 ms) [202/452] PeerConnectionRtpTest/PeerConnectionRtpTest.AddTrackWithoutStreamFiresOnAddTrack/0 (600 ms) [203/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationCreatesPooledSessionCorrectly/0 (423 ms) [204/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/0 (376 ms) [205/452] PeerConnectionMediaConfigTest.TestDisablePrerendererSmoothingTrue (235 ms) [206/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.DataChannelCloseWhenPeerConnectionClose/0 (455 ms) [207/452] PeerConnectionIceTest/PeerConnectionIceTest.SetRemoteDescriptionFailsIfNoIceCredentials/0 (486 ms) [208/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/28 (500 ms) [209/452] PeerConnectionJsepTest.SetRemoteOfferDoesNotReuseTransceiverFromAddTransceiver (329 ms) [210/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInOfferWhenDtlsEnabled/0 (362 ms) [211/452] PeerConnectionRtpTest/PeerConnectionRtpTest.CreateTwoSendersWithSameTrack/0 (331 ms) [212/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.UpdateRemoteStreams/0 (346 ms) [213/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferWithOfferToReceiveConstraints/0 (268 ms) [214/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/16 (283 ms) [215/452] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetRemoteAnswer/4 (345 ms) [216/452] PeerConnectionMediaTest/PeerConnectionMediaTest.AudioVideoOfferAnswerCreateSendRecvStreams/0 (345 ms) [217/452] PeerConnectionMediaTest/PeerConnectionMediaOfferDirectionTest.VerifyDirection/8 (220 ms) [218/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateCurrentLessThanMinFails/0 (237 ms) [219/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/0 (300 ms) [220/452] PeerConnectionRtpTestUnifiedPlan.UnsignaledSsrcCreatesReceiverStreams (316 ms) [221/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/0 (411 ms) [222/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/12 (489 ms) [223/452] PeerConnectionRtpTestUnifiedPlan.RemoveTrackClearsSenderTrack (299 ms) [224/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInOfferWhenEncryptionDisabled/0 (282 ms) [225/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreatePeerConnectionAppliesNetworkConfigToPortAllocator/0 (473 ms) [226/452] WebRtcSdpTest.DeserializeSessionDescriptionWithoutMline (213 ms) [227/452] RTCStatsCollectorTest.DoNotCrashWhenGetStatsCalledDuringCallback (229 ms) [228/452] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.MediaTransportWithoutSdesFails/0 (260 ms) [229/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferWithAudioVideoOptions/0 (276 ms) [230/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.TestReceiveOnlyDataChannel/0 (511 ms) [231/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferWithIceRestart/0 (480 ms) [232/452] PeerConnectionIntegrationTest/PeerConnectionIntegrationInteropTest.NoMediaLocalToNoMediaRemote/0 (387 ms) [233/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.TestRejectSctpDataChannelInAnswer/0 (254 ms) [234/452] PeerConnectionMediaTest/PeerConnectionMediaTest.MediaTransportForMediaAndDataChannels/0 (286 ms) [235/452] PeerConnectionIceTest/PeerConnectionIceTest.CannotAddCandidateWhenRemoteDescriptionNotSet/0 (286 ms) [236/452] PeerConnectionRtpTestUnifiedPlan.SendEncodingsPassedToSender (286 ms) [237/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.TestSendBinaryOnRtpDataChannel/0 (316 ms) [238/452] PeerConnectionIceTest/PeerConnectionIceTest.LocalCandidatesRemovedWhenNetworkDownIfGatheringContinually/0 (506 ms) [239/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferWithVideoOnlyOptions/0 (315 ms) [240/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.OnAddTrackCallback/0 (426 ms) [241/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/16 (282 ms) [242/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/12 (266 ms) [243/452] PeerConnectionBundleTest/PeerConnectionBundleTest.FailToSetDescriptionWithBundleAndNoRtcpMux/0 (328 ms) [244/452] PeerConnectionSimulcastTests.MustSupplyAllOrNoRidsInSimulcast (418 ms) [245/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.GetConfigurationAfterSetConfiguration/0 (248 ms) [246/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetIceRegatherIntervalRangeWithContinualGathering/0 (266 ms) [247/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.IceRenominationNotOffered/0 (266 ms) [248/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetLocalOfferWithNoFingerprintWhenDtlsOn/0 (342 ms) [249/452] PeerConnectionInterfaceTestPlanB.CreateAnswerWithOfferToReceiveConstraints (326 ms) [250/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.StartAndStopLoggingToOutputAfterPeerConnectionClosed/0 (296 ms) [251/452] HasStream/StatsCollectorTrackTest.TrackObjectExistsWithoutUpdateStats/1 (205 ms) [252/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/20 (253 ms) [253/452] SctpDataChannelTest.OpenAckSentIfCreatedFromOpenMessage (206 ms) [254/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.ExtraPooledSessionsDiscardedAfterApplyingAnswer/0 (319 ms) [255/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.RecvonlyDescriptionDoesntCreateStream/0 (282 ms) [256/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/36 (315 ms) [257/452] PeerConnectionMediaTest/PeerConnectionMediaTest.MediaEngineErrorPropagatedToClients/0 (361 ms) [258/452] RTCStatsCollectorTest.SingleCallback (203 ms) [259/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/12 (313 ms) [260/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/16 (299 ms) [261/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SsrcInOfferAnswer/0 (329 ms) [262/452] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.CreateAnswerWithRemoteSctpDataChannelIncludesDataSection/0 (315 ms) [263/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreatePeerConnectionWithDifferentIceTransportsTypes/0 (268 ms) [264/452] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetLocalAnswer/0 (440 ms) [265/452] PeerConnectionInterfaceTestPlanB.SdpWithMsidDontCreatesDefaultStream (266 ms) [266/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferFailsWithInvalidOfferToReceiveAudio/0 (330 ms) [267/452] PeerConnectionInterfaceTestPlanB.SdpWithoutMsidCreatesDefaultStream (315 ms) [268/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationChangesPruneTurnPortsFlag/0 (298 ms) [269/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CurrentAndPendingDescriptions/0 (377 ms) [270/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/12 (534 ms) [271/452] WebRtcSdpTest.SerializeSessionDescriptionSpecialMsid (250 ms) [272/452] WebRtcSdpTest.DeserializeSessionDescriptionWithIceOptions (187 ms) [273/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateMaxLessThanCurrentFails/0 (250 ms) [274/452] PeerConnectionInterfaceTestPlanB.CreateSenderWithStream (315 ms) [275/452] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetLocalAnswer/4 (330 ms) [276/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/20 (361 ms) [277/452] SctpDataChannelTest.OpenMessageSent (190 ms) [278/452] PeerConnectionUsageHistogramTest.FingerprintStunTurnInReconfiguration (222 ms) [279/452] SctpDataChannelTest.ClosedWhenReceivedBufferFull (253 ms) [280/452] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.SctpPortPropagatedFromSdpToTransport/0 (440 ms) [281/452] RtpSenderReceiverTest.VideoSenderCantSetPerSenderEncodingParameters (203 ms) [282/452] WebRtcSdpTest.DeserializeSdpWithSctpDataChannels (203 ms) [283/452] DisableSimulcastLayersInSender/RtpSenderReceiverTest.DisableSimulcastLayersWithoutMediaEngine/0 (203 ms) [284/452] PeerConnectionRtpTestUnifiedPlan.AddVideoTrackCreatesVideoSender (361 ms) [285/452] RtpSenderReceiverTest.SetVideoBitratePriority (189 ms) [286/452] PeerConnectionFactoryTest.CreatePCUsingSecureTurnUrl (236 ms) [287/452] PeerConnectionMediaTest/PeerConnectionMediaOfferDirectionTest.VerifyDirection/4 (218 ms) [288/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.RtpDuplicatedLabelNotAllowed/0 (266 ms) [289/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/8 (345 ms) [290/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetRemoteOfferWithNoCryptosWhenSdesOn/0 (376 ms) [291/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationChangesIceServers/0 (266 ms) [292/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/12 (329 ms) [293/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/4 (329 ms) [294/452] PeerConnectionCryptoTest/PeerConnectionCryptoTest.RTCConfigurationCryptoOptionOverridesFactory/0 (447 ms) [295/452] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.NoSctpTransportCreatedIfRtpDataChannelEnabled/0 (430 ms) [296/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.AddTrackWithoutStream/0 (322 ms) [297/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SctpDuplicatedLabelAllowed/0 (258 ms) [298/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateSctpDataChannelShouldFailForInvalidConfig/0 (305 ms) [299/452] PeerConnectionInterfaceTestPlanB.RemoveStream (289 ms) [300/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/0 (305 ms) [301/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateWithoutMinSucceeds/0 (269 ms) [302/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateMaxNegativeFails/0 (267 ms) [303/452] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/4 (285 ms) [304/452] PeerConnectionRtpTestUnifiedPlan.AddAudioTransceiverCreatesVideoSenderAndReceiver (252 ms) [305/452] RtpParametersConversionTest.ToVideoRtpCodecCapability (204 ms) [306/452] RTCStatsCollectorTest.CollectLocalRTCMediaStreamStatsAndRTCMediaStreamTrackStats_Audio (235 ms) [307/452] PeerConnectionMediaTest/PeerConnectionMediaTest.MediaTransportPropagatedToVoiceEngine/0 (314 ms) [308/452] RtpSenderReceiverTest.AudioSenderCanSetParameters (203 ms) [309/452] JsepSessionDescriptionTest.SerializeSessionDescriptionWithBothProtocols (187 ms) [310/452] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.CreateDataChannelWithSctpDisabledFails/0 (236 ms) [311/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.AttachmentIdIsSetOnAddTrack/0 (253 ms) [312/452] PeerConnectionSignalingTest/PeerConnectionSignalingTest.FailToSetNullRemoteDescription/0 (253 ms) [313/452] RtpParametersConversionTest.ToCricketStreamParamsVecWithRtx (214 ms) [314/452] SctpUtilsTest.WriteParseOpenMessageWithOrderedReliable (194 ms) [315/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationReturnsInvalidModificationError/0 (366 ms) [316/452] RtpSenderReceiverTest.VideoSenderInitParametersMovedAfterManualSimulcastAndNegotiation (194 ms) [317/452] WebRtcSdpTest.SerializeMediaContentDescriptionWithExtmapAllowMixed (207 ms) [318/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/28 (205 ms) [319/452] WebRtcSdpTest.TestDeserializeSimulcastAttributeRemovesUnknownRids (205 ms) [320/452] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/1 (259 ms) [321/452] PeerConnectionRtpTestUnifiedPlan.RemoveTrackErrorIfClosed (282 ms) [322/452] PeerConnectionRtpTestUnifiedPlan.NoRenegotiationNeededAfterTransceiverSetSameDirection (252 ms) [323/452] PeerConnectionRtpTestUnifiedPlan.AddTrackChangesDirectionFromRecvOnlyToSendRecv (299 ms) [324/452] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/0 (299 ms) [325/452] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationReturnsSyntaxErrorFromBadIceUrls/0 (253 ms) [326/452] RtpSenderReceiverTest.TestOnDestroyedSignal (268 ms) [327/452] RidDescriptionSdpSerializerTest.Deserialize_AmbiguousCase (190 ms) [328/452] RtpSenderReceiverTest.VideoSenderCantSetUnimplementedEncodingParameters (203 ms) [329/452] WebRtcSdpTest.SerializeSessionDescriptionWithIceOptions (202 ms) [330/452] WebRtcSdpTest.DeserializeSdpWithSctpDataChannelsAndBandwidth (202 ms) [331/452] SimulcastSdpSerializerTest.Deserialize_OnlySendStreams (218 ms) [332/452] WebRtcSdpTest.DeserializeSdpWithConferenceFlag (203 ms) [333/452] Encrypted/WebRtcSdpExtmapTest.DeserializeSessionDescriptionWithInvalidExtmap/1 (203 ms) [334/452] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithSendSet/4 (280 ms) [335/452] FakeAdmTest.DuplexTest (188 ms) [336/452] WebRtcSdpTest.DeserializeCandidate (188 ms) [337/452] WebRtcSdpTest.DeserializeBundleOnlyAttribute (201 ms) [338/452] RtpSenderReceiverTest.VideoSenderDoesNotHaveDtmfSender (248 ms) [339/452] RtpSenderReceiverTest.RemoteAudioTrackSetVolume (186 ms) [340/452] WebRtcSdpTest.DeserializeSdpWithReorderedPltypes (233 ms) [341/452] RidDescriptionSdpSerializerTest.Serialize_OnlyDirectionSend (187 ms) [342/452] WebRtcSdpTest.SerializeAudioFmtpWithKnownFmtpParameter (201 ms) [343/452] RtpTransceiverDirectionTest/EnumerateAllDirectionsTest.TestReversedSwapped/0 (188 ms) [344/452] SctpDataChannelTest.BufferedAmountWhenBlocked (188 ms) [345/452] RTCStatsCollectorTest.GetStatsWithNullSenderSelector (218 ms) [346/452] WebRtcSdpTest.ParseConnectionDataWithHostnameConnectionAddress (203 ms) [347/452] NumberOfSendEncodings/PeerConnectionSimulcastMetricsTests.NumberOfSendEncodingsIsLogged/4 (327 ms) [348/452] PeerConnectionMediaTest/PeerConnectionMediaTest.OfferHasDifferentDirectionForAudioVideo/0 (392 ms) [349/452] ProxyTest.WorkerMethod1 (252 ms) [350/452] MediaStreamTest.ChangeAudioTrack (220 ms) [351/452] RtpSenderReceiverTest.VideoSenderDetectTransactionIdModification (251 ms) [352/452] RTCStatsCollectorTest.CollectRTCInboundRTPStreamStats_Audio (266 ms) [353/452] SctpSidAllocatorTest.SctpIdAllocationBasedOnRole (202 ms) [354/452] WebRtcSdpTest.TestDeserializeRemovesRidsWithInvalidCodec (217 ms) [355/452] PeerConnectionFactoryTest.CheckRtpSenderDataCapabilities (438 ms) [356/452] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/17 (254 ms) [357/452] DtmfSenderTest.InsertEmptyTonesToCancelPreviousTask (207 ms) [358/452] PeerConnectionRtpTestUnifiedPlan.AddTransceiverTwiceWithSameTrackCreatesMultipleTransceivers (537 ms) [359/452] JsepSessionDescriptionTest.AddCandidateAlreadyHasUfrag (239 ms) [360/452] JsepSessionDescriptionTest.SerializeDeserialize (207 ms) [361/452] PlayoutLatencyTest.Rounding (207 ms) [362/452] SignalingProxyTest.ConstMethod0 (207 ms) [363/452] RTCStatsCollectorTest.CollectRTCTransportStats (249 ms) [364/452] RtpParametersConversionTest.ToAudioCodec (187 ms) [365/452] RtpSenderReceiverTest.VideoSenderEarlyWarmupSsrcThenTrack (187 ms) [366/452] RtpSenderReceiverTest.AudioSenderTrackSetToNull (187 ms) [367/452] RtpSenderReceiverTest.AudioSenderMustCallGetParametersBeforeSetParameters (202 ms) [368/452] RtpSenderReceiverTest.AudioSenderSetParametersOldValueFail (203 ms) [369/452] RtpSenderReceiverTest.AudioReceiverCannotSetFrameDecryptorAfterStop (203 ms) [370/452] RidDescriptionSdpSerializerTest.Deserialize_SimpleReceiveCase (202 ms) [371/452] TrackMediaInfoMapTest.SsrcLookupFunction (202 ms) [372/452] VideoTrackTest.DisableTrackBlackout (203 ms) [373/452] WebRtcSdpTest.DeserializeSessionDescriptionWithoutMsid (203 ms) [374/452] WebRtcSdpTest.DeserializeVideoFmtp (187 ms) [375/452] WebRtcSdpTest.SerializeUnifiedPlanSessionDescriptionNoSsrcSignaling (187 ms) [376/452] WebRtcSdpTest.DeserializeSessionDescriptionWithoutRtpmapButWithFmtp (264 ms) [377/452] WebRtcSdpTest.DeserializeEmptySessionName (203 ms) [378/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/32 (187 ms) [379/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/28 (187 ms) [380/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/40 (202 ms) [381/452] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithRecvSet/4 (233 ms) [382/452] DtmfSenderTest.InsertDtmf (265 ms) [383/452] IceServerParsingTest.TurnServerPrioritiesUnique (187 ms) [384/452] RtpSenderReceiverTest.AudioSenderWithoutTrackAndSsrc (203 ms) [385/452] WebRtcSdpTest.SerializeSessionDescriptionWithSctpDataChannel (203 ms) [386/452] SimulcastDeserializationErrors/SimulcastSdpSerializerTest.SimulcastDeserializationFailed/4 (375 ms) [387/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/44 (202 ms) [388/452] WebRtcSdpTest.DeserializeSctpPortInVideoDescription (250 ms) [389/452] HasStream/StatsCollectorTrackTest.TransportObjectLinkedFromSsrcObject/1 (203 ms) [390/452] HasStream/StatsCollectorTrackTest.FilterOutNegativeInitialValues/1 (203 ms) [391/452] RTCStatsCollectorTest.CollectRTCCertificateStatsSingle (202 ms) [392/452] SctpDataChannelTest.SendUnorderedAfterReceiveData (264 ms) [393/452] RtpParametersConversionTest.ToVideoCodecInvalidParameters (203 ms) [394/452] RTCStatsTraversalTest.MultipleStarts (187 ms) [395/452] RTCStatsCollectorTest.CollectRTCDataChannelStats (280 ms) [396/452] SctpDataChannelTest.ReceiveDataWithInvalidSsrc (264 ms) [397/452] RTCStatsTraversalTest.BogusReference (187 ms) [398/452] WebRtcSdpTest.IceCredentialsInCandidateStringIgnored (266 ms) [399/452] PeerConnectionRtpTestUnifiedPlan.AddTrackReusesTransceiverOfType (250 ms) [400/452] StatsCollectorTest.NoTransport (203 ms) [401/452] HasStream/StatsCollectorTrackTest.GetStatsFromRemoteStream/1 (219 ms) [402/452] DtmfSenderTest.InsertDtmfSendsAfterWait (203 ms) [403/452] IceServerParsingTest.ParseTransport (187 ms) [404/452] ProxyTest.WorkerThreadDestructor (218 ms) [405/452] RtpParametersConversionTest.ToCricketCodecsDuplicatePayloadType (203 ms) [406/452] PlayoutLatencyTest.DefaultValue (266 ms) [407/452] RtpParametersConversionTest.ToCricketStreamParamsVecMultipleEncodings (187 ms) [408/452] RtpParametersConversionTest.ToVideoRtpCodecParameters (202 ms) [409/452] RtpSenderReceiverTest.AddAndDestroyAudioRtpSender (250 ms) [410/452] RtpSenderReceiverTest.RemoteAudioTrackDisable (217 ms) [411/452] RtpSenderReceiverTest.AddAndDestroyAudioRtpReceiverWithStreams (266 ms) [412/452] RtpSenderReceiverTest.VideoReceiverCannotSetFrameDecryptorAfterStop (233 ms) [413/452] RtpSenderReceiverTest.PropagatesVideoTrackContentHint (281 ms) [414/452] SctpUtilsTest.TestIsOpenMessage (234 ms) [415/452] RtpSenderReceiverTest.SetAudioBitratePriority (296 ms) [416/452] SimulcastSdpSerializerTest.Serialize_SimpleCase (264 ms) [417/452] RidDescriptionSdpSerializerTest.Serialize_AttributeList (218 ms) [418/452] RidDescriptionSdpSerializerTest.Deserialize_WithMultipleRestrictions (202 ms) [419/452] StatsCollectorTest.OnlyOneSessionObjectExists (203 ms) [420/452] TrackMediaInfoMapTest.SingleMultiSsrcSenderPerTrack (202 ms) [421/452] WebRtcSdpTest.SerializeSessionDescriptionWithFingerprintNoCryptos (202 ms) [422/452] WebRtcSdpTest.SerializeHostnameCandidate (187 ms) [423/452] WebRtcSdpTest.DeserializeSdpWithCorruptedSctpDataChannels (187 ms) [424/452] WebRtcSdpTest.DeserializeSessionDescriptionWithInactiveContent (202 ms) [425/452] WebRtcSdpTest.SerializeSessionDescriptionWithAudioRejected (342 ms) [426/452] WebRtcSdpTest.SerializePlanBSessionDescription (217 ms) [427/452] WebRtcSdpTest.DeserializeMsidAttributeWithMissingStreamId (250 ms) [428/452] WebRtcSdpTest.UnifiedPlanHasMediaSectionMsidSignaling (265 ms) [429/452] WebRtcSdpTest.ParseMediaTransportMalformedLine (201 ms) [430/452] WebRtcSdpTest.ParseMediaTransport (233 ms) [431/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/40 (219 ms) [432/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/32 (203 ms) [433/452] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/44 (187 ms) [434/452] RtpTransceiverDirectionTest/EnumerateAllDirectionsTest.TestIdentity/0 (202 ms) [435/452] RtpTransceiverDirectionTest/EnumerateAllDirectionsTest.TestReversedIdentity/0 (187 ms) [436/452] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithSendSet/0 (187 ms) [437/452] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithRecvSet/0 (187 ms) [438/452] DisableSimulcastLayersInSender/RtpSenderReceiverTest.DisableSimulcastLayersWithMediaEngine/0 (201 ms) [439/452] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/5 (219 ms) [440/452] SimulcastDeserializationErrors/SimulcastSdpSerializerTest.SimulcastDeserializationFailed/8 (311 ms) [441/452] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/9 (202 ms) [442/452] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/13 (202 ms) [443/452] HasStream/StatsCollectorTrackTest.RemoteSsrcInfoIsPresent/1 (187 ms) [444/452] HasStream/StatsCollectorTrackTest.AudioBandwidthEstimationInfoIsReported/1 (202 ms) [445/452] HasStream/StatsCollectorTrackTest.LocalAndRemoteTracksWithSameSsrc/1 (233 ms) [446/452] HasStream/StatsCollectorTrackTest.VerifyVideoSendSsrcStats/1 (233 ms) [447/452] LocalAudioSourceTest.InitWithNoOptions (203 ms) [448/452] SimulcastDeserializationErrors/SimulcastSdpSerializerTest.SimulcastDeserializationFailed/0 (187 ms) [449/452] Encrypted/WebRtcSdpExtmapTest.DeserializeSessionDescriptionWithSessionLevelExtmap/1 (265 ms) [450/452] WebRtcSdpTest.SerializeMediaTransportSettingsTestCopy (249 ms) [451/452] WebRtcSdpTest.RoundTripSdpWithSctpDataChannelsWithCandidates (297 ms) [452/452] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.CreateDataChannelLargeTransfer/0 (56473 ms) 43bb19004982ad10: exit 0 Additional test environment: CHROME_DEVEL_SANDBOX=/opt/chromium/chrome_sandbox CHROME_HEADLESS=1 LANG=en_US.UTF-8 Command: e:\b\s\w\ir\.swarming_module_cache\vpython\37d31f\Scripts\python.exe ..\..\tools_webrtc\gtest-parallel-wrapper.py --output_dir=e:\b\s\w\ioyfnq1x\test_logs --gtest_color=no --timeout=900 --retry_failed=3 .\peerconnection_unittests.exe --isolated-script-test-output=e:\b\s\w\ioyfnq1x\output.json --isolated-script-test-perf-output=e:\b\s\w\ioyfnq1x\perftest-output.json gtest-parallel-wrapper: Executing command e:\b\s\w\ir\.swarming_module_cache\vpython\37d31f\Scripts\python.exe e:\b\s\w\ir\third_party\gtest-parallel\gtest-parallel --shard_index=2 --shard_count=4 --output_dir=e:\b\s\w\ioyfnq1x\test_logs --retry_failed=3 --gtest_color=no --timeout=900 --dump_json_test_results=e:\b\s\w\ioyfnq1x\output.json .\peerconnection_unittests.exe -- --isolated-script-test-perf-output=e:\b\s\w\ioyfnq1x\perftest-output.json [0/451] Running tests... [1/451] IceServerParsingTest.ParseRejectsUsername (437 ms) [2/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.ClosingConnectionStopsPacketFlow/1 (2434 ms) [3/451] PeerConnectionIceTest/PeerConnectionIceTest.NoIceCandidatesBeforeSetLocalDescription/1 (1656 ms) [4/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetCaptureStartNtpTimeWithOldStatsApi/1 (7694 ms) [5/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.CallTransferredForCallee/1 (7957 ms) [6/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.BundlingEnabledWhileIceRestartOccurs/1 (3952 ms) [7/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.DtmfSenderObserver/1 (1865 ms) [8/451] PeerConnectionIntegrationTestUnifiedPlan.ReOfferWithStoppedBundleTaggedTransceiver (1909 ms) [9/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AudioToVideoUpgrade/1 (2276 ms) [10/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.VideoFlowsAfterMediaSectionIsRejectedAndRecycled/1 (3069 ms) [11/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithSctpDataChannel/1 (2503 ms) [12/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.VideoRejectedInSubsequentOffer/1 (3105 ms) [13/451] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.CallWithSdesKeyNegotiation/1 (1991 ms) [14/451] RTCStatsIntegrationTest.GetStatsWithReceiverSelector (1914 ms) [15/451] RTCStatsIntegrationTest.GetStatsReferencedIds (2464 ms) [16/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithDtls10/1 (1907 ms) [17/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithGcmCipher/1 (2117 ms) [18/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.CallerDtls10ToCalleeDtls12/1 (1773 ms) [19/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportDataChannelUsesRtpBidirectionalVideo/1 (1174 ms) [20/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithIceRenomination/1 (1628 ms) [21/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithTwoVideoTracks/1 (1761 ms) [22/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithSdes/1 (1544 ms) [23/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.CodecNamesAreCaseInsensitive/1 (1825 ms) [24/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationInteropTest.OneAudioOneVideoToNoMediaRemote/1 (1421 ms) [25/451] PeerConnectionUsageHistogramTest.FingerprintWithNoHostCandidates (20643 ms) [26/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetBytesReceivedStatsWithOldStatsApi/1 (1573 ms) [27/451] PeerConnectionMediaTest/PeerConnectionMediaTest.NewStreamInLocalAnswerAddsSendStreams/1 (729 ms) [28/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationInteropTest.ReverseRolesOneAudioLocalToOneVideoRemote/1 (1841 ms) [29/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.IceStateSetupFailure/5 (21792 ms) [30/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaFlowsWhenCandidatesSetOnlyInSdp/1 (1875 ms) [31/451] PeerConnectionIceTest/PeerConnectionIceTest.CandidatesGeneratedForEachLocalInterface/1 (953 ms) [32/451] PeerConnectionUsageHistogramTest.FingerprintWithPrivateIP (764 ms) [33/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.Send1280By720ResolutionAndReceive16To9AspectRatio/1 (1965 ms) [34/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetAudioOutputLevelStatsWithOldStatsApi/1 (1485 ms) [35/451] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingRemoteRejectedAndNotRejectedLocal/1 (875 ms) [36/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.DisableAudioPlayoutStillGeneratesAudioStats/1 (1174 ms) [37/451] PeerConnectionRtpTest/PeerConnectionRtpTest.RemoveTrackWithStreamRemovesReceiver/1 (1029 ms) [38/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.IceStateSetupFailure/1 (24990 ms) [39/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.RotatedVideoWithoutCVOExtension/1 (1585 ms) [40/451] PeerConnectionBundleTest/PeerConnectionBundleTest.TwoCandidatesForEachTransportWhenNoBundleNoRtcpMux/1 (1410 ms) [41/451] PeerConnectionJsepTest.SettingTransceiverInactiveDoesNotStopIt (823 ms) [42/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetMediaStatsForUnsignaledStreamWithNewStatsApi/1 (1324 ms) [43/451] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingLocalRejectedAndNotRejectedRemote/1 (859 ms) [44/451] PeerConnectionBundleTest/PeerConnectionBundleTest.OneCandidateOnlyOnFirstTransportWhenBundleAndRtcpMux/1 (904 ms) [45/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetSourcesVideo/1 (1119 ms) [46/451] PeerConnectionRtpTest/PeerConnectionRtpTest.RemoveTrackWithoutStreamFiresOnRemoveTrack/1 (836 ms) [47/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/9 (894 ms) [48/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CnameGenerationInAnswer/1 (745 ms) [49/451] PeerConnectionIceTest/PeerConnectionIceTest.OfferContainsGatheredCandidates/1 (744 ms) [50/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.ReceiveOfferCreatePrAnswerAndAnswer/1 (766 ms) [51/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.TestReceiveOnlyDataChannel/1 (657 ms) [52/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/13 (778 ms) [53/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.SSLCertificateVerifierUsedForTurnConnections/1 (1286 ms) [54/451] PeerConnectionMediaTest/PeerConnectionMediaTest.AudioVideoOfferAnswerCreateSendRecvStreams/1 (703 ms) [55/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/9 (796 ms) [56/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/5 (741 ms) [57/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/1 (824 ms) [58/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/17 (812 ms) [59/451] PeerConnectionBundleTestUnifiedPlan.EmptyBundleGroupCreatedInAnswerWhenAppropriate (732 ms) [60/451] PeerConnectionSimulcastMetricsTests.LegacySimulcastIsLogged (746 ms) [61/451] PeerConnectionIceTest/PeerConnectionIceTest.IceRestartOfferClearsExistingCandidate/1 (763 ms) [62/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/5 (845 ms) [63/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/17 (719 ms) [64/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.GetRTCStatsBeforeAndAfterCalling/1 (799 ms) [65/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AnswererRejectsVideoSection/1 (1469 ms) [66/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/17 (820 ms) [67/451] PeerConnectionIntegrationTestPlanB.RemoveAndAddTrackWithNewStreamId (1194 ms) [68/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CreateAnswerWithDifferentSslRoles/1 (1476 ms) [69/451] PeerConnectionJsepTest.LegacyNoMidAudioOnlyOffer (724 ms) [70/451] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.SctpContentAndTransportNameSetCorrectly/1 (666 ms) [71/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/29 (804 ms) [72/451] PeerConnectionMediaTest/PeerConnectionMediaTest.MediaTransportPropagatedToVoiceEngine/1 (627 ms) [73/451] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetRemoteAnswer/5 (626 ms) [74/451] PeerConnectionInterfaceTestPlanB.LocalDescriptionChanged (961 ms) [75/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/5 (974 ms) [76/451] PeerConnectionSimulcastMetricsTests.NoSimulcastUsageIsLogged (1228 ms) [77/451] PeerConnectionRtpTest/PeerConnectionRtpTest.AudioGetParametersHasHeaderExtensions/1 (1004 ms) [78/451] PeerConnectionRtpTestUnifiedPlan.ApplyRemoteOfferAddThenRemoteOfferRemoveResultsInOneRemoveTrackEvent (1042 ms) [79/451] PeerConnectionBundleTest/PeerConnectionBundleTest.AddContentToBundleGroupInAnswerNotSupported/1 (908 ms) [80/451] PeerConnectionMsidSignalingTest.UnifiedPlanTalkingToOurself (1042 ms) [81/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AddSctpDataChannelInSubsequentOffer/1 (3127 ms) [82/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgrade/1 (12536 ms) Error: unrecognized flag --isolated-script-test-perf-output=e:\b\s\w\ioyfnq1x\perftest-output.json Note: Google Test filter = PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgrade/1 [==========] Running 1 test from 1 test suite. [----------] Global test environment set-up. [----------] 1 test from PeerConnectionIntegrationTest/PeerConnectionIntegrationTest [ RUN ] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgrade/1 [000:000] (loopback_media_transport.cc:156): LoopbackMediaTransport [000:000] (loopback_media_transport.cc:156): LoopbackMediaTransport [000:003] (audio_processing_impl.cc:421): Capture analyzer activated: 0 Capture post processor activated: 0 Render pre processor activated: 0 [000:006] (webrtc_video_engine.cc:460): WebRtcVideoEngine::WebRtcVideoEngine() [000:006] (webrtc_voice_engine.cc:194): WebRtcVoiceEngine::WebRtcVoiceEngine [000:007] (webrtc_voice_engine.cc:217): WebRtcVoiceEngine::Init [000:008] (webrtc_voice_engine.cc:224): Supported send codecs in order of preference: [000:012] (webrtc_voice_engine.cc:227): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:012] (webrtc_voice_engine.cc:227): ISAC/16000/1 (103) [000:012] (webrtc_voice_engine.cc:227): ISAC/32000/1 (104) [000:012] (webrtc_voice_engine.cc:227): G722/8000/1 (9) [000:012] (webrtc_voice_engine.cc:227): ILBC/8000/1 (102) [000:012] (webrtc_voice_engine.cc:227): PCMU/8000/1 (0) [000:012] (webrtc_voice_engine.cc:227): PCMA/8000/1 (8) [000:012] (webrtc_voice_engine.cc:227): CN/32000/1 (106) [000:012] (webrtc_voice_engine.cc:227): CN/16000/1 (105) [000:012] (webrtc_voice_engine.cc:227): CN/8000/1 (13) [000:013] (webrtc_voice_engine.cc:227): telephone-event/48000/1 (110) [000:013] (webrtc_voice_engine.cc:227): telephone-event/32000/1 (112) [000:013] (webrtc_voice_engine.cc:227): telephone-event/16000/1 (113) [000:013] (webrtc_voice_engine.cc:227): telephone-event/8000/1 (126) [000:013] (webrtc_voice_engine.cc:230): Supported recv codecs in order of preference: [000:016] (webrtc_voice_engine.cc:233): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:016] (webrtc_voice_engine.cc:233): ISAC/16000/1 (103) [000:016] (webrtc_voice_engine.cc:233): ISAC/32000/1 (104) [000:016] (webrtc_voice_engine.cc:233): G722/8000/1 (9) [000:016] (webrtc_voice_engine.cc:233): ILBC/8000/1 (102) [000:016] (webrtc_voice_engine.cc:233): PCMU/8000/1 (0) [000:016] (webrtc_voice_engine.cc:233): PCMA/8000/1 (8) [000:016] (webrtc_voice_engine.cc:233): CN/32000/1 (106) [000:016] (webrtc_voice_engine.cc:233): CN/16000/1 (105) [000:016] (webrtc_voice_engine.cc:233): CN/8000/1 (13) [000:016] (webrtc_voice_engine.cc:233): telephone-event/48000/1 (110) [000:017] (webrtc_voice_engine.cc:233): telephone-event/32000/1 (112) [000:018] (webrtc_voice_engine.cc:233): telephone-event/16000/1 (113) [000:018] (webrtc_voice_engine.cc:233): telephone-event/8000/1 (126) [000:018] (webrtc_voice_engine.cc:310): 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, } [000:020] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [000:021] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [000:021] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [000:021] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [000:021] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [000:021] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [000:022] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:023] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:023] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:023] (apm_helpers.cc:90): Echo control set to 1 with mode 0 [000:023] (apm_helpers.cc:79): AGC set to 1 with mode 0 [000:023] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:024] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:024] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:024] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:024] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:024] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:027] (apm_helpers.cc:104): NS set to 1 [000:027] (webrtc_voice_engine.cc:466): Stereo swapping enabled? 0 [000:027] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:027] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:028] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:028] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:028] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:028] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:028] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:028] (webrtc_voice_engine.cc:536): Typing detection is enabled? 1 [000:028] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:028] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:029] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:033] (rtc_event_log_impl.cc:72): Creating legacy encoder for RTC event log. [000:033] (peer_connection_factory.cc:395): Using default network controller factory [000:034] (bitrate_prober.cc:62): Bandwidth probing enabled, set to inactive [000:034] (paced_sender.cc:427): ProcessThreadAttached 0x2162430 [000:035] (cpu_info.cc:49): Available number of cores: 8 [000:035] (aimd_rate_control.cc:101): Using aimd rate control with back off factor 0.85 [000:035] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating. [000:043] (audio_processing_impl.cc:421): Capture analyzer activated: 0 Capture post processor activated: 0 Render pre processor activated: 0 [000:045] (webrtc_video_engine.cc:460): WebRtcVideoEngine::WebRtcVideoEngine() [000:045] (webrtc_voice_engine.cc:194): WebRtcVoiceEngine::WebRtcVoiceEngine [000:048] (webrtc_voice_engine.cc:217): WebRtcVoiceEngine::Init [000:048] (webrtc_voice_engine.cc:224): Supported send codecs in order of preference: [000:051] (webrtc_voice_engine.cc:227): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:052] (webrtc_voice_engine.cc:227): ISAC/16000/1 (103) [000:052] (webrtc_voice_engine.cc:227): ISAC/32000/1 (104) [000:052] (webrtc_voice_engine.cc:227): G722/8000/1 (9) [000:052] (webrtc_voice_engine.cc:227): ILBC/8000/1 (102) [000:052] (webrtc_voice_engine.cc:227): PCMU/8000/1 (0) [000:052] (webrtc_voice_engine.cc:227): PCMA/8000/1 (8) [000:052] (webrtc_voice_engine.cc:227): CN/32000/1 (106) [000:052] (webrtc_voice_engine.cc:227): CN/16000/1 (105) [000:052] (webrtc_voice_engine.cc:227): CN/8000/1 (13) [000:052] (webrtc_voice_engine.cc:227): telephone-event/48000/1 (110) [000:052] (webrtc_voice_engine.cc:227): telephone-event/32000/1 (112) [000:053] (webrtc_voice_engine.cc:227): telephone-event/16000/1 (113) [000:053] (webrtc_voice_engine.cc:227): telephone-event/8000/1 (126) [000:053] (webrtc_voice_engine.cc:230): Supported recv codecs in order of preference: [000:055] (webrtc_voice_engine.cc:233): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:055] (webrtc_voice_engine.cc:233): ISAC/16000/1 (103) [000:056] (webrtc_voice_engine.cc:233): ISAC/32000/1 (104) [000:056] (webrtc_voice_engine.cc:233): G722/8000/1 (9) [000:056] (webrtc_voice_engine.cc:233): ILBC/8000/1 (102) [000:056] (webrtc_voice_engine.cc:233): PCMU/8000/1 (0) [000:056] (webrtc_voice_engine.cc:233): PCMA/8000/1 (8) [000:056] (webrtc_voice_engine.cc:233): CN/32000/1 (106) [000:056] (webrtc_voice_engine.cc:233): CN/16000/1 (105) [000:056] (webrtc_voice_engine.cc:233): CN/8000/1 (13) [000:056] (webrtc_voice_engine.cc:233): telephone-event/48000/1 (110) [000:056] (webrtc_voice_engine.cc:233): telephone-event/32000/1 (112) [000:056] (webrtc_voice_engine.cc:233): telephone-event/16000/1 (113) [000:056] (webrtc_voice_engine.cc:233): telephone-event/8000/1 (126) [000:057] (webrtc_voice_engine.cc:310): 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, } [000:059] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [000:059] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [000:059] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [000:059] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [000:059] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [000:060] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [000:061] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:061] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:061] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:062] (apm_helpers.cc:90): Echo control set to 1 with mode 0 [000:062] (apm_helpers.cc:79): AGC set to 1 with mode 0 [000:062] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:062] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:062] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:062] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:063] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:063] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:063] (apm_helpers.cc:104): NS set to 1 [000:063] (webrtc_voice_engine.cc:466): Stereo swapping enabled? 0 [000:063] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:063] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:063] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:063] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:063] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:063] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:063] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:063] (webrtc_voice_engine.cc:536): Typing detection is enabled? 1 [000:064] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:064] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:064] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:105] (rtc_event_log_impl.cc:72): Creating legacy encoder for RTC event log. [000:106] (peer_connection_factory.cc:395): Using default network controller factory [000:106] (bitrate_prober.cc:62): Bandwidth probing enabled, set to inactive [000:107] (paced_sender.cc:427): ProcessThreadAttached 0x21960e0 [000:108] (aimd_rate_control.cc:101): Using aimd rate control with back off factor 0.85 [000:108] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating. [000:123] (peer_connection.cc:1381): Adding video transceiver in response to a call to AddTrack. [000:127] (peer_connection.cc:1381): Adding video transceiver in response to a call to AddTrack. [000:134] (jsep_transport_controller.cc:1518): Generating media transport offer! [000:135] (media_session.cc:335): Duplicate id found. Reassigning from 102 to 127 [000:135] (media_session.cc:335): Duplicate id found. Reassigning from 104 to 125 [000:136] (media_session.cc:335): Duplicate id found. Reassigning from 103 to 124 [000:137] (media_session.cc:335): Duplicate id found. Reassigning from 1 to 14 [000:137] (media_session.cc:335): Duplicate id found. Reassigning from 2 to 13 [000:137] (media_session.cc:335): Duplicate id found. Reassigning from 3 to 12 [000:137] (media_session.cc:335): Duplicate id found. Reassigning from 5 to 11 [000:141] (peer_connection_integrationtest.cc:753): Caller: SetLocalDescriptionAndSendSdpMessage [000:158] (jsep_transport_controller.cc:975): Offered media transport has now been activated. [000:158] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [000:158] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [000:158] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [000:158] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [000:158] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [000:158] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [000:158] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [000:159] (p2p_transport_channel.cc:432): Set ICE ufrag: 9/nF pwd: JPe0Xsr41ONhzjqKzrsLZ6cb on transport 0 [000:162] (webrtc_video_engine.cc:472): CreateMediaChannel. Options: VideoOptions {} [000:163] (channel.cc:144): Created channel for 0 [000:168] (channel.cc:212): BaseChannel::Init_w, media_transport=1 [000:168] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:168] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:169] (peer_connection.cc:4097): Session: 6675921963715691949 Old state: kStable New state: kHaveLocalOffer [000:169] (channel.cc:1009): Setting local video description [000:172] (webrtc_video_engine.cc:945): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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}]} [000:174] (webrtc_video_engine.cc:960): Changing recv codecs from {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} to {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} [000:175] (webrtc_video_engine.cc:1061): AddSendStream: {id:01d823f2-43cd-4e53-88de-27eebee9d1b9;ssrcs:[653648138,1380348883];ssrc_groups:{semantics:FID;ssrcs:[653648138,1380348883]};cname:vWjbMZWRqzqeHiad;stream_ids:;} [000:176] (webrtc_video_engine.cc:1100): SetLocalSsrc on all the receive streams because we added a send stream. [000:176] (channel.cc:676): Add send stream ssrc: 653648138 [000:176] (channel.cc:996): Changing video state, send=0 [000:177] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 653648138, options: VideoOptions {is_screencast : false, }, source = (source)) [000:178] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [000:178] (basic_port_allocator.cc:851): Network manager has started [000:178] (peer_connection_integrationtest.cc:683): Callee: HandleIncomingOffer [000:178] (basic_port_allocator.cc:766): Allocate ports on 1 networks [000:179] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Udp [000:179] (port.cc:318): Port[2fca080::1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port created with network cost 50 [000:179] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:179] (basic_port_allocator.cc:875): Adding allocated port for 0 [000:180] (basic_port_allocator.cc:894): Port[2fca080:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Added port to allocator [000:180] (basic_port_allocator.cc:912): Port[2fca080:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Gathered candidate: Cand[:2214029314:1:udp:2122260223:192.168.1.1:49152:local::0:9/nF:JPe0Xsr41ONhzjqKzrsLZ6cb:1:50:0] [000:180] (basic_port_allocator.cc:940): Port[2fca080:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port ready. [000:181] (basic_port_allocator.cc:1017): Port[2fca080:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port completed gathering candidates. [000:193] (peer_connection_integrationtest.cc:771): Callee: SetRemoteDescription [000:198] (jsep_transport_controller.cc:996): Returning new, client media transport. [000:199] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [000:199] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [000:199] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [000:199] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [000:199] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [000:199] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [000:199] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [000:200] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=9/nF, renomination disabled [000:202] (webrtc_video_engine.cc:472): CreateMediaChannel. Options: VideoOptions {} [000:204] (channel.cc:144): Created channel for 0 [000:217] (channel.cc:212): BaseChannel::Init_w, media_transport=1 [000:217] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:217] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:218] (peer_connection.cc:4097): Session: 3265430737459529171 Old state: kStable New state: kHaveRemoteOffer [000:218] (channel.cc:1061): Setting remote video description [000:219] (webrtc_video_engine.cc:670): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 0} [000:221] (webrtc_video_engine.cc:679): Using codec: VideoCodec[96:VP8] [000:221] (webrtc_video_engine.cc:747): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. [000:222] (webrtc_video_engine.cc:1163): AddRecvStream: {id:01d823f2-43cd-4e53-88de-27eebee9d1b9;ssrcs:[653648138,1380348883];ssrc_groups:{semantics:FID;ssrcs:[653648138,1380348883]};cname:vWjbMZWRqzqeHiad;stream_ids:;} [000:226] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 653648138, 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: 104, red_type: 102, rtx_ssrc: 1380348883, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 103 (pt) -> 102 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, target_delay_ms: 0} [000:226] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:226] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:227] (bitrate_allocator.cc:115): Current BWE 300000 [000:230] (channel.cc:719): Add remote ssrc: 653648138 [000:231] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Relay [000:233] (channel.cc:996): Changing video state, send=0 [000:233] (peer_connection.cc:2687): Processing the MSIDs for MID=0 (streams=[]). [000:234] (peer_connection.cc:2698): Processing the addition of a remote track for MID=0. [000:234] (webrtc_video_engine.cc:1293): SetSink: ssrc:653648138 (ptr) [000:237] (peer_connection_integrationtest.cc:919): Caller: OnIceCandidate [000:237] (peer_connection_integrationtest.cc:867): Callee: ReceiveIceMessage [000:238] (peer_connection.cc:6141): 0 is not ready to use the remote candidate because the local or remote description is not set. [000:241] (peer_connection.cc:5449): Local and Remote descriptions must be applied to get the SSL Role of the session. [000:249] (peer_connection_integrationtest.cc:753): Callee: SetLocalDescriptionAndSendSdpMessage [000:267] (webrtc_video_engine.cc:2685): VideoReceiveStream stats: 520373, {ssrc: 653648138, 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, discarded: 0, sync_offset_ms: 2147483647, cum_loss: 0, max_ext_seq: 0, nack: 0, fir: 0, pli: 0} [000:268] (webrtc_video_engine.cc:1338): Call stats: 520372, {send_bw_bps: 300000, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1} [000:292] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [000:292] (p2p_transport_channel.cc:432): Set ICE ufrag: I6DM pwd: /A6Y7CvF3KEQXswBCqf8iPzP on transport 0 [000:292] (dtls_transport.cc:253): DtlsTransport[0|1|__]: Other side didn't support DTLS. [000:298] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv. [000:300] (channel.cc:553): Channel enabled [000:300] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Tcp [000:300] (port.cc:318): Port[302b6e0::1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port created with network cost 50 [000:300] (basic_port_allocator.cc:875): Adding allocated port for 0 [000:301] (basic_port_allocator.cc:894): Port[302b6e0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Added port to allocator [000:301] (channel.cc:996): Changing video state, send=0 [000:301] (basic_port_allocator.cc:912): Port[302b6e0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Gathered candidate: Cand[:3446969586:1:tcp:1518280447:192.168.1.1:49153:local::0:9/nF:JPe0Xsr41ONhzjqKzrsLZ6cb:1:50:0] [000:301] (basic_port_allocator.cc:940): Port[302b6e0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port ready. [000:302] (basic_port_allocator.cc:1017): Port[302b6e0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port completed gathering candidates. [000:302] (basic_port_allocator.cc:1119): All candidates gathered for 0:1:0 [000:302] (p2p_transport_channel.cc:854): P2PTransportChannel: 0, component 1 gathering complete [000:302] (peer_connection.cc:4097): Session: 3265430737459529171 Old state: kHaveRemoteOffer New state: kStable [000:303] (channel.cc:1009): Setting local video description [000:306] (webrtc_video_engine.cc:945): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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}]} [000:308] (webrtc_video_engine.cc:960): Changing recv codecs from {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} to {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} [000:308] (webrtc_video_engine.cc:2493): MaybeRecreateWebRtcFlexfecStream (recv) because of SetRecvParameters [000:308] (webrtc_video_engine.cc:2498): RecreateWebRtcVideoStream (recv) because of SetRecvParameters [000:308] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:308] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:309] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 653648138, 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: 104, red_type: 102, rtx_ssrc: 1380348883, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 103 (pt) -> 102 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, target_delay_ms: 0} [000:317] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [000:319] (receive_statistics_proxy.cc:487): Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [000:321] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 653648138, 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: 125, red_type: 127, rtx_ssrc: 1380348883, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [000:321] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:321] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:349] (webrtc_video_engine.cc:1061): AddSendStream: {id:df6f0198-e6ab-4e4e-987c-0ed4ba12587e;ssrcs:[3350319844,3987717279];ssrc_groups:{semantics:FID;ssrcs:[3350319844,3987717279]};cname:vBzBF3+EwbdtI2fQ;stream_ids:;} [000:350] (webrtc_video_engine.cc:1859): RecreateWebRtcStream (send) because of SetCodec. [000:351] (alr_experiment.cc:66): 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 [000:358] (video_send_stream_impl.cc:249): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [3350319844], 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: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [3987717279], payload_type: 97}, c_name: vBzBF3+EwbdtI2fQ}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [000:358] (video_send_stream_impl.cc:275): ERROR: Initial encoder max bitrate = -1 which is <= 0! [000:358] (video_send_stream_impl.cc:277): Using default encoder max bitrate = 10 Mbps [000:360] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:360] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:360] (video_stream_encoder.cc:579): ConfigureEncoder requested. [000:361] (video_send_stream.cc:160): VideoSendStream::Stop [000:362] (webrtc_video_engine.cc:1100): SetLocalSsrc on all the receive streams because we added a send stream. [000:362] (webrtc_video_engine.cc:2434): RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc=3350319844 [000:362] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:362] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:362] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 653648138, 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: 125, red_type: 127, rtx_ssrc: 1380348883, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [000:362] (video_send_stream_impl.cc:415): VideoSendStream::Stop [000:363] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [000:364] (receive_statistics_proxy.cc:487): Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [000:367] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 653648138, local_ssrc: 3350319844, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 1380348883, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [000:368] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:368] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:374] (channel.cc:676): Add send stream ssrc: 3350319844 [000:375] (video_send_stream.cc:160): VideoSendStream::Stop [000:375] (channel.cc:996): Changing video state, send=0 [000:375] (video_send_stream_impl.cc:415): VideoSendStream::Stop [000:375] (message_queue.cc:518): Message took 72ms to dispatch. Posted from: cricket::BaseChannel::SetLocalContent@../../pc/channel.cc:293 [000:377] (peer_connection.cc:4033): Changing IceConnectionState 0 => 1 [000:378] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 3350319844, options: VideoOptions {is_screencast : false, }, source = (source)) [000:378] (video_stream_encoder.cc:579): ConfigureEncoder requested. [000:383] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [000:383] (basic_port_allocator.cc:851): Network manager has started [000:383] (basic_port_allocator.cc:766): Allocate ports on 1 networks [000:383] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Udp [000:384] (port.cc:318): Port[3022bc0::1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port created with network cost 50 [000:384] (basic_port_allocator.cc:875): Adding allocated port for 0 [000:384] (basic_port_allocator.cc:894): Port[3022bc0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Added port to allocator [000:385] (basic_port_allocator.cc:912): Port[3022bc0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Gathered candidate: Cand[:2214029314:1:udp:2122260223:192.168.1.1:49154:local::0:I6DM:/A6Y7CvF3KEQXswBCqf8iPzP:1:50:0] [000:385] (basic_port_allocator.cc:940): Port[3022bc0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port ready. [000:385] (port.cc:1126): Conn[3082220:0:Net[test0:192.168.1.0/24:Unknown:id=1]:+lDigL6B:1:0:local:udp:192.168.1.1:49154->3XXG8RB8:1:2122260223:local:udp:192.168.1.1:49152|C--W|-|0|0|9115038255631187454|-]: Connection created [000:385] (p2p_transport_channel.cc:1264): Channel[0|1|__]: Created connection with origin: 2, total: 1 [000:386] (p2p_transport_channel.cc:1914): Channel[0|1|__]: Transport channel state changed from 0 to 2 [000:386] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:386] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:386] (p2p_transport_channel.cc:1476): Channel[0|1|__]: Have a pingable connection for the first time; starting to ping. [000:386] (basic_port_allocator.cc:1017): Port[3022bc0:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port completed gathering candidates. [000:387] (port.cc:1756): Conn[3082220:0:Net[test0:192.168.1.0/24:Unknown:id=1]:+lDigL6B:1:0:local:udp:192.168.1.1:49154->3XXG8RB8:1:2122260223:local:udp:192.168.1.1:49152|C--W|-|0|0|9115038255631187454|-]: Sent STUN ping, id=2b765741765a5348767a5045, use_candidate=0, nomination=0 [000:388] (port.cc:528): Received STUN ping id=2b765741765a5348767a5045 from unknown address 192.168.1.1:49154 [000:388] (port.cc:1126): Conn[3082c90:0:Net[test0:192.168.1.0/24:Unknown:id=1]:2jikRuox:1:0:local:udp:192.168.1.1:49152->ApLrIaWh:1:1853824767:prflx:udp:192.168.1.1:49154|C--W|-|0|0|7962116751024340479|-]: Connection created [000:388] (p2p_transport_channel.cc:988): Adding connection from peer reflexive candidate: Cand[:2389647568:1:udp:1853824767:192.168.1.1:49154:prflx::0:I6DM::1:50:0] [000:389] (port.cc:853): Port[2fca080:0:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Sent STUN ping response, to=192.168.1.1:49154, id=2b765741765a5348767a5045 [000:391] (p2p_transport_channel.cc:1914): Channel[0|1|R_]: Transport channel state changed from 0 to 2 [000:392] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:392] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:393] (port.cc:1704): Conn[3082220:0:Net[test0:192.168.1.0/24:Unknown:id=1]:+lDigL6B:1:0:local:udp:192.168.1.1:49154->3XXG8RB8:1:2122260223:local:udp:192.168.1.1:49152|C--I|-|0|0|9115038255631187454|-]: Received STUN ping response, id=2b765741765a5348767a5045, code=0, rtt=6, pings_since_last_response=2b765741765a5348767a5045 [000:394] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed [000:394] (p2p_transport_channel.cc:1852): Channel[0|1|__]: New selected connection: Conn[3082220:0:Net[test0:192.168.1.0/24:Unknown:id=1]:+lDigL6B:1:0:local:udp:192.168.1.1:49154->3XXG8RB8:1:2122260223:local:udp:192.168.1.1:49152|CRWS|S|0|0|9115038255631187454|6] [000:394] (jsep_transport_controller.cc:1214): Transport 0 writability changed to 1. [000:394] (channel.cc:583): Channel writable (0) for the first time [000:395] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:395] (basic_port_allocator.cc:1119): All candidates gathered for 0:1:0 [000:395] (p2p_transport_channel.cc:854): P2PTransportChannel: 0, component 1 gathering complete [000:396] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:396] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:397] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:397] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:397] (peer_connection_integrationtest.cc:703): Caller: HandleIncomingAnswer [000:397] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps [000:398] (video_stream_encoder.cc:1477): Video suspend state changed to: not suspended [000:398] (channel.cc:996): Changing video state, send=1 [000:398] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:406] (peer_connection_integrationtest.cc:771): Caller: SetRemoteDescription [000:414] (webrtc_video_engine.cc:1338): Call stats: 520519, {send_bw_bps: 0, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1} [000:420] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [000:422] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=I6DM, renomination disabled [000:422] (dtls_transport.cc:253): DtlsTransport[0|1|R_]: Other side didn't support DTLS. [000:423] (p2p_transport_channel.cc:1476): Channel[0|1|R_]: Have a pingable connection for the first time; starting to ping. [000:423] (p2p_transport_channel.cc:2440): Selecting connection for triggered check: Conn[3082c90:0:Net[test0:192.168.1.0/24:Unknown:id=1]:2jikRuox:1:0:local:udp:192.168.1.1:49152->ApLrIaWh:1:1853824767:prflx:udp:192.168.1.1:49154|CR-W|-|0|0|7962116751024340479|-] [000:423] (port.cc:1756): Conn[3082c90:0:Net[test0:192.168.1.0/24:Unknown:id=1]:2jikRuox:1:0:local:udp:192.168.1.1:49152->ApLrIaWh:1:1853824767:prflx:udp:192.168.1.1:49154|CR-W|-|0|0|7962116751024340479|-]: Sent STUN ping, id=594974685135743877547071, use_candidate=1, nomination=0 [000:424] (port.cc:1704): Conn[3082c90:0:Net[test0:192.168.1.0/24:Unknown:id=1]:2jikRuox:1:0:local:udp:192.168.1.1:49152->ApLrIaWh:1:1853824767:prflx:udp:192.168.1.1:49154|CR-I|-|0|0|7962116751024340479|-]: Received STUN ping response, id=594974685135743877547071, code=0, rtt=1, pings_since_last_response=594974685135743877547071 [000:427] (video_stream_encoder.cc:1051): Video frame parameters changed: dimensions=640x480, texture=0. [000:428] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed [000:428] (p2p_transport_channel.cc:1852): Channel[0|1|R_]: New selected connection: Conn[3082c90:0:Net[test0:192.168.1.0/24:Unknown:id=1]:2jikRuox:1:0:local:udp:192.168.1.1:49152->ApLrIaWh:1:1853824767:prflx:udp:192.168.1.1:49154|CRWS|S|0|0|7962116751024340479|1] [000:428] (jsep_transport_controller.cc:1214): Transport 0 writability changed to 1. [000:429] (channel.cc:583): Channel writable (0) for the first time [000:429] (jsep_transport_controller.cc:1277): 0 Transport 1 state changed. Check if state is complete. [000:429] (video_stream_encoder.cc:233): Set max framerate: 60 [000:430] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:430] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:445] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 30000bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 1700000bps [000:467] (channel.cc:996): Changing video state, send=0 [000:469] (channel.cc:553): Channel enabled [000:472] (channel.cc:996): Changing video state, send=0 [000:477] (peer_connection.cc:4097): Session: 6675921963715691949 Old state: kHaveLocalOffer New state: kStable [000:482] (channel.cc:1061): Setting remote video description [000:483] (webrtc_video_engine.cc:670): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 0} [000:486] (webrtc_video_engine.cc:679): Using codec: VideoCodec[96:VP8] [000:486] (webrtc_video_engine.cc:1859): RecreateWebRtcStream (send) because of SetCodec. [000:487] (bitrate_allocator.cc:115): Current BWE 300000 [000:487] (alr_experiment.cc:66): 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 [000:674] (video_send_stream_impl.cc:249): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [653648138], 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: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [1380348883], payload_type: 97}, c_name: vWjbMZWRqzqeHiad}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [000:674] (video_send_stream_impl.cc:275): ERROR: Initial encoder max bitrate = -1 which is <= 0! [000:674] (video_send_stream_impl.cc:277): Using default encoder max bitrate = 10 Mbps [000:687] (video_stream_encoder.cc:579): ConfigureEncoder requested. [000:687] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:687] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:687] (video_send_stream.cc:160): VideoSendStream::Stop [000:688] (webrtc_video_engine.cc:747): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. [000:688] (webrtc_video_engine.cc:1163): AddRecvStream: {id:df6f0198-e6ab-4e4e-987c-0ed4ba12587e;ssrcs:[3350319844,3987717279];ssrc_groups:{semantics:FID;ssrcs:[3350319844,3987717279]};cname:vBzBF3+EwbdtI2fQ;stream_ids:;} [000:690] (video_send_stream_impl.cc:415): VideoSendStream::Stop [000:691] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 3350319844, local_ssrc: 653648138, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 3987717279, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [000:691] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:691] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:700] (channel.cc:719): Add remote ssrc: 3350319844 [000:704] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:704] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:705] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps [000:705] (video_stream_encoder.cc:1477): Video suspend state changed to: not suspended [000:705] (channel.cc:996): Changing video state, send=1 [000:705] (message_queue.cc:518): Message took 223ms to dispatch. Posted from: cricket::BaseChannel::SetRemoteContent@../../pc/channel.cc:302 [000:705] (peer_connection.cc:4033): Changing IceConnectionState 0 => 1 [000:705] (peer_connection.cc:2687): Processing the MSIDs for MID=0 (streams=[]). [000:706] (peer_connection.cc:2698): Processing the addition of a remote track for MID=0. [000:706] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv. [000:706] (webrtc_video_engine.cc:1293): SetSink: ssrc:3350319844 (ptr) [000:708] (peer_connection_integrationtest.cc:919): Caller: OnIceCandidate [000:708] (peer_connection_integrationtest.cc:867): Callee: ReceiveIceMessage [000:720] (peer_connection_integrationtest.cc:919): Callee: OnIceCandidate [000:721] (peer_connection_integrationtest.cc:867): Caller: ReceiveIceMessage [000:722] (video_stream_encoder.cc:1051): Video frame parameters changed: dimensions=640x480, texture=0. [000:722] (video_stream_encoder.cc:233): Set max framerate: 60 [000:740] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 30000bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 1700000bps [000:775] (message_queue.cc:518): Message took 55ms to dispatch. Posted from: webrtc::JsepTransportController::OnTransportCandidateGathered_n@../../pc/jsep_transport_controller.cc:1244 [000:777] (decoder_database.cc:140): Initializing decoder with payload type '96'. [000:847] (peer_connection.cc:1381): Adding audio transceiver in response to a call to AddTrack. [000:849] (peer_connection.cc:1381): Adding audio transceiver in response to a call to AddTrack. [000:866] (jsep_transport_controller.cc:1513): Not regenerating media transport for the new offer in existing session. [000:874] (peer_connection.cc:5981): Changing to ICE completed state because all transports are complete. [000:874] (peer_connection.cc:4033): Changing IceConnectionState 1 => 2 [000:874] (peer_connection.cc:4033): Changing IceConnectionState 2 => 3 [000:876] (peer_connection.cc:5975): Changing to ICE connected state because all transports are writable. [000:876] (peer_connection.cc:4033): Changing IceConnectionState 1 => 2 [000:877] (peer_connection_integrationtest.cc:753): Caller: SetLocalDescriptionAndSendSdpMessage [000:900] (p2p_transport_channel.cc:432): Set ICE ufrag: 9/nF pwd: JPe0Xsr41ONhzjqKzrsLZ6cb on transport 0 [000:904] (webrtc_voice_engine.cc:1488): 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, } [000:904] (webrtc_voice_engine.cc:310): 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, } [000:905] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:905] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:905] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:905] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:905] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:905] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:905] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:905] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:906] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:906] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:906] (webrtc_voice_engine.cc:1506): 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, } [000:906] (channel.cc:144): Created channel for 1 [000:906] (channel.cc:583): Channel writable (1) for the first time [000:957] (channel.cc:212): BaseChannel::Init_w, media_transport=1 [000:957] (message_queue.cc:518): Message took 53ms to dispatch. Posted from: cricket::ChannelManager::CreateVoiceChannel@../../pc/channel_manager.cc:169 [000:957] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:957] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:958] (peer_connection.cc:4097): Session: 6675921963715691949 Old state: kStable New state: kHaveLocalOffer [000:958] (channel.cc:863): Changing voice state, recv=0 send=0 [000:958] (channel.cc:1009): Setting local video description [000:964] (webrtc_video_engine.cc:945): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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}]} [000:977] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:978] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:978] (channel.cc:996): Changing video state, send=1 [000:987] (channel.cc:871): Setting local voice description [000:988] (webrtc_voice_engine.cc:1327): 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}]} [000:988] (webrtc_voice_engine.cc:1516): Setting receive voice codecs. [000:990] (webrtc_voice_engine.cc:1787): AddSendStream: {id:183cfbe2-7712-4d89-be37-e6bf530e22fa;ssrcs:[1340491237];ssrc_groups:;cname:vWjbMZWRqzqeHiad;stream_ids:;} [001:001] (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 [001:001] (audio_coding_module.cc:333): Created [001:001] (channel_send.cc:704): Setting media_transport_ rate observers. [001:001] (audio_send_stream.cc:142): AudioSendStream: 1340491237 [001:001] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1340491237, extmap-allow-mixed: false, extensions: [], c_name: vWjbMZWRqzqeHiad}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: <unset>} [001:002] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [001:002] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [001:002] (channel.cc:676): Add send stream ssrc: 1340491237 [001:002] (channel_send.cc:816): Trying to call unset encoder. [001:007] (channel.cc:863): Changing voice state, recv=0 send=0 [001:010] (webrtc_voice_engine.cc:1488): Setting voice channel options: AudioOptions {hf: 0, } [001:010] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {hf: 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, } [001:010] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [001:010] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [001:010] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [001:010] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [001:010] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [001:011] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [001:011] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [001:011] (audio_processing_impl.cc:696): Highpass filter activated: 0 [001:012] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [001:012] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [001:012] (webrtc_voice_engine.cc:1506): Set voice channel options. Current options: AudioOptions {hf: 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, } [001:014] (peer_connection_integrationtest.cc:683): Callee: HandleIncomingOffer [001:015] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [001:025] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [001:031] (peer_connection_integrationtest.cc:771): Callee: SetRemoteDescription [001:058] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=9/nF, renomination disabled [001:065] (webrtc_voice_engine.cc:1488): 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, } [001:065] (webrtc_voice_engine.cc:310): 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, } [001:066] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [001:066] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [001:066] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [001:066] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [001:066] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [001:066] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [001:066] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [001:066] (audio_processing_impl.cc:696): Highpass filter activated: 1 [001:067] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [001:067] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [001:067] (webrtc_voice_engine.cc:1506): 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, } [001:067] (channel.cc:144): Created channel for 1 [001:067] (channel.cc:583): Channel writable (1) for the first time [001:071] (channel.cc:212): BaseChannel::Init_w, media_transport=1 [001:071] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [001:072] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [001:072] (peer_connection.cc:4097): Session: 3265430737459529171 Old state: kStable New state: kHaveRemoteOffer [001:073] (channel.cc:863): Changing voice state, recv=0 send=0 [001:073] (channel.cc:1061): Setting remote video description [001:074] (webrtc_video_engine.cc:670): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 0} [001:079] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [001:079] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [001:079] (channel.cc:996): Changing video state, send=1 [001:080] (channel.cc:923): Setting remote voice description [001:081] (webrtc_voice_engine.cc:1282): 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: 1, options: AudioOptions {}} [001:081] (webrtc_voice_engine.cc:1705): Recreate all the receive streams because the send codec has changed. [001:081] (webrtc_voice_engine.cc:2166): WebRtcVoiceMediaChannel::SetMaxSendBitrate. [001:082] (webrtc_voice_engine.cc:1488): Setting voice channel options: AudioOptions {} [001:083] (webrtc_voice_engine.cc:310): 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, } [001:083] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [001:084] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [001:084] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [001:084] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [001:084] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [001:084] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [001:084] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [001:084] (audio_processing_impl.cc:696): Highpass filter activated: 1 [001:086] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [001:087] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [001:087] (webrtc_voice_engine.cc:1506): 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, } [001:088] (webrtc_voice_engine.cc:1852): AddRecvStream: {id:183cfbe2-7712-4d89-be37-e6bf530e22fa;ssrcs:[1340491237];ssrc_groups:;cname:vWjbMZWRqzqeHiad;stream_ids:;} [001:090] (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 [001:091] (audio_coding_module.cc:333): Created [001:091] (audio_receive_stream.cc:117): AudioReceiveStream: 1340491237 [001:091] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 1340491237, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: (Transport)} [001:091] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [001:091] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [001:092] (channel.cc:719): Add remote ssrc: 1340491237 [001:092] (channel.cc:863): Changing voice state, recv=0 send=0 [001:094] (peer_connection.cc:2687): Processing the MSIDs for MID=0 (streams=[]). [001:094] (peer_connection.cc:2687): Processing the MSIDs for MID=1 (streams=[]). [001:095] (peer_connection.cc:2698): Processing the addition of a remote track for MID=1. [001:096] (webrtc_voice_engine.cc:1965): SetOutputVolume() to 1 for recv stream with ssrc 1340491237 [001:138] (peer_connection_integrationtest.cc:753): Callee: SetLocalDescriptionAndSendSdpMessage [001:159] (srtp_filter.cc:194): Applying the same SRTP send parameters again. No-op. [001:160] (srtp_filter.cc:225): Applying the same SRTP recv parameters again. No-op. [001:160] (srtp_transport.cc:309): SRTP updated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [001:160] (p2p_transport_channel.cc:432): Set ICE ufrag: I6DM pwd: /A6Y7CvF3KEQXswBCqf8iPzP on transport 0 [001:160] (dtls_transport.cc:253): DtlsTransport[0|1|RW]: Other side didn't support DTLS. [001:161] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from kSendRecv to kSendRecv. [001:161] (rtp_transceiver.cc:169): Changing transceiver (MID=1) current direction from <not set> to kSendRecv. [001:162] (channel.cc:553): Channel enabled [001:162] (channel.cc:863): Changing voice state, recv=0 send=0 [001:162] (peer_connection.cc:4097): Session: 3265430737459529171 Old state: kHaveRemoteOffer New state: kStable [001:163] (channel.cc:1009): Setting local video description [001:165] (webrtc_video_engine.cc:945): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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}]} [001:168] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [001:169] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [001:169] (channel.cc:996): Changing video state, send=1 [001:169] (channel.cc:871): Setting local voice description [001:170] (webrtc_voice_engine.cc:1327): 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}]} [001:170] (webrtc_voice_engine.cc:1516): Setting receive voice codecs. [001:171] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 1340491237, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: (Transport)} [001:172] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [001:172] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [001:172] (audio_receive_stream.cc:140): ~AudioReceiveStream: 1340491237 [001:173] (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 [001:173] (audio_coding_module.cc:333): Created [001:173] (audio_receive_stream.cc:117): AudioReceiveStream: 1340491237 [001:175] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 1340491237, 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: (Transport)} [001:176] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [001:176] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [001:176] (webrtc_voice_engine.cc:1787): AddSendStream: {id:a945aed0-cc8a-4bcb-8f34-c33853c70068;ssrcs:[80504648];ssrc_groups:;cname:vBzBF3+EwbdtI2fQ;stream_ids:;} [001:178] (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 [001:178] (audio_coding_module.cc:333): Created [001:178] (channel_send.cc:704): Setting media_transport_ rate observers. [001:179] (audio_send_stream.cc:142): AudioSendStream: 80504648 [001:179] (channel_send.cc:816): Trying to call unset encoder. [001:179] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 80504648, 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: vBzBF3+EwbdtI2fQ}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), 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}}}} [001:179] (audio_encoder_opus.cc:772): Set Opus bitrate to 32000 bps. [001:180] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [001:180] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [001:180] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 1340491237, local_ssrc: 80504648, 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: (Transport)} [001:180] (channel.cc:676): Add send stream ssrc: 80504648 [001:181] (webrtc_voice_engine.cc:310): 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, } [001:181] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [001:181] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [001:181] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [001:181] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [001:181] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [001:181] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [001:181] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [001:181] (audio_processing_impl.cc:696): Highpass filter activated: 1 [001:182] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [001:182] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [001:182] (channel.cc:863): Changing voice state, recv=1 send=1 [001:190] (webrtc_voice_engine.cc:1488): Setting voice channel options: AudioOptions {hf: 0, } [001:190] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {hf: 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, } [001:190] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [001:191] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [001:191] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [001:191] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [001:191] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [001:191] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [001:191] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [001:191] (audio_processing_impl.cc:696): Highpass filter activated: 0 [001:192] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [001:192] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [001:192] (webrtc_voice_engine.cc:1506): Set voice channel options. Current options: AudioOptions {hf: 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, } [001:193] (peer_connection_integrationtest.cc:703): Caller: HandleIncomingAnswer [001:194] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [001:199] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [001:199] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [001:199] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [001:199] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [001:199] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [001:199] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [001:203] (render_delay_buffer.cc:306): Receiving a first externally reported audio buffer delay of 0 ms. [001:203] (agc_manager_direct.cc:383): [agc] Initial GetMicVolume()=0 [001:203] (agc_manager_direct.cc:388): [agc] Initial volume too low, raising to 12 [001:204] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [001:207] (render_delay_buffer.cc:334): Applying total delay of 1 blocks. [001:207] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 1: 2 blocks [001:208] (echo_remover.cc:213): Gain change detected at block 1 [001:208] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 2: 3 blocks [001:209] (render_delay_buffer.cc:255): Render buffer underrun detected at block 2 [001:210] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 3: 4 blocks [001:210] (render_delay_buffer.cc:255): Render buffer underrun detected at block 3 [001:211] (peer_connection_integrationtest.cc:771): Caller: SetRemoteDescription [001:239] (srtp_filter.cc:194): Applying the same SRTP send parameters again. No-op. [001:239] (srtp_filter.cc:225): Applying the same SRTP recv parameters again. No-op. [001:239] (srtp_transport.cc:309): SRTP updated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [001:240] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=I6DM, renomination disabled [001:240] (dtls_transport.cc:253): DtlsTransport[0|1|RW]: Other side didn't support DTLS. [001:240] (channel.cc:553): Channel enabled [001:241] (channel.cc:863): Changing voice state, recv=1 send=0 [001:241] (peer_connection.cc:4097): Session: 6675921963715691949 Old state: kHaveLocalOffer New state: kStable [001:241] (channel.cc:1061): Setting remote video description [001:242] (webrtc_video_engine.cc:670): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 0} [001:259] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [001:259] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [001:259] (channel.cc:996): Changing video state, send=1 [001:263] (channel.cc:923): Setting remote voice description [001:264] (webrtc_voice_engine.cc:1282): 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: 1, options: AudioOptions {}} [001:265] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1340491237, extmap-allow-mixed: false, extensions: [], c_name: vWjbMZWRqzqeHiad}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), 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}}}} [001:265] (audio_encoder_opus.cc:772): Set Opus bitrate to 32000 bps. [001:265] (webrtc_voice_engine.cc:1705): Recreate all the receive streams because the send codec has changed. [001:266] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1340491237, 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: vWjbMZWRqzqeHiad}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), 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}}}} [001:266] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1340491237, 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: vWjbMZWRqzqeHiad}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), 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}}}} [001:266] (webrtc_voice_engine.cc:2166): WebRtcVoiceMediaChannel::SetMaxSendBitrate. [001:266] (webrtc_voice_engine.cc:1488): Setting voice channel options: AudioOptions {} [001:266] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {hf: 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, } [001:266] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [001:266] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [001:267] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [001:267] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [001:267] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [001:267] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [001:267] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [001:267] (audio_processing_impl.cc:696): Highpass filter activated: 0 [001:267] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [001:267] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [001:268] (webrtc_voice_engine.cc:1506): Set voice channel options. Current options: AudioOptions {hf: 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, } [001:268] (webrtc_voice_engine.cc:1852): AddRecvStream: {id:a945aed0-cc8a-4bcb-8f34-c33853c70068;ssrcs:[80504648];ssrc_groups:;cname:vBzBF3+EwbdtI2fQ;stream_ids:;} [001:268] (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 [001:269] (audio_coding_module.cc:333): Created [001:269] (audio_receive_stream.cc:117): AudioReceiveStream: 80504648 [001:269] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 80504648, local_ssrc: 1340491237, 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: (Transport)} [001:270] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [001:270] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [001:274] (channel.cc:719): Add remote ssrc: 80504648 [001:278] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [001:278] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [001:278] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [001:278] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [001:278] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [001:278] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [001:282] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {hf: 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, } [001:282] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [001:282] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [001:282] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [001:282] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [001:282] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [001:282] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [001:282] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [001:282] (audio_processing_impl.cc:696): Highpass filter activated: 0 [001:283] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [001:283] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [001:283] (channel.cc:863): Changing voice state, recv=1 send=1 [001:285] (peer_connection.cc:2687): Processing the MSIDs for MID=0 (streams=[]). [001:286] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from kSendRecv to kSendRecv. [001:286] (peer_connection.cc:2687): Processing the MSIDs for MID=1 (streams=[]). [001:286] (peer_connection.cc:2698): Processing the addition of a remote track for MID=1. [001:286] (rtp_transceiver.cc:169): Changing transceiver (MID=1) current direction from <not set> to kSendRecv. [001:289] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [001:289] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [001:289] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [001:289] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [001:289] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [001:289] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [001:293] (render_delay_buffer.cc:306): Receiving a first externally reported audio buffer delay of 0 ms. [001:293] (agc_manager_direct.cc:383): [agc] Initial GetMicVolume()=0 [001:293] (agc_manager_direct.cc:388): [agc] Initial volume too low, raising to 12 [001:297] (render_delay_buffer.cc:334): Applying total delay of 1 blocks. [001:297] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 1: 2 blocks [001:298] (echo_remover.cc:213): Gain change detected at block 1 [001:298] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 2: 3 blocks [001:298] (render_delay_buffer.cc:255): Render buffer underrun detected at block 2 [001:300] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 3: 4 blocks [001:300] (render_delay_buffer.cc:255): Render buffer underrun detected at block 3 [001:303] (webrtc_voice_engine.cc:1965): SetOutputVolume() to 1 for recv stream with ssrc 80504648 [001:317] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [001:317] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [001:317] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [001:317] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [001:317] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [001:318] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [001:323] (render_delay_buffer.cc:306): Receiving a first externally reported audio buffer delay of 0 ms. [001:323] (render_delay_buffer.cc:334): Applying total delay of 1 blocks. [001:323] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 1: 2 blocks [001:324] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 2: 3 blocks [001:325] (render_delay_buffer.cc:255): Render buffer underrun detected at block 2 [001:326] (agc_manager_direct.cc:383): [agc] Initial GetMicVolume()=12 [001:825] (message_queue.cc:518): Message took 66ms to dispatch. Posted from: FakeAudioCaptureModule::ProcessFrameP@../../pc/test/fake_audio_capture_module.cc:487 [001:921] (message_queue.cc:518): Message took 61ms to dispatch. Posted from: FakeAudioCaptureModule::ProcessFrameP@../../pc/test/fake_audio_capture_module.cc:487 [002:321] (message_queue.cc:518): Message took 57ms to dispatch. Posted from: FakeAudioCaptureModule::ProcessFrameP@../../pc/test/fake_audio_capture_module.cc:487 [003:686] (message_queue.cc:518): Message took 54ms to dispatch. Posted from: FakeAudioCaptureModule::ProcessFrameP@../../pc/test/fake_audio_capture_module.cc:487 [008:361] (message_queue.cc:518): Message took 53ms to dispatch. Posted from: FakeAudioCaptureModule::ProcessFrameP@../../pc/test/fake_audio_capture_module.cc:487 [010:392] (message_queue.cc:518): Message took 100ms to dispatch. Posted from: FakeAudioCaptureModule::ProcessFrameP@../../pc/test/fake_audio_capture_module.cc:487 [010:710] (message_queue.cc:518): Message took 72ms to dispatch. Posted from: FakeAudioCaptureModule::ProcessFrameP@../../pc/test/fake_audio_capture_module.cc:487 ../../pc/peer_connection_integrationtest.cc(1502): error: Value of: caller()->audio_frames_received() >= total_caller_audio_frames_expected && caller()->min_video_frames_received_per_track() >= total_caller_video_frames_expected && callee()->audio_frames_received() >= total_callee_audio_frames_expected && callee()->min_video_frames_received_per_track() >= total_callee_video_frames_expected Actual: false Expected: true Stack trace: 000000013F9F6B10: testing::DefaultValue<rtc::AsyncResolverInterface * __ptr64>::Exists 000000013FA4B6E9: webrtc::VP9EncoderImpl::PopulateCodecSpecific 00000001402F625C: testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void> 00000001402F60FB: testing::internal::HandleExceptionsInMethodIfSupported<testing::Test,void> 00000001403195DA: testing::Test::Run 0000000140319725: testing::TestInfo::Run ... Google Test internal frames ... ../../pc/peer_connection_integrationtest.cc(1518): error: Expected: (caller()->min_video_frames_received_per_track()) >= (total_caller_video_frames_expected), actual: 0 vs 3 Stack trace: 000000013F9F6D66: testing::DefaultValue<rtc::AsyncResolverInterface * __ptr64>::Exists 000000013FA4B6E9: webrtc::VP9EncoderImpl::PopulateCodecSpecific 00000001402F625C: testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void> 00000001402F60FB: testing::internal::HandleExceptionsInMethodIfSupported<testing::Test,void> 00000001403195DA: testing::Test::Run 0000000140319725: testing::TestInfo::Run ... Google Test internal frames ... ../../pc/peer_connection_integrationtest.cc(3614): error: Value of: ExpectNewFrames(media_expectations) Actual: false Expected: true Stack trace: 000000013FA4B79E: webrtc::VP9EncoderImpl::PopulateCodecSpecific 00000001402F625C: testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void> 00000001402F60FB: testing::internal::HandleExceptionsInMethodIfSupported<testing::Test,void> 00000001403195DA: testing::Test::Run 0000000140319725: testing::TestInfo::Run ... Google Test internal frames ... [011:491] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 653648138, options: nullptr, source = nullptr) [011:492] (webrtc_video_engine.cc:1293): SetSink: ssrc:3350319844 nullptr [011:519] (webrtc_voice_engine.cc:1965): SetOutputVolume() to 0 for recv stream with ssrc 80504648 [011:521] (channel.cc:563): Channel disabled [011:521] (video_send_stream.cc:160): VideoSendStream::Stop [011:521] (channel.cc:996): Changing video state, send=0 [011:521] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:522] (rtp_rtcp_impl.cc:359): Failed to send RTCP BYE [011:522] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps [011:522] (video_stream_encoder.cc:1477): Video suspend state changed to: suspended [011:529] (video_send_stream.cc:160): VideoSendStream::Stop [011:530] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:559] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:559] (video_send_stream_impl.cc:337): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [653648138], 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: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [1380348883], payload_type: 97}, c_name: vWjbMZWRqzqeHiad}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [011:561] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:561] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:562] (send_statistics_proxy.cc:654): Frames encoded 108 WebRTC.Video.DroppedFrames.Capturer 0 WebRTC.Video.DroppedFrames.EncoderQueue 0 WebRTC.Video.DroppedFrames.Encoder 0 WebRTC.Video.DroppedFrames.Ratelimiter 0 [011:562] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:562] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:563] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 3350319844, local_ssrc: 653648138, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 3987717279, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [011:565] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [011:584] (receive_statistics_proxy.cc:487): WebRTC.Video.ReceiveStreamLifetimeInSeconds 10 Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [011:585] (channel.cc:162): Destroyed channel: 0 [011:586] (message_queue.cc:518): Message took 65ms to dispatch. Posted from: cricket::ChannelManager::DestroyVideoChannel@../../pc/channel_manager.cc:276 [011:587] (channel.cc:563): Channel disabled [011:589] (channel.cc:863): Changing voice state, recv=0 send=0 [011:589] (webrtc_voice_engine.cc:1826): RemoveSendStream: 1340491237 [011:590] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:590] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:590] (audio_send_stream.cc:179): ~AudioSendStream: 1340491237 [011:605] (webrtc_voice_engine.cc:1903): RemoveRecvStream: 80504648 [011:606] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [011:606] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [011:606] (audio_receive_stream.cc:140): ~AudioReceiveStream: 80504648 [011:606] (paced_sender.cc:115): PacedSender paused. [011:607] (channel.cc:162): Destroyed channel: 1 [011:607] (peer_connection.cc:901): Session: 6675921963715691949 is destroyed. [011:619] (paced_sender.cc:427): ProcessThreadAttached 0x0 [011:620] (paced_sender.cc:427): ProcessThreadAttached 0x0 [011:621] (rtc_event_log_impl.cc:212): Stopping WebRTC event log. [011:628] (rtc_event_log_impl.cc:229): WebRTC event log successfully stopped. [011:633] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue. [011:640] (webrtc_video_engine.cc:464): WebRtcVideoEngine::~WebRtcVideoEngine [011:642] (webrtc_voice_engine.cc:203): WebRtcVoiceEngine::~WebRtcVoiceEngine [011:646] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 3350319844, options: nullptr, source = nullptr) [011:647] (webrtc_video_engine.cc:1293): SetSink: ssrc:653648138 nullptr [011:658] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [011:659] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 256 ms. [011:659] (matched_filter.cc:450): Filter 1: start: 192 ms, end: 448 ms. [011:659] (matched_filter.cc:450): Filter 2: start: 384 ms, end: 640 ms. [011:660] (matched_filter.cc:450): Filter 3: start: 576 ms, end: 832 ms. [011:660] (matched_filter.cc:450): Filter 4: start: 768 ms, end: 1024 ms. [011:663] (render_delay_buffer.cc:306): Receiving a first externally reported audio buffer delay of 0 ms. [011:664] (agc_manager_direct.cc:383): [agc] Initial GetMicVolume()=12 [011:665] (render_delay_buffer.cc:334): Applying total delay of 1 blocks. [011:666] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 1: 2 blocks [011:671] (render_delay_buffer.cc:255): Render buffer underrun detected at block 4 [011:676] (webrtc_voice_engine.cc:1965): SetOutputVolume() to 0 for recv stream with ssrc 1340491237 [011:677] (channel.cc:563): Channel disabled [011:678] (video_send_stream.cc:160): VideoSendStream::Stop [011:678] (channel.cc:996): Changing video state, send=0 [011:678] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:679] (rtp_rtcp_impl.cc:359): Failed to send RTCP BYE [011:679] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps [011:680] (video_send_stream.cc:160): VideoSendStream::Stop [011:680] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:681] (video_stream_encoder.cc:1477): Video suspend state changed to: suspended [011:687] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:688] (video_send_stream_impl.cc:337): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [3350319844], 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: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [3987717279], payload_type: 97}, c_name: vBzBF3+EwbdtI2fQ}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [011:688] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:688] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:689] (send_statistics_proxy.cc:654): Frames encoded 112 WebRTC.Video.DroppedFrames.Capturer 0 WebRTC.Video.DroppedFrames.EncoderQueue 1 WebRTC.Video.DroppedFrames.Encoder 0 WebRTC.Video.DroppedFrames.Ratelimiter 0 [011:689] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:689] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:690] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 653648138, local_ssrc: 3350319844, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 1380348883, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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, target_delay_ms: 0} [011:692] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [011:694] (receive_statistics_proxy.cc:487): WebRTC.Video.ReceiveStreamLifetimeInSeconds 11 Frames decoded 108 WebRTC.Video.DroppedFrames.Receiver -108 WebRTC.Video.InterframeDelay95PercentileInMs 111 WebRTC.Video.InterframeDelay95PercentileInMs.S0 111 WebRTC.Video.MediaBitrateReceivedInKbps.S0 7 [011:694] (video_quality_observer.cc:135): WebRTC.Video.MeanTimeBetweenFreezesMs 10652 WebRTC.Video.TimeInHdPercentage 0 WebRTC.Video.TimeInBlockyVideoPercentage 0 WebRTC.Video.NumberResolutionDownswitchesPerMinute 0 WebRTC.Video.NumberFreezesPerMinute 0 WebRTC.Video.HarmonicFrameRate 10 [011:695] (channel.cc:162): Destroyed channel: 0 [011:700] (channel.cc:563): Channel disabled [011:700] (channel.cc:863): Changing voice state, recv=0 send=0 [011:701] (webrtc_voice_engine.cc:1826): RemoveSendStream: 80504648 [011:701] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:701] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:701] (audio_send_stream.cc:179): ~AudioSendStream: 80504648 [011:702] (webrtc_voice_engine.cc:1903): RemoveRecvStream: 1340491237 [011:703] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [011:703] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [011:703] (audio_receive_stream.cc:140): ~AudioReceiveStream: 1340491237 [011:703] (paced_sender.cc:115): PacedSender paused. [011:704] (channel.cc:162): Destroyed channel: 1 [011:704] (peer_connection.cc:901): Session: 3265430737459529171 is destroyed. [011:707] (paced_sender.cc:427): ProcessThreadAttached 0x0 [011:707] (paced_sender.cc:427): ProcessThreadAttached 0x0 [011:708] (rtc_event_log_impl.cc:212): Stopping WebRTC event log. [011:708] (rtc_event_log_impl.cc:229): WebRTC event log successfully stopped. [011:712] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue. [011:719] (webrtc_video_engine.cc:464): WebRtcVideoEngine::~WebRtcVideoEngine [011:719] (webrtc_voice_engine.cc:203): WebRtcVoiceEngine::~WebRtcVoiceEngine [011:723] (loopback_media_transport.cc:160): ~LoopbackMediaTransport [011:723] (loopback_media_transport.cc:160): ~LoopbackMediaTransport [ FAILED ] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgrade/1, where GetParam() = 4-byte object <01-00 00-00> (11731 ms) [----------] 1 test from PeerConnectionIntegrationTest/PeerConnectionIntegrationTest (11732 ms total) [----------] Global test environment tear-down [==========] 1 test from 1 test suite ran. (11733 ms total) [ PASSED ] 0 tests. [ FAILED ] 1 test, listed below: [ FAILED ] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgrade/1, where GetParam() = 4-byte object <01-00 00-00> 1 FAILED TEST [82/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgrade/1 returned/aborted with exit code 1 (12536 ms) [83/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.TurnCustomizerUsedForTurnConnections/1 (1075 ms) [84/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.MediaSectionCountEnforcedForSubsequentOffer/1 (786 ms) [85/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/17 (526 ms) [86/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/33 (528 ms) [87/451] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetRemoteAnswer/1 (521 ms) [88/451] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetLocalAnswer/5 (599 ms) [89/451] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingLocalRejectedAndNoRemote/1 (634 ms) [90/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.GetStatsForVideoTrack/1 (752 ms) [91/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CloseAndTestStreamsAndStates/1 (822 ms) [92/451] PeerConnectionRtpTest/PeerConnectionRtpTest.CreateTwoSendersWithSameTrack/1 (769 ms) [93/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.Dtls12CipherStatsAndUmaMetrics/1 (1407 ms) [94/451] PeerConnectionMediaTest/PeerConnectionMediaTest.MediaEngineErrorPropagatedToClients/1 (811 ms) [95/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/13 (779 ms) [96/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportBidirectionalAudio/1 (1053 ms) [97/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgrade/1 (1835 ms) [98/451] PeerConnectionSignalingTest/PeerConnectionSignalingTest.SessionVersionIncrementedInSubsequentDifferentOffer/1 (601 ms) [99/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/1 (610 ms) [100/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/13 (673 ms) [101/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/17 (657 ms) [102/451] PeerConnectionBundleTest/PeerConnectionBundleTest.RejectDescriptionChangingBundleTag/1 (722 ms) [103/451] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.CreateAnswerWithRemoteSctpDataChannelIncludesDataSection/1 (570 ms) [104/451] PeerConnectionMediaTest/PeerConnectionMediaTest.MediaTransportForMediaAndDataChannels/1 (540 ms) [105/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationNotCausingIceRestart/1 (942 ms) [106/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.NonGcmCipherUsedWhenGcmNotSupported/1 (1457 ms) [107/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/17 (833 ms) [108/451] PeerConnectionBundleTest/PeerConnectionBundleTest.BundleOnFirstMidInAnswer/1 (798 ms) [109/451] PeerConnectionJsepTest.OfferAnswerWithChangedMids (822 ms) [110/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetRemoteAnswerWithNoCryptosWhenSdesOn/1 (798 ms) [111/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/25 (742 ms) [112/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CurrentAndPendingDescriptions/1 (559 ms) [113/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/33 (549 ms) [114/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/45 (602 ms) [115/451] PeerConnectionMediaTest/PeerConnectionMediaOfferDirectionTest.VerifyDirection/1 (645 ms) [116/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.Aes128Sha1_32_CipherNotUsedWhenOnlyCalleeSupported/1 (1560 ms) [117/451] PeerConnectionRtpTest/PeerConnectionRtpTest.AddTrackWithoutStreamFiresOnAddTrack/1 (1098 ms) [118/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/45 (1068 ms) [119/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.NonGcmCipherUsedWhenOnlyCallerSupportsGcm/1 (2539 ms) [120/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/9 (1321 ms) [121/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.RtpDataChannelSignaledClosedInCalleeOffer/1 (2603 ms) [122/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetStatsForUnsignaledStreamWithNewStatsApi/1 (1177 ms) [123/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.SctpDataChannelConfigSentToOtherSide/1 (1662 ms) [124/451] PeerConnectionRtpTest/PeerConnectionRtpTest.RemoveTrackWithSharedStreamFiresOnRemoveTrack/1 (1072 ms) [125/451] PeerConnectionJsepTest.IncludeMsidEvenIfDirectionHasChanged (862 ms) [126/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferReceivePrAnswerAndAnswer/1 (767 ms) [127/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/5 (909 ms) [128/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/1 (749 ms) [129/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.RtpDataChannelsRejectedByCallee/1 (1269 ms) [130/451] PeerConnectionSignalingTest/PeerConnectionSignalingTest.SetRemoteOfferTwiceWorks/1 (620 ms) [131/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/9 (1141 ms) [132/451] PeerConnectionJsepTest.SetRemoteOfferDoesNotReuseTransceiverOfWrongType (763 ms) [133/451] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.SctpPortPropagatedFromSdpToTransport/1 (500 ms) [134/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/1 (621 ms) [135/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SsrcInOfferAnswer/1 (603 ms) [136/451] PeerConnectionJsepTest.SetRemoteOfferCreatesTransceivers (657 ms) [137/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/13 (704 ms) [138/451] PeerConnectionIntegrationTestUnifiedPlan.NoStreamsMsidLinePresent (993 ms) [139/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/9 (564 ms) [140/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/13 (549 ms) [141/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/21 (565 ms) [142/451] PeerConnectionRtpTest/PeerConnectionRtpTest.LegacyObserverOnSuccess/1 (573 ms) [143/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/1 (637 ms) [144/451] PeerConnectionInterfaceTestPlanB.AddTrackRemoveTrack (590 ms) [145/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetLocalAnswerWithNoFingerprintWhenDtlsOn/1 (674 ms) [146/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.TestRejectSctpDataChannelInAnswer/1 (596 ms) [147/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/5 (1009 ms) [148/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/1 (572 ms) [149/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/5 (602 ms) [150/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/17 (555 ms) [151/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/17 (664 ms) [152/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInAnswerWithSdesAndGcm/1 (563 ms) [153/451] PeerConnectionJsepTest.SetRemoteOfferWithOneTrackNoStreamFiresOnAddTrack (642 ms) [154/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/9 (628 ms) [155/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/9 (865 ms) [156/451] PeerConnectionIceTest/PeerConnectionIceTest.CanSetRemoteSessionDescriptionWithRemoteCandidates/1 (579 ms) [157/451] PeerConnectionIceTest/PeerConnectionIceTest.CannotAddCandidateWhenRemoteDescriptionNotSet/1 (477 ms) [158/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateSubsequentRecvOnlyOffer/1 (539 ms) [159/451] PeerConnectionMediaTest/PeerConnectionMediaTest.FailToSetRemoteDescriptionIfCreateMediaChannelFails/1 (538 ms) [160/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/21 (521 ms) [161/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/17 (549 ms) [162/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.ExchangeOfferAnswerWhenSdesOn/1 (810 ms) [163/451] PeerConnectionRtpTestUnifiedPlan.ReofferDoesNotCallOnTrack (916 ms) [164/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationInteropTest.NoMediaLocalToNoMediaRemote/1 (599 ms) [165/451] PeerConnectionIceTest/PeerConnectionIceTest.OfferFromLiteIceControlledAndAnswerFromFullIceControlling/1 (791 ms) [166/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/5 (940 ms) [167/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.IceStatesReachCompletionWithRemoteHostname/1 (1228 ms) [168/451] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.CreateDataChannelBeforeNegotiate/1 (830 ms) [169/451] PeerConnectionJsepTest/RecycleMediaSectionTest.CurrentLocalAndCurrentRemoteRejected/1 (891 ms) [170/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/9 (953 ms) [171/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.VerifyBestConnection/3 (1287 ms) [172/451] PeerConnectionJsepTest.RecycleMediaSectionWhenStoppingTransceiverOnAnswerer (786 ms) [173/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/9 (784 ms) [174/451] PeerConnectionIceTest/PeerConnectionIceTest.CandidateInSubsequentOfferIsAddedToRemoteDescription/1 (867 ms) [175/451] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.MessageTransferBetweenTwoPairsOfDataChannels/1 (837 ms) [176/451] PeerConnectionIceTest/PeerConnectionIceTest.LaterAnswerHasSameIceCredentialsIfNoIceRestart/1 (885 ms) [177/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/13 (575 ms) [178/451] PeerConnectionIceTest/PeerConnectionIceUfragPwdAnswerTest.TestIncludedInAnswer/5 (888 ms) [179/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.DataChannelWorksWhenSpecCompliantSctpOfferReceived/1 (759 ms) [180/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/17 (828 ms) [181/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/5 (792 ms) [182/451] PeerConnectionIceTest/PeerConnectionIceTest.DuplicateIceCandidateIgnoredWhenAdded/1 (806 ms) [183/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/13 (739 ms) [184/451] PeerConnectionIceTest/PeerConnectionIceUfragPwdAnswerTest.TestIncludedInAnswer/1 (881 ms) [185/451] PeerConnectionRtpTest/PeerConnectionRtpTest.AddTrackWithStreamAddsReceiver/1 (637 ms) [186/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/13 (823 ms) [187/451] PeerConnectionMediaConfigTest.TestEnableExperimentCpuLoadEstimator (524 ms) [188/451] SdpFormatReceivedTest.DataChannelOnlyIsReportedAsNoTracks (549 ms) [189/451] PeerConnectionRtpTestUnifiedPlan.CreateAnswerSameTrackLabel (618 ms) [190/451] PeerConnectionSignalingTest/PeerConnectionSignalingTest.InitiatorFlagSetOnCallerAndNotOnCallee/1 (841 ms) [191/451] PeerConnectionBundleTest/PeerConnectionBundleTest.FailToSetDescriptionWithBundleAndNoRtcpMux/1 (552 ms) [192/451] PeerConnectionBundleTest/PeerConnectionBundleTest.NeverCreateRtcpTransportWithRtcpMuxRequired/1 (848 ms) [193/451] PeerConnectionIceTest/PeerConnectionIceTest.SetRemoteDescriptionFailsIfNoIceCredentials/1 (644 ms) [194/451] PeerConnectionMediaTest/PeerConnectionMediaOfferDirectionTest.VerifyDirection/9 (472 ms) [195/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.TestSendBinaryOnRtpDataChannel/1 (587 ms) [196/451] PeerConnectionIceTest/PeerConnectionIceTest.LocalCandidatesRemovedWhenNetworkDownIfGatheringContinually/1 (611 ms) [197/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.DataChannelCloseWhenPeerConnectionClose/1 (608 ms) [198/451] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.MediaTransportDataChannelCreatedEvenIfSctpAvailable/1 (472 ms) [199/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/1 (513 ms) [200/451] PeerConnectionBundleTest/PeerConnectionBundleTest.RemoveContentFromBundleGroup/1 (825 ms) [201/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.VerifyIceStates/3 (49799 ms) [202/451] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.CreateDataChannelLargeTransfer/1 (49861 ms) [203/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/17 (864 ms) [204/451] PeerConnectionIceTest/PeerConnectionIceTest.TwoTrickledCandidatesAddedToRemoteDescription/1 (901 ms) [205/451] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.DataChannelFromOpenWorksWhilePreviousChannelClosing/1 (947 ms) [206/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/1 (469 ms) [207/451] PeerConnectionSignalingTest/PeerConnectionSignalingTest.FailToSetNullRemoteDescription/1 (550 ms) [208/451] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.NoSctpTransportCreatedIfRtpDataChannelEnabled/1 (470 ms) [209/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/5 (500 ms) [210/451] PeerConnectionSimulcastTests.CanCreateTransceiverWithRid (531 ms) [211/451] PeerConnectionJsepTest/RecycleMediaSectionTest.CurrentRemoteOnlyRejected/1 (2689 ms) [212/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/1 (2361 ms) [213/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/9 (2721 ms) [214/451] PeerConnectionIceTest/PeerConnectionIceTest.AddRemoveCandidateWithEmptyTransportDoesNotCrash/1 (2706 ms) [215/451] PeerConnectionJsepTest/RecycleMediaSectionTest.CurrentLocalOnlyRejected/1 (2722 ms) [216/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInOfferWhenEncryptionDisabled/1 (769 ms) [217/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.GetConfigurationAfterSetConfiguration/1 (661 ms) [218/451] PeerConnectionInterfaceTestPlanB.CloseAndTestMethods (2761 ms) [219/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/5 (2816 ms) [220/451] PeerConnectionBundleTest/PeerConnectionBundleMatrixTest.VerifyTransportsBeforeAndAfterSettingRemoteAnswer/7 (2801 ms) [221/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/5 (1038 ms) [222/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetLocalOfferWithNoFingerprintWhenDtlsOn/1 (411 ms) [223/451] PeerConnectionBundleTest/PeerConnectionBundleTest.TransportsSameForMaxBundleWithBundleInRemoteOffer/1 (853 ms) [224/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.ExchangeOfferAnswerWhenNoEncryption/1 (665 ms) [225/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.UpdateRemoteStreams/1 (473 ms) [226/451] PeerConnectionBundleTest/PeerConnectionBundleMatrixTest.VerifyTransportsBeforeAndAfterSettingRemoteAnswer/3 (855 ms) [227/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.IceEventsGeneratedAndLoggedInRtcEventLog/1 (600 ms) [228/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/1 (779 ms) [229/451] PeerConnectionIceConfigTest.SetStunCandidateKeepaliveInterval (443 ms) [230/451] PeerConnectionInterfaceTestPlanB.DefaultTracksNotDestroyedAndRecreated (424 ms) [231/451] PeerConnectionInterfaceTestPlanB.MediaStreamAddTrackRemoveTrackRenegotiate (442 ms) [232/451] PeerConnectionRtpTestUnifiedPlan.RemoveTrackNoErrorIfTrackAlreadyRemoved (754 ms) [233/451] SdpFormatReceivedTest.ComplexPlanBIsReportedAsComplexPlanB (748 ms) [234/451] PeerConnectionJsepTest.AddingTrackWithAddTrackSpecifiesTrackId (1000 ms) [235/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.RTCConfigurationCryptoOptionOverridesFactory/1 (948 ms) [236/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetRemoteOfferWithNoCryptosWhenSdesOn/1 (804 ms) [237/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SctpDuplicatedLabelAllowed/1 (686 ms) [238/451] PeerConnectionIceTest/PeerConnectionIceTest.IceCredentialsCreateOffer/1 (733 ms) [239/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.RecvonlyDescriptionDoesntCreateStream/1 (714 ms) [240/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.IceRenominationNotOffered/1 (563 ms) [241/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferWithOfferToReceiveConstraints/1 (603 ms) [242/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateWithoutMinSucceeds/1 (401 ms) [243/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetIceRegatherIntervalRangeWithContinualGathering/1 (444 ms) [244/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferWithVideoOnlyOptions/1 (491 ms) [245/451] PeerConnectionInterfaceTestPlanB.SendOnlySdpWithoutMsidCreatesDefaultStream (460 ms) [246/451] PeerConnectionMediaTest/PeerConnectionMediaOfferDirectionTest.VerifyDirection/5 (507 ms) [247/451] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingRemoteRejectedAndNoLocal/1 (648 ms) [248/451] RTCStatsCollectorTest.CollectRTCInboundRTPStreamStats_Video (333 ms) [249/451] PeerConnectionJsepTest.VideoOnlyInitialOffer (589 ms) [250/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInOfferWhenDtlsEnabled/1 (478 ms) [251/451] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.MediaTransportWithoutSdesFails/1 (399 ms) [252/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.AttachmentIdIsSetOnAddTrack/1 (398 ms) [253/451] PeerConnectionMediaTest/PeerConnectionMediaTest.OfferHasDifferentDirectionForAudioVideo/1 (444 ms) [254/451] PeerConnectionJsepTest.CurrentDirectionResetWhenRtpTransceiverStopped (792 ms) [255/451] PeerConnectionFactoryTest.CreatePCUsingIceServersUrls (489 ms) [256/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/13 (625 ms) [257/451] PeerConnectionJsepTest.MediaSectionsInInitialOfferHaveDifferentMids (447 ms) [258/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.DtlsSdesFallbackNotSupported/1 (650 ms) [259/451] PeerConnectionRtpTestUnifiedPlan.NoRenegotiationNeededAfterSetDirectionOnStoppedTransceiver (604 ms) [260/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/13 (712 ms) [261/451] PeerConnectionMediaTestPlanB.EmptyRemoteOfferRemovesRecvStreams (728 ms) [262/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/21 (534 ms) [263/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationChangesPruneTurnPortsFlag/1 (549 ms) [264/451] PeerConnectionSimulcastTests.TransceiverIsNotRecycledWithSimulcast (848 ms) [265/451] PeerConnectionSimulcastTests.PausedSimulcastLayersAreDisabledInSender (911 ms) [266/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferWithIceRestart/1 (673 ms) [267/451] PeerConnectionJsepTest.CreateAnswerRejectsStoppedTransceiver (657 ms) [268/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/13 (674 ms) [269/451] PeerConnectionJsepTest.LegacyNoMidTwoRemoteOffers (721 ms) [270/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportDataChannelCalleeCloses/1 (690 ms) [271/451] PeerConnectionRtpTestPlanB.RemoteStreamIdChangesFiresOnRemoveAndOnAddTrack (736 ms) [272/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.OnAddTrackCallback/1 (621 ms) [273/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.ExtraPooledSessionsDiscardedAfterApplyingAnswer/1 (621 ms) [274/451] PeerConnectionRtpTestUnifiedPlan.AddTransceiverWithInvalidKindReturnsError (354 ms) [275/451] PeerConnectionRtpTestUnifiedPlan.AddTrackDoesNotReuseTransceiverOfWrongType (368 ms) [276/451] PeerConnectionRtpTestUnifiedPlan.AddTrackCreatesSenderWithTrackId (384 ms) [277/451] PeerConnectionSimulcastTests.SingleRidIsRemovedFromSessionDescription (432 ms) [278/451] RTCStatsCollectorTest.GetStatsWithNullReceiverSelector (447 ms) [279/451] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.CreateDataChannelWithSctpDisabledFails/1 (462 ms) [280/451] WebRtcSdpTest.SerializeSessionDescriptionWithRecvOnlyContent (424 ms) [281/451] NumberOfSendEncodings/PeerConnectionSimulcastMetricsTests.NumberOfSendEncodingsIsLogged/1 (550 ms) [282/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateMaxNegativeFails/1 (565 ms) [283/451] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetLocalAnswer/1 (565 ms) [284/451] PeerConnectionRtpTestUnifiedPlan.AddFirstTrackCreatesTransceiver (567 ms) [285/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreatePeerConnectionWithDifferentIceTransportsTypes/1 (598 ms) [286/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationChangesIceServers/1 (348 ms) [287/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.AddTrackWithoutStream/1 (427 ms) [288/451] HasStream/StatsCollectorTrackTest.BytesCounterHandles64Bits/0 (284 ms) [289/451] RTCStatsCollectorTest.CollectRTCCodecStats (326 ms) [290/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationReturnsSyntaxErrorFromBadIceUrls/1 (453 ms) [291/451] HasStream/StatsCollectorTrackTest.VideoBandwidthEstimationInfoIsReported/0 (358 ms) [292/451] RTCStatsCollectorTest.CollectRTCIceCandidateStats (358 ms) [293/451] RTCStatsCollectorTest.CollectNoStreamRTCOutboundRTPStreamStats_Audio (388 ms) [294/451] RtpParametersConversionTest.ToVideoCodec (325 ms) [295/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateMaxLessThanCurrentFails/1 (609 ms) [296/451] RtpSenderReceiverTest.AddAndDestroyVideoRtpReceiverWithStreams (312 ms) [297/451] RtpParametersConversionTest.ToCricketRtpHeaderExtensions (388 ms) [298/451] WebRtcSdpTest.DeserializeSessionDescriptionWithUfragPwd (299 ms) [299/451] SimulcastDeserializationErrors/SimulcastSdpSerializerTest.SimulcastDeserializationFailed/1 (256 ms) [300/451] DisableSimulcastLayersInSender/RtpSenderReceiverTest.DisableSimulcastLayersWithMediaEngine/1 (302 ms) [301/451] PeerConnectionMediaTest/PeerConnectionMediaTest.AnswerHasSameMidsAsOffer/1 (538 ms) [302/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateCurrentLessThanMinFails/1 (570 ms) [303/451] HasStream/StatsCollectorTrackTest.ReportsFromRemoteTrack/0 (335 ms) [304/451] SctpDataChannelTest.QueuedOpenMessageSent (317 ms) [305/451] HasStream/StatsCollectorTrackTest.GetStatsFromLocalAudioTrack/0 (366 ms) [306/451] SctpDataChannelTest.OpenWaitsForOpenMesssage (395 ms) [307/451] SctpDataChannelTest.ReceiveDataWithValidSsrc (407 ms) [308/451] SignalingProxyTest.Method1 (358 ms) [309/451] SignalingProxyTest.SignalingThreadDestructor (375 ms) [310/451] ProxyTest.ConstMethod1 (391 ms) [311/451] RTCStatsCollectorTest.CollectRemoteRTCMediaStreamStatsAndRTCMediaStreamTrackStats_Audio (297 ms) [312/451] RTCStatsTraversalTest.MultipleStartsLeadingToSameNode (282 ms) [313/451] PeerConnectionInterfaceTestPlanB.CreateOfferAnswerWithInvalidStream (533 ms) [314/451] RTCStatsCollectorTest.MultipleCallbacks (423 ms) [315/451] RtpParametersConversionTest.ToCricketCodecInvalidPayloadType (296 ms) [316/451] RtpParametersConversionTest.ToCricketStreamParamsVecNoEncodings (266 ms) [317/451] RtpSenderReceiverTest.AddAndDestroyVideoRtpSender (297 ms) [318/451] RtpSenderReceiverTest.VideoSenderCanSetFrameEncryptor (266 ms) [319/451] RtpSenderReceiverTest.VideoSenderCantSetReadOnlyEncodingParameters (358 ms) [320/451] RtpSenderReceiverTest.VideoSenderCheckTransactionIdRefresh (374 ms) [321/451] WebRtcSdpTest.DeserializeVideoFmtpWithSprops (233 ms) [322/451] RtpSenderReceiverTest.InternalParameterMethodsDoNotInvalidateTransaction (342 ms) [323/451] RtpSenderReceiverTest.AudioReceiverCanSetParameters (405 ms) [324/451] WebRtcSdpTest.IgnoreBundleOnlyWithNonzeroPort (280 ms) [325/451] WebRtcSdpTest.EmptyDescriptionHasNoMsidSignaling (233 ms) [326/451] WebRtcSdpTest.SerializeDtlsSetupAttribute (374 ms) [327/451] WebRtcSdpTest.DeserializeLargeBandwidthLimit (233 ms) [328/451] WebRtcSdpTest.DeserializeSimulcastNegative_EmptyAttribute (217 ms) [329/451] PeerConnectionMediaTest/PeerConnectionMediaTest.TestAVOfferWithAudioOnlyAnswer/1 (233 ms) [330/451] WebRtcSdpTest.ParseConnectionData (265 ms) [331/451] WebRtcSdpTest.DeserializeUnifiedPlanSessionDescription (404 ms) [332/451] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithSendSet/1 (250 ms) [333/451] DisableSimulcastLayersInSender/RtpSenderReceiverTest.DisableSimulcastLayersWithoutMediaEngine/1 (265 ms) [334/451] RtpTransceiverDirectionTest/EnumerateAllDirectionsTest.TestReversedIdentity/1 (312 ms) [335/451] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithRecvSet/5 (342 ms) [336/451] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/14 (296 ms) [337/451] SimulcastDeserializationErrors/SimulcastSdpSerializerTest.SimulcastDeserializationFailed/5 (390 ms) [338/451] SimulcastDeserializationErrors/SimulcastSdpSerializerTest.SimulcastDeserializationFailed/9 (375 ms) [339/451] HasStream/StatsCollectorTrackTest.TrackAndSsrcObjectExistAfterUpdateSsrcStats/0 (387 ms) [340/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/1 (812 ms) [341/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferFailsWithInvalidOfferToReceiveAudio/1 (625 ms) [342/451] PeerConnectionMediaTest/PeerConnectionMediaTest.SetRemoteDescriptionFailsWithDuplicateMids/1 (611 ms) [343/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferWithAudioVideoOptions/1 (657 ms) [344/451] PeerConnectionMediaTest/PeerConnectionMediaTest.CreateOfferWithNoVoiceActivityDetectionIncludesNoComfortNoiseCodecs/1 (563 ms) [345/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInOfferWhenDtlsDisabled/1 (563 ms) [346/451] PeerConnectionFactoryTest.CreatePCUsingIPLiteralAddress (427 ms) [347/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/9 (659 ms) [348/451] SctpDataChannelTest.OpenAckRoleInitialization (252 ms) [349/451] SctpDataChannelTest.SendEmptyData (234 ms) [350/451] JsepSessionDescriptionTest.SerializeSessionDescriptionWithTCPOnly (234 ms) [351/451] PeerConnectionFactoryTest.CheckRtpReceiverAudioCapabilities (282 ms) [352/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.StartAndStopLoggingToOutputAfterPeerConnectionClosed/1 (533 ms) [353/451] PeerConnectionRtpTestUnifiedPlan.PeerConnectionHasNoTransceivers (282 ms) [354/451] PeerConnectionRtpTestUnifiedPlan.RemoveTrackChangesDirectionFromSendRecvToRecvOnly (250 ms) [355/451] PeerConnectionRtpTestUnifiedPlan.AddTransceiverShowsInLists (250 ms) [356/451] PlayoutLatencyTest.GetLatency (204 ms) [357/451] RTCStatsCollectorTestWithFakeCollector.ThreadUsageAndResultsMerging (205 ms) [358/451] RtpSenderReceiverTest.RemoteAudioSourceLatency (205 ms) [359/451] RtpSenderReceiverTest.VideoSenderTrackSetToNull (217 ms) [360/451] RtpSenderReceiverTest.VideoSenderEarlyWarmupTrackThenSsrc (217 ms) [361/451] RtpSenderReceiverTest.AudioSenderCanSetFrameEncryptor (217 ms) [362/451] SimulcastSdpSerializerTest.Deserialize_OnlyReceiveStreams (203 ms) [363/451] SimulcastSdpSerializerTest.Serialize_OnlySend (219 ms) [364/451] RidDescriptionSdpSerializerTest.Serialize_FormatAndAttributeList (202 ms) [365/451] StatsCollectorTest.FilterOutNegativeDataChannelId (218 ms) [366/451] StatsCollectorTest.IceCandidateReport (218 ms) [367/451] WebRtcSdpTest.SerializeSessionDescriptionWithExtmap (218 ms) [368/451] WebRtcSdpTest.SerializeTcpCandidates (218 ms) [369/451] TrackMediaInfoMapTest.GetAttachmentIdByTrack (218 ms) [370/451] WebRtcSdpTest.UnifiedPlanDeserializeSessionDescriptionSpecialMsid (217 ms) [371/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/29 (202 ms) [372/451] RtpTransceiverDirectionTest/EnumerateAllDirectionsTest.TestIdentity/1 (203 ms) [373/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateSctpDataChannelShouldFailForInvalidConfig/1 (390 ms) [374/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationReturnsInvalidModificationError/1 (312 ms) [375/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationCreatesPooledSessionCorrectly/1 (345 ms) [376/451] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/10 (205 ms) [377/451] SctpDataChannelTest.QueuedDataSentWhenUnblocked (190 ms) [378/451] NumberOfSendEncodings/PeerConnectionSimulcastMetricsTests.NumberOfSendEncodingsIsLogged/5 (267 ms) [379/451] RTCStatsTraversalTest.Tree (1437 ms) [380/451] RtpSenderReceiverTest.AudioSenderSetParametersInvalidatesTransactionId (1623 ms) [381/451] RtpSenderReceiverTest.VideoSenderWithoutTrackAndSsrc (1623 ms) [382/451] RtpSenderReceiverTest.CanInsertDtmf (1621 ms) [383/451] RtpSenderReceiverTest.VideoSenderCanSetParameters (1652 ms) [384/451] RtpSenderReceiverTest.AudioSenderCantSetUnimplementedRtpParameters (1701 ms) [385/451] SimulcastSdpSerializerTest.Deserialize_SimpleCaseNoAlternatives (1715 ms) [386/451] WebRtcSdpTest.SerializeSessionDescriptionWithoutCandidates (467 ms) [387/451] TrackMediaInfoMapTest.SingleSenderReceiverPerTrackWithOneSsrc (1746 ms) [388/451] WebRtcSdpTest.DeserializeRawCandidateAttribute (358 ms) [389/451] WebRtcSdpTest.SerializeWithSctpDataChannelAndNewPort (358 ms) [390/451] WebRtcSdpTest.DeserializeSdpWithSctpDataChannelsWithSctpPort (389 ms) [391/451] WebRtcSdpTest.DeserializeSdpWithSctpDataChannelAndUnusualPort (406 ms) [392/451] WebRtcSdpTest.DeserializeSessionDescriptionWithoutEndLineBreak (358 ms) [393/451] WebRtcSdpTest.SerializeSdpWithConferenceFlag (374 ms) [394/451] WebRtcSdpTest.SerializeOnlyMediaSectionMsid (342 ms) [395/451] WebRtcSdpTest.DeserializeSerializeCodecParams (467 ms) [396/451] WebRtcSdpTest.ParseConnectionDataFailure (318 ms) [397/451] WebRtcSdpTest.TestDeserializeIgnoresDuplicateRidLines (310 ms) [398/451] WebRtcSdpTest.TestDeserializeSimulcastAttributeRemovesDuplicateSendReceive (355 ms) [399/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/29 (309 ms) [400/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.RtpDuplicatedLabelNotAllowed/1 (404 ms) [401/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/37 (325 ms) [402/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreatePeerConnectionAppliesNetworkConfigToPortAllocator/1 (450 ms) [403/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/41 (294 ms) [404/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/41 (280 ms) [405/451] RtpTransceiverDirectionTest/EnumerateAllDirectionsTest.TestReversedSwapped/1 (266 ms) [406/451] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithSendSet/5 (265 ms) [407/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/25 (398 ms) [408/451] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/6 (250 ms) [409/451] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithRecvSet/1 (279 ms) [410/451] HasStream/StatsCollectorTrackTest.GetStatsAfterRemoveAudioStream/0 (250 ms) [411/451] SctpSidAllocatorTest.SctpIdAllocationNoReuse (265 ms) [412/451] DtmfSenderTest.InsertDtmfTwice (265 ms) [413/451] SctpDataChannelTest.ConnectedToTransportOnCreated (358 ms) [414/451] IceServerParsingTest.ParseStunPrefixes (250 ms) [415/451] JsepSessionDescriptionTest.CheckSessionDescription (248 ms) [416/451] JsepSessionDescriptionTest.AddBadCandidate (249 ms) [417/451] MediaStreamTest.GetTrackInfo (249 ms) [418/451] RtpParametersConversionTest.ToRtcpFeedback (250 ms) [419/451] ProxyTest.VoidMethod0 (264 ms) [420/451] JsepSessionDescriptionTest.SerializeDeserializeWithCandidates (405 ms) [421/451] RtpParametersConversionTest.ToRtpEncodingsWithEmptyStreamParamsVec (280 ms) [422/451] RtpParametersConversionTest.ToRtpCodecCapabilityUnknownFeedbackParam (296 ms) [423/451] RtpSenderReceiverTest.LocalVideoTrackDisable (281 ms) [424/451] RtpSenderReceiverTest.AudioSenderCanSetParametersBeforeNegotiation (280 ms) [425/451] RtpSenderReceiverTest.VideoSenderMustCallGetParametersBeforeSetParametersBeforeNegotiation (265 ms) [426/451] RtpSenderReceiverTest.PropagatesVideoTrackContentHintForScreencastSource (250 ms) [427/451] RtpSenderReceiverTest.VideoSenderCanSetScaleResolutionDownBy (280 ms) [428/451] SctpUtilsTest.WriteParseOpenMessageWithMaxRetransmitTime (265 ms) [429/451] RidDescriptionSdpSerializerTest.Serialize_OnlyDirectionReceive (265 ms) [430/451] RidDescriptionSdpSerializerTest.Deserialize_WithFormat (266 ms) [431/451] RidDescriptionSdpSerializerTest.Deserialize_WithCustomRestrictions (264 ms) [432/451] StatsCollectorTest.UnsupportedDigestIgnored (265 ms) [433/451] TrackMediaInfoMapTest.MultipleOneSsrcSendersPerTrack (266 ms) [434/451] WebRtcSdpTest.SerializeSessionDescription (265 ms) [435/451] WebRtcSdpTest.SerializeSessionDescriptionWithVideoRejected (279 ms) [436/451] WebRtcSdpTest.DeserializeSessionDescriptionWithoutCarriageReturn (264 ms) [437/451] WebRtcSdpTest.DeserializeJsepSessionDescriptionWithFingerprint (264 ms) [438/451] WebRtcSdpTest.DeserializeSessionDescriptionWithRejectedAudio (264 ms) [439/451] WebRtcSdpTest.DeserializeSessionDescriptionWithExtmapAllowMixed (265 ms) [440/451] WebRtcSdpTest.AttributeWithPartialMatchingNameIsIgnored (248 ms) [441/451] WebRtcSdpTest.SerializeAudioFmtpWithPTimeAndMaxPTime (280 ms) [442/451] WebRtcSdpTest.ParseMediaTransportMalformedLine2 (250 ms) [443/451] JsepSessionDescriptionTest/EnumerateAllSdpTypesTest.TestIdentity/0 (265 ms) [444/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/25 (250 ms) [445/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/37 (249 ms) [446/451] WebRtcSdpTest.ParseMediaTransportInvalidBase64 (342 ms) [447/451] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/2 (233 ms) [448/451] HasStream/StatsCollectorTrackTest.TwoLocalTracksWithSameSsrc/0 (250 ms) [449/451] HasStream/StatsCollectorTrackTest.VerifyVideoReceiveSsrcStatsNew/0 (265 ms) [450/451] Encrypted/WebRtcSdpExtmapTest.DeserializeSessionDescriptionWithMediaLevelExtmap/0 (233 ms) [451/451] HasStream/StatsCollectorTrackTest.RemoteSsrcInfoIsAbsent/0 (218 ms) [452/451] DtmfSenderTest.InsertDtmfWithCommaAsDelay (264 ms) FAILED TESTS (1/451): 12536 ms: .\peerconnection_unittests.exe PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgrade/1 (try #1) 43bb190715c1b610: exit 0 Additional test environment: CHROME_DEVEL_SANDBOX=/opt/chromium/chrome_sandbox CHROME_HEADLESS=1 LANG=en_US.UTF-8 Command: e:\b\s\w\ir\.swarming_module_cache\vpython\37d31f\Scripts\python.exe ..\..\tools_webrtc\gtest-parallel-wrapper.py --output_dir=e:\b\s\w\io1wy5tq\test_logs --gtest_color=no --timeout=900 --retry_failed=3 .\peerconnection_unittests.exe --isolated-script-test-output=e:\b\s\w\io1wy5tq\output.json --isolated-script-test-perf-output=e:\b\s\w\io1wy5tq\perftest-output.json gtest-parallel-wrapper: Executing command e:\b\s\w\ir\.swarming_module_cache\vpython\37d31f\Scripts\python.exe e:\b\s\w\ir\third_party\gtest-parallel\gtest-parallel --shard_index=3 --shard_count=4 --output_dir=e:\b\s\w\io1wy5tq\test_logs --retry_failed=3 --gtest_color=no --timeout=900 --dump_json_test_results=e:\b\s\w\io1wy5tq\output.json .\peerconnection_unittests.exe -- --isolated-script-test-perf-output=e:\b\s\w\io1wy5tq\perftest-output.json [0/451] Running tests... [1/451] PeerConnectionIceTest/PeerConnectionIceTest.LocalCandidatesNotRemovedWhenNetworkDownIfGatheringOnce/0 (1712 ms) [2/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AddRtpDataChannelInSubsequentOffer/0 (2904 ms) [3/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithDtls/0 (4039 ms) [4/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndConnectionTimeWithTurnTurnPair/0 (3189 ms) [5/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.CallTransferredForCaller/0 (5098 ms) [6/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.StressTestUnorderedSctpDataChannel/0 (2627 ms) [7/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.DisableAndEnableAudioPlayout/0 (2247 ms) [8/451] RTCStatsIntegrationTest.GetStatsFromCaller (1455 ms) [9/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.NewGetStatsManyAudioAndManyVideoStreams/0 (1299 ms) [10/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.DisableAndEnableAudioRecording/0 (1845 ms) [11/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges/0 (1583 ms) [12/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaContinuesFlowingAfterIceRestart/0 (1500 ms) [13/451] RTCStatsIntegrationTest.GetStatsWithInvalidSenderSelector (1677 ms) [14/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithoutSsrcOrMsidSignaling/0 (1044 ms) [15/451] PeerConnectionIceTest/PeerConnectionIceTest.NoIceCandidatesBeforeAnswerSetAsLocalDescription/0 (1688 ms) [16/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.TCPUsedForTurnConnections/0 (1450 ms) [17/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AddAudioToVideoOnlyCall/0 (1500 ms) [18/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetBytesSentStatsWithOldStatsApi/0 (923 ms) [19/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.SSLCertificateVerifierFailureUsedForTurnConnectionsFailsConnection/0 (10690 ms) [20/451] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.Call/0 (1835 ms) [21/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithRtpDataChannel/0 (1473 ms) [22/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.VerifyBestConnection/4 (1676 ms) [23/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.RtcEventLogOutputWriteCalled/0 (1626 ms) [24/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.RtpReceiverObserverOnFirstPacketReceived/0 (1450 ms) [25/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.OneWayMediaCall/0 (1430 ms) [26/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.Dtls10CipherStatsAndUmaMetrics/0 (1172 ms) [27/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.CallerDtls12ToCalleeDtls10/0 (1628 ms) [28/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.SctpDataChannelToAudioVideoUpgrade/0 (2334 ms) [29/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.IceStatesReachCompletion/0 (1461 ms) [30/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.Aes128Sha1_32_CipherNotUsedWhenOnlyCallerSupported/0 (1737 ms) [31/451] PeerConnectionIntegrationTestPlanB.EnableAudioAfterRejecting (1763 ms) [32/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportBidirectionalVideo/0 (1005 ms) [33/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationInteropTest.OneAudioOneVideoLocalToOneAudioOneVideoRemote/0 (1614 ms) [34/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.CalleeClosesSctpDataChannel/0 (1851 ms) [35/451] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.CallWithCustomCodec/0 (1743 ms) [36/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.EndToEndCallWithSpecCompliantMaxBundleOffer/0 (1702 ms) [37/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.RotatedVideoWithCVOExtension/0 (1463 ms) [38/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AddMediaToConnectedBundleDoesNotRestartIce/0 (1252 ms) [39/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GcmCipherUsedWhenGcmSupported/0 (2227 ms) [40/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AnswererRejectsAudioSection/0 (2022 ms) [41/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.VerifyBestConnection/0 (1451 ms) [42/451] PeerConnectionIntegrationTestPlanB.TwoVideoUnifiedPlanToNoMediaPlanB (1510 ms) [43/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.Aes128Sha1_32_CipherUsedWhenSupported/0 (1849 ms) [44/451] PeerConnectionUsageHistogramTest.FingerprintDataOnly (1423 ms) [45/451] PeerConnectionJsepTest.ReOfferMediaSectionForAssociatedStoppedTransceiverIsRejected (746 ms) [46/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/18 (983 ms) [47/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.NonGcmCipherUsedWhenOnlyCalleeSupportsGcm/0 (1603 ms) [48/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetSourcesAudio/0 (894 ms) [49/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetStatsForUnsignaledStreamWithOldStatsApi/0 (1063 ms) [50/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/14 (1504 ms) [51/451] PeerConnectionIceTest/PeerConnectionIceTest.VerifyUfragPwdLength/0 (1221 ms) [52/451] PeerConnectionIceTest/PeerConnectionIceTest.CreateAnswerHasNewUfragPwdForOnlyMediaSectionWhichRestarted/0 (825 ms) [53/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/14 (1192 ms) [54/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.ExchangeOfferAnswerWhenDtlsCertificateInConfig/0 (711 ms) [55/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetAudioInputLevelStatsWithOldStatsApi/0 (1414 ms) [56/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.IceStateSetupFailure/2 (22106 ms) [57/451] PeerConnectionBundleTest/PeerConnectionBundleMatrixTest.VerifyTransportsBeforeAndAfterSettingRemoteAnswer/4 (973 ms) [58/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/6 (875 ms) [59/451] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate/0 (556 ms) [60/451] PeerConnectionRtpTest/PeerConnectionRtpTest.RemoveTrackWithStreamFiresOnRemoveTrack/0 (728 ms) [61/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/2 (1043 ms) [62/451] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.DataChannelFromOpenWorksAfterPreviousChannelClosed/0 (993 ms) [63/451] PeerConnectionMediaTest/PeerConnectionMediaTest.TestAVOfferWithVideoOnlyAnswer/0 (671 ms) [64/451] PeerConnectionBundleTest/PeerConnectionBundleTest.IgnoreCandidatesForUnusedTransportWhenBundling/0 (1223 ms) [65/451] PeerConnectionRtpTest/PeerConnectionRtpTest.RemoveTrackWithoutStreamRemovesReceiver/0 (667 ms) [66/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/18 (894 ms) [67/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/14 (791 ms) [68/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/18 (717 ms) [69/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/18 (965 ms) [70/451] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.CloseDataChannelRemotelyWhileNotReferenced/0 (916 ms) [71/451] PeerConnectionRtpTestUnifiedPlan.SetDirectionCallsOnTrack (624 ms) [72/451] PeerConnectionSimulcastMetricsTests.SpecComplianceIsLogged (511 ms) [73/451] PeerConnectionIceTest/PeerConnectionIceTest.RemoveCandidateRemovesFromRemoteDescription/0 (926 ms) [74/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/14 (930 ms) [75/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetRemoteAnswerWithNoFingerprintWhenDtlsOn/0 (692 ms) [76/451] PeerConnectionRtpTestUnifiedPlan.AddRemoveAddTrackGeneratesNewSenderId (1043 ms) [77/451] PeerConnectionRtpTest/PeerConnectionRtpTest.RemoveTrackWithSharedStreamRemovesReceiver/0 (840 ms) [78/451] PeerConnectionIceTest/PeerConnectionIceTest.AnswerContainsGatheredCandidates/0 (962 ms) [79/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.ReceiveUpdatedAudioOfferWithBadCodecs/0 (666 ms) [80/451] PeerConnectionMediaTest/PeerConnectionMediaTest.CombinedAudioVideoBweConfigPropagatedToMediaEngine/0 (624 ms) [81/451] PeerConnectionJsepTest.CreateOfferRecyclesWhenOfferingTwice (798 ms) [82/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferWithDefaultOfferAnswerOptions/0 (728 ms) [83/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/2 (659 ms) [84/451] PeerConnectionIntegrationTestUnifiedPlan.NoStreamsMsidLineMissing (961 ms) [85/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/0 (11786 ms) Error: unrecognized flag --isolated-script-test-perf-output=e:\b\s\w\io1wy5tq\perftest-output.json Note: Google Test filter = PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/0 [==========] Running 1 test from 1 test suite. [----------] Global test environment set-up. [----------] 1 test from PeerConnectionIntegrationTest/PeerConnectionIntegrationTest [ RUN ] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/0 [000:000] (loopback_media_transport.cc:156): LoopbackMediaTransport [000:000] (loopback_media_transport.cc:156): LoopbackMediaTransport [000:002] (audio_processing_impl.cc:421): Capture analyzer activated: 0 Capture post processor activated: 0 Render pre processor activated: 0 [000:003] (webrtc_video_engine.cc:460): WebRtcVideoEngine::WebRtcVideoEngine() [000:003] (webrtc_voice_engine.cc:194): WebRtcVoiceEngine::WebRtcVoiceEngine [000:044] (webrtc_voice_engine.cc:217): WebRtcVoiceEngine::Init [000:064] (webrtc_voice_engine.cc:224): Supported send codecs in order of preference: [000:067] (webrtc_voice_engine.cc:227): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:067] (webrtc_voice_engine.cc:227): ISAC/16000/1 (103) [000:067] (webrtc_voice_engine.cc:227): ISAC/32000/1 (104) [000:067] (webrtc_voice_engine.cc:227): G722/8000/1 (9) [000:067] (webrtc_voice_engine.cc:227): ILBC/8000/1 (102) [000:067] (webrtc_voice_engine.cc:227): PCMU/8000/1 (0) [000:067] (webrtc_voice_engine.cc:227): PCMA/8000/1 (8) [000:067] (webrtc_voice_engine.cc:227): CN/32000/1 (106) [000:067] (webrtc_voice_engine.cc:227): CN/16000/1 (105) [000:068] (webrtc_voice_engine.cc:227): CN/8000/1 (13) [000:068] (webrtc_voice_engine.cc:227): telephone-event/48000/1 (110) [000:068] (webrtc_voice_engine.cc:227): telephone-event/32000/1 (112) [000:068] (webrtc_voice_engine.cc:227): telephone-event/16000/1 (113) [000:068] (webrtc_voice_engine.cc:227): telephone-event/8000/1 (126) [000:068] (webrtc_voice_engine.cc:230): Supported recv codecs in order of preference: [000:070] (webrtc_voice_engine.cc:233): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:070] (webrtc_voice_engine.cc:233): ISAC/16000/1 (103) [000:070] (webrtc_voice_engine.cc:233): ISAC/32000/1 (104) [000:071] (webrtc_voice_engine.cc:233): G722/8000/1 (9) [000:071] (webrtc_voice_engine.cc:233): ILBC/8000/1 (102) [000:071] (webrtc_voice_engine.cc:233): PCMU/8000/1 (0) [000:071] (webrtc_voice_engine.cc:233): PCMA/8000/1 (8) [000:071] (webrtc_voice_engine.cc:233): CN/32000/1 (106) [000:071] (webrtc_voice_engine.cc:233): CN/16000/1 (105) [000:071] (webrtc_voice_engine.cc:233): CN/8000/1 (13) [000:071] (webrtc_voice_engine.cc:233): telephone-event/48000/1 (110) [000:071] (webrtc_voice_engine.cc:233): telephone-event/32000/1 (112) [000:071] (webrtc_voice_engine.cc:233): telephone-event/16000/1 (113) [000:071] (webrtc_voice_engine.cc:233): telephone-event/8000/1 (126) [000:072] (webrtc_voice_engine.cc:310): 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, } [000:074] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [000:074] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [000:074] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [000:074] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [000:074] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [000:074] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [000:075] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:076] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:076] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:076] (apm_helpers.cc:90): Echo control set to 1 with mode 0 [000:076] (apm_helpers.cc:79): AGC set to 1 with mode 0 [000:076] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:077] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:077] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:077] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:077] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:077] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:077] (apm_helpers.cc:104): NS set to 1 [000:077] (webrtc_voice_engine.cc:466): Stereo swapping enabled? 0 [000:077] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:078] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:078] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:078] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:078] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:078] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:078] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:078] (webrtc_voice_engine.cc:536): Typing detection is enabled? 1 [000:078] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:079] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:082] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:164] (rtc_event_log_impl.cc:72): Creating legacy encoder for RTC event log. [000:165] (peer_connection_factory.cc:395): Using default network controller factory [000:166] (bitrate_prober.cc:62): Bandwidth probing enabled, set to inactive [000:167] (paced_sender.cc:427): ProcessThreadAttached 0x1e6ebb0 [000:168] (cpu_info.cc:49): Available number of cores: 8 [000:169] (aimd_rate_control.cc:101): Using aimd rate control with back off factor 0.85 [000:169] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating. [000:228] (audio_processing_impl.cc:421): Capture analyzer activated: 0 Capture post processor activated: 0 Render pre processor activated: 0 [000:229] (webrtc_video_engine.cc:460): WebRtcVideoEngine::WebRtcVideoEngine() [000:229] (webrtc_voice_engine.cc:194): WebRtcVoiceEngine::WebRtcVoiceEngine [000:260] (webrtc_voice_engine.cc:217): WebRtcVoiceEngine::Init [000:277] (webrtc_voice_engine.cc:224): Supported send codecs in order of preference: [000:280] (webrtc_voice_engine.cc:227): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:280] (webrtc_voice_engine.cc:227): ISAC/16000/1 (103) [000:280] (webrtc_voice_engine.cc:227): ISAC/32000/1 (104) [000:280] (webrtc_voice_engine.cc:227): G722/8000/1 (9) [000:280] (webrtc_voice_engine.cc:227): ILBC/8000/1 (102) [000:280] (webrtc_voice_engine.cc:227): PCMU/8000/1 (0) [000:280] (webrtc_voice_engine.cc:227): PCMA/8000/1 (8) [000:280] (webrtc_voice_engine.cc:227): CN/32000/1 (106) [000:280] (webrtc_voice_engine.cc:227): CN/16000/1 (105) [000:280] (webrtc_voice_engine.cc:227): CN/8000/1 (13) [000:281] (webrtc_voice_engine.cc:227): telephone-event/48000/1 (110) [000:281] (webrtc_voice_engine.cc:227): telephone-event/32000/1 (112) [000:281] (webrtc_voice_engine.cc:227): telephone-event/16000/1 (113) [000:281] (webrtc_voice_engine.cc:227): telephone-event/8000/1 (126) [000:281] (webrtc_voice_engine.cc:230): Supported recv codecs in order of preference: [000:283] (webrtc_voice_engine.cc:233): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:283] (webrtc_voice_engine.cc:233): ISAC/16000/1 (103) [000:283] (webrtc_voice_engine.cc:233): ISAC/32000/1 (104) [000:283] (webrtc_voice_engine.cc:233): G722/8000/1 (9) [000:283] (webrtc_voice_engine.cc:233): ILBC/8000/1 (102) [000:283] (webrtc_voice_engine.cc:233): PCMU/8000/1 (0) [000:283] (webrtc_voice_engine.cc:233): PCMA/8000/1 (8) [000:284] (webrtc_voice_engine.cc:233): CN/32000/1 (106) [000:284] (webrtc_voice_engine.cc:233): CN/16000/1 (105) [000:284] (webrtc_voice_engine.cc:233): CN/8000/1 (13) [000:284] (webrtc_voice_engine.cc:233): telephone-event/48000/1 (110) [000:284] (webrtc_voice_engine.cc:233): telephone-event/32000/1 (112) [000:284] (webrtc_voice_engine.cc:233): telephone-event/16000/1 (113) [000:284] (webrtc_voice_engine.cc:233): telephone-event/8000/1 (126) [000:284] (webrtc_voice_engine.cc:310): 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, } [000:286] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [000:286] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [000:286] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [000:286] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [000:286] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [000:286] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [000:288] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:288] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:288] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:288] (apm_helpers.cc:90): Echo control set to 1 with mode 0 [000:288] (apm_helpers.cc:79): AGC set to 1 with mode 0 [000:288] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:289] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:289] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:289] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:289] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:289] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:289] (apm_helpers.cc:104): NS set to 1 [000:289] (webrtc_voice_engine.cc:466): Stereo swapping enabled? 0 [000:289] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:289] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:289] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:289] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:289] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:290] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:290] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:290] (webrtc_voice_engine.cc:536): Typing detection is enabled? 1 [000:290] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:290] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:290] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:305] (rtc_event_log_impl.cc:72): Creating legacy encoder for RTC event log. [000:305] (peer_connection_factory.cc:395): Using default network controller factory [000:305] (bitrate_prober.cc:62): Bandwidth probing enabled, set to inactive [000:312] (paced_sender.cc:427): ProcessThreadAttached 0x1f660d0 [000:313] (aimd_rate_control.cc:101): Using aimd rate control with back off factor 0.85 [000:313] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating. [000:354] (jsep_transport_controller.cc:1518): Generating media transport offer! [000:355] (media_session.cc:335): Duplicate id found. Reassigning from 102 to 127 [000:355] (media_session.cc:335): Duplicate id found. Reassigning from 104 to 125 [000:355] (media_session.cc:335): Duplicate id found. Reassigning from 103 to 124 [000:355] (media_session.cc:335): Duplicate id found. Reassigning from 1 to 14 [000:355] (media_session.cc:335): Duplicate id found. Reassigning from 2 to 13 [000:358] (peer_connection_integrationtest.cc:753): Caller: SetLocalDescriptionAndSendSdpMessage [000:363] (jsep_transport_controller.cc:975): Offered media transport has now been activated. [000:363] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [000:365] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [000:366] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [000:366] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [000:370] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [000:370] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [000:371] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [000:372] (p2p_transport_channel.cc:432): Set ICE ufrag: Y4Rs pwd: pb71CRQM6CWAnEUFdyOhP+tP on transport video [000:373] (webrtc_video_engine.cc:472): CreateMediaChannel. Options: VideoOptions {} [000:379] (channel.cc:144): Created channel for video [000:381] (channel.cc:212): BaseChannel::Init_w, media_transport=1 [000:381] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:381] (peer_connection.cc:4097): Session: 6831287259824631555 Old state: kStable New state: kHaveLocalOffer [000:386] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:387] (channel.cc:1009): Setting local video description [000:390] (webrtc_video_engine.cc:945): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 3}, {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: 5}, {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}]} [000:392] (webrtc_video_engine.cc:960): Changing recv codecs from {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} to {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} [000:394] (webrtc_video_engine.cc:1061): AddSendStream: {id:cf3d8d56-49d4-4ca3-a6cd-abc6011a8389;ssrcs:[3680665287,3230205430];ssrc_groups:{semantics:FID;ssrcs:[3680665287,3230205430]};cname:KXFA9ah7fNMadlij;stream_ids:2da8937e-a99d-40dc-876b-1ea61492ba89;} [000:394] (webrtc_video_engine.cc:1100): SetLocalSsrc on all the receive streams because we added a send stream. [000:394] (channel.cc:676): Add send stream ssrc: 3680665287 [000:400] (channel.cc:996): Changing video state, send=0 [000:400] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 3680665287, options: VideoOptions {is_screencast : false, }, source = (source)) [000:403] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [000:403] (peer_connection_integrationtest.cc:683): Callee: HandleIncomingOffer [000:403] (basic_port_allocator.cc:851): Network manager has started [000:403] (basic_port_allocator.cc:766): Allocate ports on 1 networks [000:403] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:404] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Udp [000:404] (port.cc:318): Port[2c8a2b0::1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port created with network cost 50 [000:405] (basic_port_allocator.cc:875): Adding allocated port for video [000:406] (basic_port_allocator.cc:894): Port[2c8a2b0:video:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Added port to allocator [000:407] (peer_connection_integrationtest.cc:771): Callee: SetRemoteDescription [000:408] (basic_port_allocator.cc:912): Port[2c8a2b0:video:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Gathered candidate: Cand[:2214029314:1:udp:2122260223:192.168.1.1:49152:local::0:Y4Rs:pb71CRQM6CWAnEUFdyOhP+tP:1:50:0] [000:409] (basic_port_allocator.cc:940): Port[2c8a2b0:video:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port ready. [000:409] (basic_port_allocator.cc:1017): Port[2c8a2b0:video:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port completed gathering candidates. [000:410] (jsep_transport_controller.cc:996): Returning new, client media transport. [000:410] (p2p_transport_channel.cc:495): Set backup connection ping interval to 25000 milliseconds. [000:411] (p2p_transport_channel.cc:504): Set ICE receiving timeout to 2500 milliseconds [000:411] (p2p_transport_channel.cc:511): Set ping most likely connection to 0 [000:411] (p2p_transport_channel.cc:518): Set stable_writable_connection_ping_interval to 2500 [000:412] (p2p_transport_channel.cc:531): Set presume writable when fully relayed to 0 [000:413] (p2p_transport_channel.cc:540): Set regather_on_failed_networks_interval to 300000 [000:413] (p2p_transport_channel.cc:559): Set receiving_switching_delay to 1000 [000:414] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=Y4Rs, renomination disabled [000:415] (webrtc_video_engine.cc:472): CreateMediaChannel. Options: VideoOptions {} [000:417] (channel.cc:144): Created channel for video [000:419] (channel.cc:212): BaseChannel::Init_w, media_transport=1 [000:420] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:420] (peer_connection.cc:4097): Session: 7628613091676420589 Old state: kStable New state: kHaveRemoteOffer [000:420] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:421] (channel.cc:1061): Setting remote video description [000:421] (webrtc_video_engine.cc:670): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 3}, {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: 5}, {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}], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: video} [000:423] (webrtc_video_engine.cc:679): Using codec: VideoCodec[96:VP8] [000:423] (webrtc_video_engine.cc:747): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. [000:424] (webrtc_video_engine.cc:1163): AddRecvStream: {id:cf3d8d56-49d4-4ca3-a6cd-abc6011a8389;ssrcs:[3680665287,3230205430];ssrc_groups:{semantics:FID;ssrcs:[3680665287,3230205430]};cname:KXFA9ah7fNMadlij;stream_ids:2da8937e-a99d-40dc-876b-1ea61492ba89;} [000:424] (bitrate_allocator.cc:115): Current BWE 300000 [000:426] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 3680665287, 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: 104, red_type: 102, rtx_ssrc: 3230205430, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 103 (pt) -> 102 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: 2da8937e-a99d-40dc-876b-1ea61492ba89, target_delay_ms: 0} [000:426] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:426] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:427] (channel.cc:719): Add remote ssrc: 3680665287 [000:430] (channel.cc:996): Changing video state, send=0 [000:431] (peer_connection.cc:4794): Creating video receiver for track_id=cf3d8d56-49d4-4ca3-a6cd-abc6011a8389 and stream_id=2da8937e-a99d-40dc-876b-1ea61492ba89 [000:431] (webrtc_video_engine.cc:1293): SetSink: ssrc:3680665287 (ptr) [000:436] (peer_connection_integrationtest.cc:919): Caller: OnIceCandidate [000:436] (peer_connection_integrationtest.cc:867): Callee: ReceiveIceMessage [000:437] (peer_connection.cc:6141): video is not ready to use the remote candidate because the local or remote description is not set. [000:439] (peer_connection.cc:5449): Local and Remote descriptions must be applied to get the SSL Role of the session. [000:445] (peer_connection_integrationtest.cc:753): Callee: SetLocalDescriptionAndSendSdpMessage [000:472] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [000:472] (p2p_transport_channel.cc:432): Set ICE ufrag: CkGj pwd: aLVah5ZwHgDBq/fjXXV7Kjht on transport video [000:472] (dtls_transport.cc:253): DtlsTransport[video|1|__]: Other side didn't support DTLS. [000:476] (channel.cc:553): Channel enabled [000:478] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Relay [000:482] (channel.cc:996): Changing video state, send=0 [000:482] (peer_connection.cc:4097): Session: 7628613091676420589 Old state: kHaveRemoteOffer New state: kStable [000:482] (channel.cc:1009): Setting local video description [000:485] (webrtc_video_engine.cc:945): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 3}, {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: 5}, {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}]} [000:487] (webrtc_video_engine.cc:960): Changing recv codecs from {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} to {VideoCodec[96:VP8], VideoCodec[98:VP9], VideoCodec[100:VP9]} [000:487] (webrtc_video_engine.cc:2493): MaybeRecreateWebRtcFlexfecStream (recv) because of SetRecvParameters [000:487] (webrtc_video_engine.cc:2498): RecreateWebRtcVideoStream (recv) because of SetRecvParameters [000:487] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:488] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:488] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 3680665287, 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: 104, red_type: 102, rtx_ssrc: 3230205430, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 103 (pt) -> 102 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: 2da8937e-a99d-40dc-876b-1ea61492ba89, target_delay_ms: 0} [000:488] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [000:489] (receive_statistics_proxy.cc:487): Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [000:491] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 3680665287, 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: 125, red_type: 127, rtx_ssrc: 3230205430, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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: 5}, {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: 3}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: 2da8937e-a99d-40dc-876b-1ea61492ba89, target_delay_ms: 0} [000:491] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:491] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:504] (webrtc_video_engine.cc:1061): AddSendStream: {id:c712f9a2-02f7-4be6-a5aa-ef4f2633ada3;ssrcs:[3046216017,3833851890];ssrc_groups:{semantics:FID;ssrcs:[3046216017,3833851890]};cname:gCILwhwe1giyItyF;stream_ids:8928845d-cbdc-48ef-bf07-433ec7a574c9;} [000:504] (webrtc_video_engine.cc:1859): RecreateWebRtcStream (send) because of SetCodec. [000:505] (alr_experiment.cc:66): 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 [000:510] (video_send_stream_impl.cc:249): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [3046216017], 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: 5}, {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: 3}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [3833851890], payload_type: 97}, c_name: gCILwhwe1giyItyF}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [000:510] (video_send_stream_impl.cc:275): ERROR: Initial encoder max bitrate = -1 which is <= 0! [000:510] (video_send_stream_impl.cc:277): Using default encoder max bitrate = 10 Mbps [000:511] (video_stream_encoder.cc:579): ConfigureEncoder requested. [000:511] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:511] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:511] (video_send_stream.cc:160): VideoSendStream::Stop [000:511] (video_send_stream_impl.cc:415): VideoSendStream::Stop [000:511] (webrtc_video_engine.cc:1100): SetLocalSsrc on all the receive streams because we added a send stream. [000:511] (webrtc_video_engine.cc:2434): RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc=3046216017 [000:512] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:512] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:512] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 3680665287, 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: 125, red_type: 127, rtx_ssrc: 3230205430, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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: 5}, {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: 3}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: 2da8937e-a99d-40dc-876b-1ea61492ba89, target_delay_ms: 0} [000:515] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [000:515] (receive_statistics_proxy.cc:487): Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [000:517] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 3680665287, local_ssrc: 3046216017, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 3230205430, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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: 5}, {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: 3}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: 2da8937e-a99d-40dc-876b-1ea61492ba89, target_delay_ms: 0} [000:517] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:517] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:519] (channel.cc:676): Add send stream ssrc: 3046216017 [000:520] (video_send_stream.cc:160): VideoSendStream::Stop [000:520] (channel.cc:996): Changing video state, send=0 [000:523] (video_send_stream_impl.cc:415): VideoSendStream::Stop [000:523] (peer_connection.cc:4033): Changing IceConnectionState 0 => 1 [000:528] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Tcp [000:528] (port.cc:318): Port[2d34a50::1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port created with network cost 50 [000:528] (basic_port_allocator.cc:875): Adding allocated port for video [000:528] (basic_port_allocator.cc:894): Port[2d34a50:video:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Added port to allocator [000:529] (basic_port_allocator.cc:912): Port[2d34a50:video:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Gathered candidate: Cand[:3446969586:1:tcp:1518280447:192.168.1.1:49153:local::0:Y4Rs:pb71CRQM6CWAnEUFdyOhP+tP:1:50:0] [000:529] (basic_port_allocator.cc:940): Port[2d34a50:video:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port ready. [000:529] (basic_port_allocator.cc:1017): Port[2d34a50:video:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port completed gathering candidates. [000:529] (basic_port_allocator.cc:1119): All candidates gathered for video:1:0 [000:529] (p2p_transport_channel.cc:854): P2PTransportChannel: video, component 1 gathering complete [000:529] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 3046216017, options: VideoOptions {is_screencast : false, }, source = (source)) [000:530] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [000:530] (basic_port_allocator.cc:851): Network manager has started [000:530] (basic_port_allocator.cc:766): Allocate ports on 1 networks [000:530] (basic_port_allocator.cc:1318): Net[test0:192.168.1.0/24:Unknown:id=1]: Allocation Phase=Udp [000:530] (port.cc:318): Port[2d45b90::1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port created with network cost 50 [000:531] (basic_port_allocator.cc:875): Adding allocated port for video [000:531] (basic_port_allocator.cc:894): Port[2d45b90:video:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Added port to allocator [000:531] (basic_port_allocator.cc:912): Port[2d45b90:video:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Gathered candidate: Cand[:2214029314:1:udp:2122260223:192.168.1.1:49154:local::0:CkGj:aLVah5ZwHgDBq/fjXXV7Kjht:1:50:0] [000:531] (basic_port_allocator.cc:940): Port[2d45b90:video:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port ready. [000:531] (port.cc:1126): Conn[2d460b0:video:Net[test0:192.168.1.0/24:Unknown:id=1]:FGH8X0LP:1:0:local:udp:192.168.1.1:49154->sOuIMTJw:1:2122260223:local:udp:192.168.1.1:49152|C--W|-|0|0|9115038255631187454|-]: Connection created [000:531] (p2p_transport_channel.cc:1264): Channel[video|1|__]: Created connection with origin: 2, total: 1 [000:531] (video_stream_encoder.cc:579): ConfigureEncoder requested. [000:532] (p2p_transport_channel.cc:1914): Channel[video|1|__]: Transport channel state changed from 0 to 2 [000:532] (jsep_transport_controller.cc:1277): video Transport 1 state changed. Check if state is complete. [000:532] (jsep_transport_controller.cc:1277): video Transport 1 state changed. Check if state is complete. [000:532] (p2p_transport_channel.cc:1476): Channel[video|1|__]: Have a pingable connection for the first time; starting to ping. [000:532] (basic_port_allocator.cc:1017): Port[2d45b90:video:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Port completed gathering candidates. [000:533] (port.cc:1756): Conn[2d460b0:video:Net[test0:192.168.1.0/24:Unknown:id=1]:FGH8X0LP:1:0:local:udp:192.168.1.1:49154->sOuIMTJw:1:2122260223:local:udp:192.168.1.1:49152|C--W|-|0|0|9115038255631187454|-]: Sent STUN ping, id=6c715342545951374c796f4f, use_candidate=0, nomination=0 [000:533] (port.cc:528): Received STUN ping id=6c715342545951374c796f4f from unknown address 192.168.1.1:49154 [000:533] (port.cc:1126): Conn[2d47020:video:Net[test0:192.168.1.0/24:Unknown:id=1]:2ktfLGJJ:1:0:local:udp:192.168.1.1:49152->JIOfI5/s:1:1853824767:prflx:udp:192.168.1.1:49154|C--W|-|0|0|7962116751024340479|-]: Connection created [000:533] (p2p_transport_channel.cc:988): Adding connection from peer reflexive candidate: Cand[:4215022485:1:udp:1853824767:192.168.1.1:49154:prflx::0:CkGj::1:50:0] [000:533] (peer_connection_integrationtest.cc:703): Caller: HandleIncomingAnswer [000:534] (port.cc:853): Port[2c8a2b0:video:1:0:local:Net[test0:192.168.1.0/24:Unknown:id=1]]: Sent STUN ping response, to=192.168.1.1:49154, id=6c715342545951374c796f4f [000:534] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:534] (p2p_transport_channel.cc:1914): Channel[video|1|R_]: Transport channel state changed from 0 to 2 [000:534] (jsep_transport_controller.cc:1277): video Transport 1 state changed. Check if state is complete. [000:534] (jsep_transport_controller.cc:1277): video Transport 1 state changed. Check if state is complete. [000:534] (port.cc:1704): Conn[2d460b0:video:Net[test0:192.168.1.0/24:Unknown:id=1]:FGH8X0LP:1:0:local:udp:192.168.1.1:49154->sOuIMTJw:1:2122260223:local:udp:192.168.1.1:49152|C--I|-|0|0|9115038255631187454|-]: Received STUN ping response, id=6c715342545951374c796f4f, code=0, rtt=2, pings_since_last_response=6c715342545951374c796f4f [000:535] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed [000:535] (p2p_transport_channel.cc:1852): Channel[video|1|__]: New selected connection: Conn[2d460b0:video:Net[test0:192.168.1.0/24:Unknown:id=1]:FGH8X0LP:1:0:local:udp:192.168.1.1:49154->sOuIMTJw:1:2122260223:local:udp:192.168.1.1:49152|CRWS|S|0|0|9115038255631187454|2] [000:535] (jsep_transport_controller.cc:1214): Transport video writability changed to 1. [000:535] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:535] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:535] (channel.cc:583): Channel writable (video) for the first time [000:535] (jsep_transport_controller.cc:1277): video Transport 1 state changed. Check if state is complete. [000:536] (basic_port_allocator.cc:1119): All candidates gathered for video:1:0 [000:536] (p2p_transport_channel.cc:854): P2PTransportChannel: video, component 1 gathering complete [000:536] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:536] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:536] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps [000:536] (video_stream_encoder.cc:1477): Video suspend state changed to: not suspended [000:536] (channel.cc:996): Changing video state, send=1 [000:541] (peer_connection_integrationtest.cc:771): Caller: SetRemoteDescription [000:545] (webrtc_video_engine.cc:1338): Call stats: 6568861, {send_bw_bps: 0, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1} [000:546] (srtp_transport.cc:309): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [000:546] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=CkGj, renomination disabled [000:546] (dtls_transport.cc:253): DtlsTransport[video|1|R_]: Other side didn't support DTLS. [000:547] (p2p_transport_channel.cc:1476): Channel[video|1|R_]: Have a pingable connection for the first time; starting to ping. [000:547] (p2p_transport_channel.cc:2440): Selecting connection for triggered check: Conn[2d47020:video:Net[test0:192.168.1.0/24:Unknown:id=1]:2ktfLGJJ:1:0:local:udp:192.168.1.1:49152->JIOfI5/s:1:1853824767:prflx:udp:192.168.1.1:49154|CR-W|-|0|0|7962116751024340479|-] [000:547] (channel.cc:553): Channel enabled [000:547] (port.cc:1756): Conn[2d47020:video:Net[test0:192.168.1.0/24:Unknown:id=1]:2ktfLGJJ:1:0:local:udp:192.168.1.1:49152->JIOfI5/s:1:1853824767:prflx:udp:192.168.1.1:49154|CR-W|-|0|0|7962116751024340479|-]: Sent STUN ping, id=447770504751306463306546, use_candidate=1, nomination=0 [000:547] (channel.cc:996): Changing video state, send=0 [000:548] (peer_connection.cc:4097): Session: 6831287259824631555 Old state: kHaveLocalOffer New state: kStable [000:548] (channel.cc:1061): Setting remote video description [000:548] (port.cc:1704): Conn[2d47020:video:Net[test0:192.168.1.0/24:Unknown:id=1]:2ktfLGJJ:1:0:local:udp:192.168.1.1:49152->JIOfI5/s:1:1853824767:prflx:udp:192.168.1.1:49154|CR-I|-|0|0|7962116751024340479|-]: Received STUN ping response, id=447770504751306463306546, code=0, rtt=1, pings_since_last_response=447770504751306463306546 [000:548] (webrtc_video_engine.cc:670): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 3}, {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: 5}, {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}], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: video} [000:548] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed [000:548] (p2p_transport_channel.cc:1852): Channel[video|1|R_]: New selected connection: Conn[2d47020:video:Net[test0:192.168.1.0/24:Unknown:id=1]:2ktfLGJJ:1:0:local:udp:192.168.1.1:49152->JIOfI5/s:1:1853824767:prflx:udp:192.168.1.1:49154|CRWS|S|0|0|7962116751024340479|1] [000:549] (jsep_transport_controller.cc:1214): Transport video writability changed to 1. [000:549] (channel.cc:583): Channel writable (video) for the first time [000:549] (jsep_transport_controller.cc:1277): video Transport 1 state changed. Check if state is complete. [000:549] (webrtc_video_engine.cc:679): Using codec: VideoCodec[96:VP8] [000:549] (webrtc_video_engine.cc:1859): RecreateWebRtcStream (send) because of SetCodec. [000:550] (alr_experiment.cc:66): 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 [000:551] (video_stream_encoder.cc:1051): Video frame parameters changed: dimensions=640x480, texture=0. [000:551] (video_stream_encoder.cc:233): Set max framerate: 60 [000:555] (bitrate_allocator.cc:115): Current BWE 300000 [000:559] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 30000bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 1700000bps [000:578] (video_send_stream_impl.cc:249): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [3680665287], 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: 5}, {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: 3}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [3230205430], payload_type: 97}, c_name: KXFA9ah7fNMadlij}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [000:578] (video_send_stream_impl.cc:275): ERROR: Initial encoder max bitrate = -1 which is <= 0! [000:578] (video_send_stream_impl.cc:277): Using default encoder max bitrate = 10 Mbps [000:579] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:579] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:580] (video_send_stream.cc:160): VideoSendStream::Stop [000:580] (webrtc_video_engine.cc:747): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. [000:580] (webrtc_video_engine.cc:1163): AddRecvStream: {id:c712f9a2-02f7-4be6-a5aa-ef4f2633ada3;ssrcs:[3046216017,3833851890];ssrc_groups:{semantics:FID;ssrcs:[3046216017,3833851890]};cname:gCILwhwe1giyItyF;stream_ids:8928845d-cbdc-48ef-bf07-433ec7a574c9;} [000:581] (video_receive_stream.cc:209): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 3046216017, local_ssrc: 3680665287, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 3833851890, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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: 5}, {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: 3}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: 8928845d-cbdc-48ef-bf07-433ec7a574c9, target_delay_ms: 0} [000:581] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [000:581] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [000:582] (video_stream_encoder.cc:579): ConfigureEncoder requested. [000:582] (video_send_stream_impl.cc:415): VideoSendStream::Stop [000:592] (channel.cc:719): Add remote ssrc: 3046216017 [000:597] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:597] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:597] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps [000:598] (video_stream_encoder.cc:1477): Video suspend state changed to: not suspended [000:598] (channel.cc:996): Changing video state, send=1 [000:598] (message_queue.cc:518): Message took 50ms to dispatch. Posted from: cricket::BaseChannel::SetRemoteContent@../../pc/channel.cc:302 [000:598] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:598] (peer_connection.cc:4033): Changing IceConnectionState 0 => 1 [000:598] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:598] (peer_connection.cc:4794): Creating video receiver for track_id=c712f9a2-02f7-4be6-a5aa-ef4f2633ada3 and stream_id=8928845d-cbdc-48ef-bf07-433ec7a574c9 [000:599] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:600] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:600] (channel.cc:996): Changing video state, send=1 [000:600] (webrtc_video_engine.cc:1293): SetSink: ssrc:3046216017 (ptr) [000:601] (peer_connection_integrationtest.cc:919): Caller: OnIceCandidate [000:601] (peer_connection_integrationtest.cc:867): Callee: ReceiveIceMessage [000:605] (peer_connection_integrationtest.cc:919): Callee: OnIceCandidate [000:605] (peer_connection_integrationtest.cc:867): Caller: ReceiveIceMessage [000:616] (jsep_transport_controller.cc:1513): Not regenerating media transport for the new offer in existing session. [000:622] (peer_connection.cc:5981): Changing to ICE completed state because all transports are complete. [000:623] (peer_connection.cc:4033): Changing IceConnectionState 1 => 2 [000:623] (peer_connection.cc:4033): Changing IceConnectionState 2 => 3 [000:628] (peer_connection.cc:5975): Changing to ICE connected state because all transports are writable. [000:628] (peer_connection.cc:4033): Changing IceConnectionState 1 => 2 [000:630] (peer_connection_integrationtest.cc:753): Callee: SetLocalDescriptionAndSendSdpMessage [000:636] (webrtc_video_engine.cc:2205): VideoSendStream stats: 6568952, {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: 3046216017, width: 640, height: 480, 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} [000:636] (webrtc_video_engine.cc:2685): VideoReceiveStream stats: 6568952, {ssrc: 3680665287, 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, discarded: 0, sync_offset_ms: 2147483647, cum_loss: 0, max_ext_seq: 0, nack: 0, fir: 0, pli: 0} [000:638] (webrtc_video_engine.cc:1338): Call stats: 6568952, {send_bw_bps: 300000, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1} [000:640] (p2p_transport_channel.cc:432): Set ICE ufrag: CkGj pwd: aLVah5ZwHgDBq/fjXXV7Kjht on transport video [000:641] (webrtc_voice_engine.cc:1488): 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, } [000:641] (webrtc_voice_engine.cc:310): 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, } [000:641] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:641] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:641] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:641] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:641] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:642] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:642] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:642] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:642] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:642] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:642] (webrtc_voice_engine.cc:1506): 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, } [000:642] (channel.cc:144): Created channel for audio [000:643] (channel.cc:583): Channel writable (audio) for the first time [000:650] (channel.cc:212): BaseChannel::Init_w, media_transport=1 [000:650] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:650] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:650] (peer_connection.cc:4097): Session: 7628613091676420589 Old state: kStable New state: kHaveLocalOffer [000:650] (channel.cc:863): Changing voice state, recv=0 send=0 [000:651] (channel.cc:871): Setting local voice description [000:651] (webrtc_voice_engine.cc:1327): 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}]} [000:651] (webrtc_voice_engine.cc:1516): Setting receive voice codecs. [000:654] (webrtc_voice_engine.cc:1787): AddSendStream: {id:38fa1ad4-3149-4d09-bbbc-73f1ab3a014e;ssrcs:[3882877651];ssrc_groups:;cname:gCILwhwe1giyItyF;stream_ids:04905f70-f4b4-4e6c-9ecf-09bb271a73fd;} [000:660] (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 [000:660] (audio_coding_module.cc:333): Created [000:661] (channel_send.cc:704): Setting media_transport_ rate observers. [000:661] (audio_send_stream.cc:142): AudioSendStream: 3882877651 [000:661] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 3882877651, extmap-allow-mixed: false, extensions: [], c_name: gCILwhwe1giyItyF}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: <unset>} [000:661] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:661] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:662] (channel.cc:676): Add send stream ssrc: 3882877651 [000:662] (channel_send.cc:816): Trying to call unset encoder. [000:663] (channel.cc:863): Changing voice state, recv=0 send=0 [000:663] (channel.cc:1009): Setting local video description [000:664] (video_stream_encoder.cc:1051): Video frame parameters changed: dimensions=640x480, texture=0. [000:664] (video_stream_encoder.cc:233): Set max framerate: 60 [000:675] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 30000bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 1700000bps [000:686] (webrtc_video_engine.cc:945): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 3}, {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: 5}, {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}]} [000:691] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:692] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:692] (channel.cc:996): Changing video state, send=1 [000:695] (webrtc_voice_engine.cc:1488): Setting voice channel options: AudioOptions {hf: 0, } [000:695] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {hf: 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, } [000:695] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:695] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:695] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:695] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:695] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:695] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:695] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:695] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:695] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:695] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:696] (webrtc_voice_engine.cc:1506): Set voice channel options. Current options: AudioOptions {hf: 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, } [000:696] (peer_connection_integrationtest.cc:683): Caller: HandleIncomingOffer [000:696] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:700] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:702] (peer_connection_integrationtest.cc:771): Caller: SetRemoteDescription [000:704] (decoder_database.cc:140): Initializing decoder with payload type '96'. [000:719] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=CkGj, renomination disabled [000:719] (webrtc_voice_engine.cc:1488): 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, } [000:719] (webrtc_voice_engine.cc:310): 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, } [000:719] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:719] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:719] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:719] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:719] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:720] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:720] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:720] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:720] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:720] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:720] (webrtc_voice_engine.cc:1506): 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, } [000:720] (channel.cc:144): Created channel for audio [000:720] (channel.cc:583): Channel writable (audio) for the first time [000:721] (channel.cc:212): BaseChannel::Init_w, media_transport=1 [000:721] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:721] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:721] (channel.cc:863): Changing voice state, recv=0 send=0 [000:721] (peer_connection.cc:4097): Session: 6831287259824631555 Old state: kStable New state: kHaveRemoteOffer [000:722] (channel.cc:923): Setting remote voice description [000:722] (webrtc_voice_engine.cc:1282): 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}], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: audio, options: AudioOptions {}} [000:723] (webrtc_voice_engine.cc:1705): Recreate all the receive streams because the send codec has changed. [000:723] (webrtc_voice_engine.cc:2166): WebRtcVoiceMediaChannel::SetMaxSendBitrate. [000:723] (webrtc_voice_engine.cc:1488): Setting voice channel options: AudioOptions {} [000:723] (webrtc_voice_engine.cc:310): 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, } [000:723] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:723] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:723] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:723] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:723] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:723] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:723] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:723] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:724] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:724] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:724] (webrtc_voice_engine.cc:1506): 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, } [000:724] (webrtc_voice_engine.cc:1852): AddRecvStream: {id:38fa1ad4-3149-4d09-bbbc-73f1ab3a014e;ssrcs:[3882877651];ssrc_groups:;cname:gCILwhwe1giyItyF;stream_ids:04905f70-f4b4-4e6c-9ecf-09bb271a73fd;} [000:724] (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 [000:724] (audio_coding_module.cc:333): Created [000:725] (audio_receive_stream.cc:117): AudioReceiveStream: 3882877651 [000:725] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3882877651, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: (Transport), sync_group: 04905f70-f4b4-4e6c-9ecf-09bb271a73fd} [000:725] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:725] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:725] (channel.cc:719): Add remote ssrc: 3882877651 [000:732] (channel.cc:863): Changing voice state, recv=0 send=0 [000:732] (channel.cc:1061): Setting remote video description [000:733] (webrtc_video_engine.cc:670): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 3}, {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: 5}, {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}], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: video} [000:737] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:737] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:738] (channel.cc:996): Changing video state, send=1 [000:739] (peer_connection.cc:4794): Creating audio receiver for track_id=38fa1ad4-3149-4d09-bbbc-73f1ab3a014e and stream_id=04905f70-f4b4-4e6c-9ecf-09bb271a73fd [000:739] (webrtc_voice_engine.cc:1965): SetOutputVolume() to 1 for recv stream with ssrc 3882877651 [000:753] (peer_connection_integrationtest.cc:753): Caller: SetLocalDescriptionAndSendSdpMessage [000:784] (srtp_filter.cc:194): Applying the same SRTP send parameters again. No-op. [000:784] (srtp_filter.cc:225): Applying the same SRTP recv parameters again. No-op. [000:785] (srtp_transport.cc:309): SRTP updated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [000:789] (p2p_transport_channel.cc:432): Set ICE ufrag: Y4Rs pwd: pb71CRQM6CWAnEUFdyOhP+tP on transport video [000:796] (dtls_transport.cc:253): DtlsTransport[video|1|RW]: Other side didn't support DTLS. [000:799] (channel.cc:553): Channel enabled [000:799] (channel.cc:863): Changing voice state, recv=0 send=0 [000:799] (peer_connection.cc:4097): Session: 6831287259824631555 Old state: kHaveRemoteOffer New state: kStable [000:803] (channel.cc:871): Setting local voice description [000:806] (webrtc_voice_engine.cc:1327): 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}]} [000:806] (webrtc_voice_engine.cc:1516): Setting receive voice codecs. [000:807] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3882877651, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: (Transport), sync_group: 04905f70-f4b4-4e6c-9ecf-09bb271a73fd} [000:807] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:808] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:808] (audio_receive_stream.cc:140): ~AudioReceiveStream: 3882877651 [000:808] (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 [000:808] (audio_coding_module.cc:333): Created [000:808] (audio_receive_stream.cc:117): AudioReceiveStream: 3882877651 [000:808] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3882877651, 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:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), media_transport: (Transport), sync_group: 04905f70-f4b4-4e6c-9ecf-09bb271a73fd} [000:809] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:809] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:816] (webrtc_voice_engine.cc:1787): AddSendStream: {id:5d5434cf-ae15-4d9d-b019-5d090c2e3408;ssrcs:[894518488];ssrc_groups:;cname:KXFA9ah7fNMadlij;stream_ids:996776ce-a139-43b1-a9bc-0036aeeff2f8;} [000:879] (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 [000:896] (audio_coding_module.cc:333): Created [000:902] (channel_send.cc:704): Setting media_transport_ rate observers. [000:914] (audio_send_stream.cc:142): AudioSendStream: 894518488 [000:924] (channel_send.cc:816): Trying to call unset encoder. [000:926] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 894518488, 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:ssrc-audio-level, id: 1}], c_name: KXFA9ah7fNMadlij}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), 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}}}} [000:931] (audio_encoder_opus.cc:772): Set Opus bitrate to 32000 bps. [000:933] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [000:934] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [000:935] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 3882877651, local_ssrc: 894518488, 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:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), media_transport: (Transport), sync_group: 04905f70-f4b4-4e6c-9ecf-09bb271a73fd} [000:935] (channel.cc:676): Add send stream ssrc: 894518488 [000:941] (webrtc_voice_engine.cc:310): 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, } [000:943] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:945] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:945] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:945] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:945] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:946] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:946] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:946] (audio_processing_impl.cc:696): Highpass filter activated: 1 [000:947] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:948] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:952] (channel.cc:863): Changing voice state, recv=1 send=1 [000:962] (message_queue.cc:518): Message took 159ms to dispatch. Posted from: cricket::BaseChannel::SetLocalContent@../../pc/channel.cc:293 [000:963] (channel.cc:1009): Setting local video description [000:972] (webrtc_video_engine.cc:945): SetRecvParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 3}, {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: 5}, {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}]} [000:976] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [000:976] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [000:976] (channel.cc:996): Changing video state, send=1 [000:992] (webrtc_voice_engine.cc:1488): Setting voice channel options: AudioOptions {hf: 0, } [000:992] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {hf: 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, } [000:992] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [000:992] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:992] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:992] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:993] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:993] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:993] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:993] (audio_processing_impl.cc:696): Highpass filter activated: 0 [000:993] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [000:993] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [000:993] (webrtc_voice_engine.cc:1506): Set voice channel options. Current options: AudioOptions {hf: 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, } [001:003] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [001:003] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [001:003] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [001:003] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [001:003] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [001:003] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [001:005] (render_delay_buffer.cc:306): Receiving a first externally reported audio buffer delay of 0 ms. [001:005] (agc_manager_direct.cc:383): [agc] Initial GetMicVolume()=0 [001:005] (agc_manager_direct.cc:388): [agc] Initial volume too low, raising to 12 [001:019] (render_delay_buffer.cc:334): Applying total delay of 1 blocks. [001:020] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 1: 2 blocks [001:020] (echo_remover.cc:213): Gain change detected at block 1 [001:021] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 2: 3 blocks [001:021] (peer_connection_integrationtest.cc:703): Callee: HandleIncomingAnswer [001:021] (render_delay_buffer.cc:255): Render buffer underrun detected at block 2 [001:021] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [001:022] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 3: 4 blocks [001:022] (render_delay_buffer.cc:255): Render buffer underrun detected at block 3 [001:025] (webrtc_sdp.cc:3221): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [001:027] (peer_connection_integrationtest.cc:771): Callee: SetRemoteDescription [001:042] (srtp_filter.cc:194): Applying the same SRTP send parameters again. No-op. [001:042] (srtp_filter.cc:225): Applying the same SRTP recv parameters again. No-op. [001:042] (srtp_transport.cc:309): SRTP updated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [001:042] (p2p_transport_channel.cc:443): Received remote ICE parameters: ufrag=Y4Rs, renomination disabled [001:042] (dtls_transport.cc:253): DtlsTransport[video|1|RW]: Other side didn't support DTLS. [001:044] (channel.cc:553): Channel enabled [001:044] (channel.cc:863): Changing voice state, recv=1 send=0 [001:045] (peer_connection.cc:4097): Session: 7628613091676420589 Old state: kHaveLocalOffer New state: kStable [001:046] (channel.cc:923): Setting remote voice description [001:046] (webrtc_voice_engine.cc:1282): 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}], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: audio, options: AudioOptions {}} [001:046] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 3882877651, extmap-allow-mixed: false, extensions: [], c_name: gCILwhwe1giyItyF}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), 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}}}} [001:047] (audio_encoder_opus.cc:772): Set Opus bitrate to 32000 bps. [001:047] (webrtc_voice_engine.cc:1705): Recreate all the receive streams because the send codec has changed. [001:047] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 3882877651, 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:ssrc-audio-level, id: 1}], c_name: gCILwhwe1giyItyF}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), 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}}}} [001:047] (audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 3882877651, 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:ssrc-audio-level, id: 1}], c_name: gCILwhwe1giyItyF}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: (Transport), 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}}}} [001:047] (webrtc_voice_engine.cc:2166): WebRtcVoiceMediaChannel::SetMaxSendBitrate. [001:047] (webrtc_voice_engine.cc:1488): Setting voice channel options: AudioOptions {} [001:047] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {hf: 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, } [001:047] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [001:047] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [001:047] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [001:047] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [001:047] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [001:047] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [001:047] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [001:047] (audio_processing_impl.cc:696): Highpass filter activated: 0 [001:048] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [001:048] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [001:048] (webrtc_voice_engine.cc:1506): Set voice channel options. Current options: AudioOptions {hf: 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, } [001:048] (webrtc_voice_engine.cc:1852): AddRecvStream: {id:5d5434cf-ae15-4d9d-b019-5d090c2e3408;ssrcs:[894518488];ssrc_groups:;cname:KXFA9ah7fNMadlij;stream_ids:996776ce-a139-43b1-a9bc-0036aeeff2f8;} [001:048] (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 [001:048] (audio_coding_module.cc:333): Created [001:049] (audio_receive_stream.cc:117): AudioReceiveStream: 894518488 [001:049] (audio_receive_stream.cc:352): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 894518488, local_ssrc: 3882877651, 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:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), media_transport: (Transport), sync_group: 996776ce-a139-43b1-a9bc-0036aeeff2f8} [001:049] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [001:049] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [001:058] (channel.cc:719): Add remote ssrc: 894518488 [001:060] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {hf: 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, } [001:060] (webrtc_voice_engine.cc:471): NetEq capacity is 200 [001:060] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [001:060] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [001:060] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [001:060] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [001:060] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [001:060] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [001:060] (audio_processing_impl.cc:696): Highpass filter activated: 0 [001:061] (audio_processing_impl.cc:710): Gain Controller 2 activated: 0 [001:061] (audio_processing_impl.cc:712): Pre-amplifier activated: 0 [001:061] (channel.cc:863): Changing voice state, recv=1 send=1 [001:061] (channel.cc:1061): Setting remote video description [001:062] (webrtc_video_engine.cc:670): SetSendParameters: {codecs: [VideoCodec[96:VP8], VideoCodec[97:rtx], VideoCodec[98:VP9], VideoCodec[99:rtx], VideoCodec[100:VP9], VideoCodec[101:rtx], VideoCodec[127:red], VideoCodec[124:rtx], VideoCodec[125: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: 3}, {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: 5}, {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}], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: video} [001:073] (video_send_stream.cc:133): VideoSendStream::UpdateActiveSimulcastLayers [001:073] (video_send_stream_impl.cc:362): VideoSendStream::UpdateActiveSimulcastLayers [001:073] (channel.cc:996): Changing video state, send=1 [001:083] (peer_connection.cc:4794): Creating audio receiver for track_id=5d5434cf-ae15-4d9d-b019-5d090c2e3408 and stream_id=996776ce-a139-43b1-a9bc-0036aeeff2f8 [001:088] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [001:088] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [001:088] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [001:088] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [001:088] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [001:088] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [001:089] (webrtc_voice_engine.cc:1965): SetOutputVolume() to 1 for recv stream with ssrc 894518488 [001:092] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [001:092] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [001:092] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [001:092] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [001:092] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [001:092] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [001:094] (render_delay_buffer.cc:306): Receiving a first externally reported audio buffer delay of 0 ms. [001:094] (agc_manager_direct.cc:383): [agc] Initial GetMicVolume()=0 [001:095] (agc_manager_direct.cc:388): [agc] Initial volume too low, raising to 12 [001:103] (render_delay_buffer.cc:334): Applying total delay of 1 blocks. [001:104] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 1: 2 blocks [001:104] (echo_remover.cc:213): Gain change detected at block 1 [001:104] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 2: 3 blocks [001:104] (render_delay_buffer.cc:255): Render buffer underrun detected at block 2 [001:106] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 3: 4 blocks [001:106] (render_delay_buffer.cc:255): Render buffer underrun detected at block 3 [001:116] (render_delay_buffer.cc:334): Applying total delay of 5 blocks. [001:116] (matched_filter.cc:450): Filter 0: start: 0 ms, end: 128 ms. [001:116] (matched_filter.cc:450): Filter 1: start: 96 ms, end: 224 ms. [001:116] (matched_filter.cc:450): Filter 2: start: 192 ms, end: 320 ms. [001:116] (matched_filter.cc:450): Filter 3: start: 288 ms, end: 416 ms. [001:116] (matched_filter.cc:450): Filter 4: start: 384 ms, end: 512 ms. [001:120] (render_delay_buffer.cc:306): Receiving a first externally reported audio buffer delay of 0 ms. [001:120] (render_delay_buffer.cc:334): Applying total delay of 1 blocks. [001:120] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 1: 2 blocks [001:121] (render_delay_buffer.cc:237): New max number api jitter observed at capture block 2: 3 blocks [001:121] (render_delay_buffer.cc:255): Render buffer underrun detected at block 2 [001:121] (agc_manager_direct.cc:383): [agc] Initial GetMicVolume()=12 [003:484] (message_queue.cc:518): Message took 116ms to dispatch. Posted from: webrtc::MediaTransportPair::LoopbackMediaTransport::SendAudioFrame@../../api/test/loopback_media_transport.cc:182 ../../pc/peer_connection_integrationtest.cc(1502): error: Value of: caller()->audio_frames_received() >= total_caller_audio_frames_expected && caller()->min_video_frames_received_per_track() >= total_caller_video_frames_expected && callee()->audio_frames_received() >= total_callee_audio_frames_expected && callee()->min_video_frames_received_per_track() >= total_callee_video_frames_expected Actual: false Expected: true Stack trace: 000000013FB26B10: testing::DefaultValue<rtc::AsyncResolverInterface * __ptr64>::Exists 000000013FB7AAB9: webrtc::VP9EncoderImpl::PopulateCodecSpecific 000000014042625C: testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void> 00000001404260FB: testing::internal::HandleExceptionsInMethodIfSupported<testing::Test,void> 00000001404495DA: testing::Test::Run 0000000140449725: testing::TestInfo::Run ... Google Test internal frames ... ../../pc/peer_connection_integrationtest.cc(1518): error: Expected: (caller()->min_video_frames_received_per_track()) >= (total_caller_video_frames_expected), actual: 0 vs 3 Stack trace: 000000013FB26D66: testing::DefaultValue<rtc::AsyncResolverInterface * __ptr64>::Exists 000000013FB7AAB9: webrtc::VP9EncoderImpl::PopulateCodecSpecific 000000014042625C: testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void> 00000001404260FB: testing::internal::HandleExceptionsInMethodIfSupported<testing::Test,void> 00000001404495DA: testing::Test::Run 0000000140449725: testing::TestInfo::Run ... Google Test internal frames ... ../../pc/peer_connection_integrationtest.cc(3652): error: Value of: ExpectNewFrames(media_expectations) Actual: false Expected: true Stack trace: 000000013FB7AB6E: webrtc::VP9EncoderImpl::PopulateCodecSpecific 000000014042625C: testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void> 00000001404260FB: testing::internal::HandleExceptionsInMethodIfSupported<testing::Test,void> 00000001404495DA: testing::Test::Run 0000000140449725: testing::TestInfo::Run ... Google Test internal frames ... [011:172] (webrtc_voice_engine.cc:1965): SetOutputVolume() to 0 for recv stream with ssrc 3882877651 [011:173] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 3680665287, options: nullptr, source = nullptr) [011:173] (webrtc_video_engine.cc:1293): SetSink: ssrc:3046216017 nullptr [011:174] (channel.cc:563): Channel disabled [011:174] (video_send_stream.cc:160): VideoSendStream::Stop [011:175] (channel.cc:996): Changing video state, send=0 [011:175] (video_send_stream.cc:160): VideoSendStream::Stop [011:175] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:175] (rtp_rtcp_impl.cc:359): Failed to send RTCP BYE [011:175] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps [011:176] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:187] (video_stream_encoder.cc:1477): Video suspend state changed to: suspended [011:190] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:190] (video_send_stream_impl.cc:337): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [3680665287], 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: 5}, {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: 3}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [3230205430], payload_type: 97}, c_name: KXFA9ah7fNMadlij}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [011:191] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:191] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:192] (send_statistics_proxy.cc:654): Frames encoded 106 WebRTC.Video.DroppedFrames.Capturer 0 WebRTC.Video.DroppedFrames.EncoderQueue 0 WebRTC.Video.DroppedFrames.Encoder 0 WebRTC.Video.DroppedFrames.Ratelimiter 0 [011:192] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:192] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:192] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 3046216017, local_ssrc: 3680665287, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 3833851890, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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: 5}, {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: 3}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: 8928845d-cbdc-48ef-bf07-433ec7a574c9, target_delay_ms: 0} [011:197] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [011:211] (receive_statistics_proxy.cc:487): WebRTC.Video.ReceiveStreamLifetimeInSeconds 10 Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [011:211] (channel.cc:162): Destroyed channel: video [011:212] (channel.cc:563): Channel disabled [011:214] (channel.cc:863): Changing voice state, recv=0 send=0 [011:224] (webrtc_voice_engine.cc:1826): RemoveSendStream: 894518488 [011:224] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:224] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:224] (audio_send_stream.cc:179): ~AudioSendStream: 894518488 [011:227] (webrtc_voice_engine.cc:1903): RemoveRecvStream: 3882877651 [011:227] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [011:227] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [011:227] (audio_receive_stream.cc:140): ~AudioReceiveStream: 3882877651 [011:228] (paced_sender.cc:115): PacedSender paused. [011:228] (channel.cc:162): Destroyed channel: audio [011:228] (peer_connection.cc:901): Session: 6831287259824631555 is destroyed. [011:236] (paced_sender.cc:427): ProcessThreadAttached 0x0 [011:236] (paced_sender.cc:427): ProcessThreadAttached 0x0 [011:237] (rtc_event_log_impl.cc:212): Stopping WebRTC event log. [011:238] (rtc_event_log_impl.cc:229): WebRTC event log successfully stopped. [011:240] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue. [011:242] (webrtc_video_engine.cc:464): WebRtcVideoEngine::~WebRtcVideoEngine [011:243] (webrtc_voice_engine.cc:203): WebRtcVoiceEngine::~WebRtcVoiceEngine [011:262] (webrtc_voice_engine.cc:1965): SetOutputVolume() to 0 for recv stream with ssrc 894518488 [011:263] (webrtc_video_engine.cc:1020): SetVideoSend (ssrc= 3046216017, options: nullptr, source = nullptr) [011:263] (webrtc_video_engine.cc:1293): SetSink: ssrc:3680665287 nullptr [011:264] (channel.cc:563): Channel disabled [011:268] (video_send_stream.cc:160): VideoSendStream::Stop [011:268] (channel.cc:996): Changing video state, send=0 [011:268] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:269] (rtp_rtcp_impl.cc:359): Failed to send RTCP BYE [011:269] (video_send_stream.cc:160): VideoSendStream::Stop [011:269] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps [011:275] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:275] (video_stream_encoder.cc:1477): Video suspend state changed to: suspended [011:275] (video_send_stream_impl.cc:415): VideoSendStream::Stop [011:275] (video_send_stream_impl.cc:337): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [3046216017], 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: 5}, {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: 3}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 125, red_payload_type: 127, red_rtx_payload_type: 124}, payload_name: VP8, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [3833851890], payload_type: 97}, c_name: gCILwhwe1giyItyF}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: (Transport), render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [011:276] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:276] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:276] (send_statistics_proxy.cc:654): Frames encoded 108 WebRTC.Video.DroppedFrames.Capturer 0 WebRTC.Video.DroppedFrames.EncoderQueue 0 WebRTC.Video.DroppedFrames.Encoder 0 WebRTC.Video.DroppedFrames.Ratelimiter 0 [011:276] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:276] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:277] (video_receive_stream.cc:278): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: VP8, codec_params: {}}, {payload_type: 98, payload_name: VP9, codec_params: {profile-id: 0}}, {payload_type: 100, payload_name: VP9, codec_params: {profile-id: 2}}], rtp: {remote_ssrc: 3680665287, local_ssrc: 3046216017, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 125, red_type: 127, rtx_ssrc: 3230205430, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 124 (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: 5}, {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: 3}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 14}]}, renderer: (renderer), render_delay_ms: 10, sync_group: 2da8937e-a99d-40dc-876b-1ea61492ba89, target_delay_ms: 0} [011:278] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [011:279] (receive_statistics_proxy.cc:487): WebRTC.Video.ReceiveStreamLifetimeInSeconds 10 Frames decoded 106 WebRTC.Video.DroppedFrames.Receiver -106 WebRTC.Video.InterframeDelay95PercentileInMs 137 WebRTC.Video.InterframeDelay95PercentileInMs.S0 137 WebRTC.Video.MediaBitrateReceivedInKbps.S0 7 [011:279] (video_quality_observer.cc:135): WebRTC.Video.MeanTimeBetweenFreezesMs 10466 WebRTC.Video.TimeInHdPercentage 0 WebRTC.Video.TimeInBlockyVideoPercentage 0 WebRTC.Video.NumberResolutionDownswitchesPerMinute 0 WebRTC.Video.NumberFreezesPerMinute 0 WebRTC.Video.HarmonicFrameRate 9 [011:279] (channel.cc:162): Destroyed channel: video [011:280] (channel.cc:563): Channel disabled [011:280] (channel.cc:863): Changing voice state, recv=0 send=0 [011:284] (webrtc_voice_engine.cc:1826): RemoveSendStream: 3882877651 [011:284] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=up [011:284] (rtp_transport_controller_send.cc:259): SignalNetworkState Up [011:284] (audio_send_stream.cc:179): ~AudioSendStream: 3882877651 [011:286] (webrtc_voice_engine.cc:1903): RemoveRecvStream: 894518488 [011:286] (call.cc:1168): UpdateAggregateNetworkState: aggregate_state=down [011:286] (rtp_transport_controller_send.cc:259): SignalNetworkState Down [011:286] (audio_receive_stream.cc:140): ~AudioReceiveStream: 894518488 [011:286] (paced_sender.cc:115): PacedSender paused. [011:287] (channel.cc:162): Destroyed channel: audio [011:287] (peer_connection.cc:901): Session: 7628613091676420589 is destroyed. [011:297] (paced_sender.cc:427): ProcessThreadAttached 0x0 [011:297] (paced_sender.cc:427): ProcessThreadAttached 0x0 [011:298] (rtc_event_log_impl.cc:212): Stopping WebRTC event log. [011:298] (rtc_event_log_impl.cc:229): WebRTC event log successfully stopped. [011:299] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue. [011:300] (webrtc_video_engine.cc:464): WebRtcVideoEngine::~WebRtcVideoEngine [011:300] (webrtc_voice_engine.cc:203): WebRtcVoiceEngine::~WebRtcVoiceEngine [011:303] (loopback_media_transport.cc:160): ~LoopbackMediaTransport [011:303] (loopback_media_transport.cc:160): ~LoopbackMediaTransport [ FAILED ] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/0, where GetParam() = 4-byte object <00-00 00-00> (11307 ms) [----------] 1 test from PeerConnectionIntegrationTest/PeerConnectionIntegrationTest (11307 ms total) [----------] Global test environment tear-down [==========] 1 test from 1 test suite ran. (11308 ms total) [ PASSED ] 0 tests. [ FAILED ] 1 test, listed below: [ FAILED ] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/0, where GetParam() = 4-byte object <00-00 00-00> 1 FAILED TEST [85/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/0 returned/aborted with exit code 1 (11786 ms) [86/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/18 (570 ms) [87/451] PeerConnectionEndToEndTest/PeerConnectionEndToEndTest.DataChannelIdAssignment/0 (716 ms) [88/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/10 (378 ms) [89/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/14 (463 ms) [90/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/18 (713 ms) [91/451] PeerConnectionJsepTest.SetRemoteOfferDoesNotReuseStoppedTransceiver (470 ms) [92/451] PeerConnectionIceTest/PeerConnectionIceTest.OfferFromLiteIceControllingAndAnswerFromLiteIceControlled/0 (661 ms) [93/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/2 (721 ms) [94/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/6 (825 ms) [95/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/10 (439 ms) [96/451] PeerConnectionJsepTest.SetRemoteOfferWithOneTrackOneStreamFiresOnAddTrack (640 ms) [97/451] PeerConnectionMediaConfigTest.TestCreateAndClose (571 ms) [98/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.AnswererRejectsAudioAndVideoSections/0 (1290 ms) [99/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationCausingPartialIceRestart/0 (775 ms) [100/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/0 (1614 ms) [101/451] PeerConnectionInterfaceTestPlanB.AddLocalStreamAfterLocalDescriptionChanged (651 ms) [102/451] WebRtcSdpTest.TestDeserializeIgnoresEmptyRidLines (466 ms) [103/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/14 (1055 ms) [104/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/2 (766 ms) [105/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalPrAnswer/10 (911 ms) [106/451] PeerConnectionRtpTestUnifiedPlan.CheckForUnsupportedEncodingParameters (456 ms) [107/451] HasStream/StatsCollectorTrackTest.GetStatsAfterRemoveAudioStream/1 (292 ms) [108/451] PeerConnectionBundleTest/PeerConnectionBundleMatrixTest.VerifyTransportsBeforeAndAfterSettingRemoteAnswer/8 (882 ms) [109/451] HasStream/StatsCollectorTrackTest.BytesCounterHandles64Bits/1 (314 ms) [110/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.TransportStatsReportedForDataChannelOnlyConnection/0 (1032 ms) [111/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.GetStatsForSpecificTrack/0 (594 ms) [112/451] PeerConnectionSignalingUnifiedPlanTest.DtlsTransportsInstantiateInOfferAnswer (624 ms) [113/451] PeerConnectionInterfaceTestPlanB.VerifyDefaultStreamIsNotCreated (493 ms) [114/451] PeerConnectionBundleTest/PeerConnectionBundleTest.ApplyDescriptionWithConflictedDemuxCriteriaFail/0 (621 ms) [115/451] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.RtpDataChannelCreatedEvenIfSctpAvailable/0 (528 ms) [116/451] PeerConnectionSimulcastTests.ParametersAreInvalidatedWhenLayersChange (603 ms) [117/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/14 (778 ms) [118/451] PeerConnectionJsepTest/RecycleMediaSectionTest.CurrentLocalAndCurrentRemoteRejected/2 (838 ms) [119/451] PeerConnectionIceTest/PeerConnectionIceTest.CannotRemoveIceCandidatesWhenPeerConnectionClosed/0 (747 ms) [120/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationCausingIceRestart/0 (898 ms) [121/451] PeerConnectionBundleTest/PeerConnectionBundleTest.OneCandidateForEachTransportWhenNoBundleButRtcpMux/0 (736 ms) [122/451] PeerConnectionRtpTestUnifiedPlan.ChangeDirectionInAnswerResultsInRemoveTrackEvent (805 ms) [123/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.ExchangeOfferAnswerWhenDtlsOn/0 (787 ms) [124/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/10 (697 ms) [125/451] PeerConnectionIceTest/PeerConnectionIceUfragPwdAnswerTest.TestIncludedInAnswer/2 (927 ms) [126/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CanSetSdesGcmRemoteOfferAndLocalAnswer/0 (801 ms) [127/451] PeerConnectionJsepTest/RecycleMediaSectionTest.CurrentLocalOnlyRejected/2 (967 ms) [128/451] PeerConnectionBundleTest/PeerConnectionBundleTest.CreateRtcpTransportOnlyBeforeAnswerWithRtcpMuxNegotiate/0 (649 ms) [129/451] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingRemoteRejectedAndNotRejectedLocal/2 (677 ms) [130/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/6 (899 ms) [131/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/18 (714 ms) [132/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportDataChannelEndToEnd/0 (650 ms) [133/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.AddTrackBeforeConnecting/0 (355 ms) [134/451] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingRemoteRejectedAndNoLocal/2 (533 ms) [135/451] PeerConnectionIceTest/PeerConnectionIceTest.CannotAddCandidateWhenPeerConnectionClosed/0 (648 ms) [136/451] PeerConnectionFactoryTest.CheckRtpReceiverVideoCapabilities (252 ms) [137/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/18 (853 ms) [138/451] PeerConnectionBundleTest/PeerConnectionBundleTest.RemovingContentAndRejectBundleGroup/0 (490 ms) [139/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInAnswerWhenDtlsEnabled/0 (356 ms) [140/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.DataBufferedUntilRtpDataChannelObserverRegistered/0 (394 ms) [141/451] PeerConnectionJsepTest.CreateAnswerNegotiatesDirection (549 ms) [142/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.GetRemoteAudioSSLCertificateReturnsExchangedCertificate/0 (601 ms) [143/451] PeerConnectionRtpTestUnifiedPlan.AddTransceiverWithDirectionIsReflected (353 ms) [144/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/6 (376 ms) [145/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/14 (411 ms) [146/451] PeerConnectionInterfaceTestPlanB.AddStreams (403 ms) [147/451] PeerConnectionRtpTestUnifiedPlan.AddRemoveAddTrackOffersWorksAudio (512 ms) [148/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationInteropTest.OneAudioLocalToNoMediaRemote/0 (871 ms) [149/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/6 (496 ms) [150/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetLocalAnswerWithNoCryptosWhenSdesOn/0 (414 ms) [151/451] PeerConnectionBundleTest/PeerConnectionBundleMatrixTest.VerifyTransportsBeforeAndAfterSettingRemoteAnswer/0 (615 ms) [152/451] PeerConnectionRtpTest/PeerConnectionRtpTest.AddTrackWithoutStreamAddsReceiver/0 (383 ms) [153/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.StartAndStopLoggingToFileAfterPeerConnectionClosed/0 (279 ms) [154/451] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingLocalRejectedAndNotRejectedRemote/2 (661 ms) [155/451] PeerConnectionJsepTest.CreateOfferGeneratesUniqueMidIfAlreadyTaken (552 ms) [156/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportDataChannelConfigSentToOtherSide/0 (492 ms) [157/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteAnswer/6 (536 ms) [158/451] PeerConnectionRtpTestUnifiedPlan.CanClosePeerConnectionWithoutCrashing (334 ms) [159/451] WebRtcSdpTest.SerializeSessionDescriptionWithSendOnlyContent (236 ms) [160/451] PeerConnectionUsageHistogramTest.NotableUsageNoted (519 ms) [161/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/26 (414 ms) [162/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetLocalOfferWithNoCryptosWhenSdesOn/0 (376 ms) [163/451] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/11 (216 ms) [164/451] PeerConnectionJsepTest/RecycleMediaSectionTest.CurrentRemoteOnlyRejected/2 (640 ms) [165/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/14 (579 ms) [166/451] PeerConnectionRtpTest/PeerConnectionRtpTest.VideoGetParametersHasHeaderExtensions/0 (489 ms) [167/451] PeerConnectionJsepTest.LegacyNoMidAudioVideoOffer (495 ms) [168/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreatePeerConnectionWithDifferentConfigurations/0 (430 ms) [169/451] PeerConnectionIceTest/PeerConnectionIceTest.IceRestartOfferCandidateReplacesExistingCandidate/0 (638 ms) [170/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/10 (401 ms) [171/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateMaxLessThanMinFails/0 (332 ms) [172/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/42 (355 ms) [173/451] PeerConnectionJsepTest.RemoveMsidIfTransceiverStopped (676 ms) [174/451] PeerConnectionRtpTestUnifiedPlan.AddTrackReusesFirstMatchingTransceiver (299 ms) [175/451] PeerConnectionMsidSignalingTest.PlanBOfferToUnifiedPlanAnswer (500 ms) [176/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/10 (420 ms) [177/451] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetRemoteAnswer/2 (319 ms) [178/451] RtpSenderReceiverTest.VideoReceiverCanSetParameters (243 ms) [179/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.ReceiveOfferCreateAnswer/0 (530 ms) [180/451] PeerConnectionMediaTest/PeerConnectionMediaTest.MediaTransportOnlyForDataChannels/0 (341 ms) [181/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemotePrAnswer/10 (718 ms) [182/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.DataChannelOnlyOfferWithMaxBundlePolicy/0 (420 ms) [183/451] PeerConnectionSimulcastTests.SimulcastRejectedRemovesExtraLayers (516 ms) [184/451] PeerConnectionMediaTest/PeerConnectionMediaTest.CreateAnswerWithNoStreamsAndDefaultOptionsDoesNotReject/0 (385 ms) [185/451] PeerConnectionSimulcastMetricsTests.SimulcastDisabledIsLogged (574 ms) [186/451] PeerConnectionInterfaceTestPlanB.AddRemoveTrackFromExistingRemoteMediaStream (434 ms) [187/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CantChangePoolSizeAfterSetLocalDescription/0 (498 ms) [188/451] PeerConnectionInterfaceTestPlanB.RemoveAlreadyGoneRemoteStream (623 ms) [189/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.SessionErrorIfFingerprintInvalid/0 (646 ms) [190/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferReceiveAnswer/0 (588 ms) [191/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.ReceiveFireFoxOffer/0 (593 ms) [192/451] PeerConnectionBundleTest/PeerConnectionBundleTest.FailToSetRemoteOfferWithNoBundleWhenBundlePolicyMaxBundle/0 (523 ms) [193/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CloseAndGetStats/0 (701 ms) [194/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.TestRejectRtpDataChannelInAnswer/0 (331 ms) [195/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/14 (607 ms) [196/451] PeerConnectionSignalingTest/PeerConnectionSignalingTest.CreateOffersAndShutdown/0 (386 ms) [197/451] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetLocalAnswer/6 (403 ms) [198/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.RejectMediaContent/0 (476 ms) [199/451] PeerConnectionJsepTest.SetRemoteOfferReusesTransceiverFromAddTrack (466 ms) [200/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/6 (345 ms) [201/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.GetStatsForInvalidTrack/0 (643 ms) [202/451] PeerConnectionRtpTestPlanB.StatesCorrelateWithSetRemoteDescriptionCall (409 ms) [203/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateSubsequentInactiveOffer/0 (683 ms) [204/451] PeerConnectionBundleTest/PeerConnectionBundleTest.MaxBundleNotSupportedInAnswer/0 (743 ms) [205/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateSctpDataChannel/0 (434 ms) [206/451] SimulcastSdpSerializerTest.Deserialize_SimpleCaseWithAlternatives (384 ms) [207/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/42 (345 ms) [208/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.OffersAndAnswersHaveTrickleIceOption/0 (414 ms) [209/451] HasStream/StatsCollectorTrackTest.VerifyVideoReceiveSsrcStatsNew/1 (237 ms) [210/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/18 (625 ms) [211/451] PeerConnectionSignalingTest/PeerConnectionSignalingTest.SetLocalOfferTwiceWorks/0 (383 ms) [212/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalAnswer/10 (543 ms) [213/451] PeerConnectionIceTest/PeerConnectionIceTest.TrickledSingleCandidateAddedToRemoteDescription/0 (549 ms) [214/451] PeerConnectionMediaTest/PeerConnectionMediaTest.NewStreamInRemoteOfferAddsRecvStreams/0 (552 ms) [215/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateNegativeMinFails/0 (385 ms) [216/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.ExtmapAllowMixedIsConfigurable/0 (536 ms) [217/451] HasStream/StatsCollectorTrackTest.TwoLocalTracksWithSameSsrc/1 (292 ms) [218/451] SignalingProxyTest.ConstMethod1 (274 ms) [219/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationChangesIceCheckInterval/0 (327 ms) [220/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/6 (577 ms) [221/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/10 (386 ms) [222/451] SignalingProxyTest.VoidMethod0 (321 ms) [223/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetRemoteOffer/2 (519 ms) [224/451] PeerConnectionMediaTest/PeerConnectionMediaTest.FailToApplyDescriptionIfVideoEncoderHasEverFailed/0 (427 ms) [225/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.IceRenominationOptionInOfferAndAnswer/0 (462 ms) [226/451] PeerConnectionIceTest/PeerConnectionIceTest.LocalDescriptionUpdatedWhenContinualGathering/0 (514 ms) [227/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CnameGenerationInOffer/0 (577 ms) [228/451] PeerConnectionJsepTest.AnswerBeforeOfferFails (536 ms) [229/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationReturnsInvalidParameterIfCredentialsMissing/0 (394 ms) [230/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/2 (498 ms) [231/451] PeerConnectionBundleTest/PeerConnectionBundleTest.RejectBundleGroupWithNonExistingMid/0 (705 ms) [232/451] PeerConnectionMediaTest/PeerConnectionMediaTest.ReOfferHasSameMidsAsFirstOffer/0 (476 ms) [233/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.IceCandidates/0 (700 ms) [234/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInAnswerWhenDtlsDisabled/0 (555 ms) [235/451] SdpFormatReceivedTest.SimpleUnifiedPlanIsReportedAsSimple (655 ms) [236/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/30 (657 ms) [237/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/2 (526 ms) [238/451] PeerConnectionSignalingTest/PeerConnectionSignalingTest.CreateAnswerFailsIfStable/0 (568 ms) [239/451] PeerConnectionMediaTest/PeerConnectionMediaTest.CreateAnswerWithNoVoiceActivityDetectionIncludesNoComfortNoiseCodecs/0 (506 ms) [240/451] PeerConnectionRtpTest/PeerConnectionRtpTest.AddTrackWithStreamFiresOnAddTrack/0 (711 ms) [241/451] PeerConnectionJsepTest.StoppedTransceiverHasNoMediaSectionInInitialOffer (381 ms) [242/451] PeerConnectionRtpTestUnifiedPlan.AddTrackErrorIfClosed (427 ms) [243/451] WebRtcSdpTest.UnifiedPlanSerializeSessionDescriptionSpecialMsid (410 ms) [244/451] PeerConnectionJsepTest/RecycleMediaSectionTest.PendingLocalRejectedAndNoRemote/2 (776 ms) [245/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/2 (407 ms) [246/451] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetRemoteAnswer/6 (500 ms) [247/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/22 (521 ms) [248/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateCurrentNegativeFails/0 (493 ms) [249/451] RtpSenderReceiverTest.RemoteAudioTrackLatency (441 ms) [250/451] PeerConnectionInterfaceTestPlanB.AttachmentIdIsSetOnAddStream (463 ms) [251/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.ChangeSsrcOnTrackInLocalSessionDescription/0 (868 ms) [252/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInAnswerWhenEncryptionDisabled/0 (514 ms) [253/451] PeerConnectionIceTest/PeerConnectionIceTest.OfferAnswerWithNoTransportsDoesNotChangeIceConnectionState/0 (566 ms) [254/451] PeerConnectionMediaTest/PeerConnectionMediaTest.MediaTransportNotPropagatedToVoiceEngine/0 (515 ms) [255/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.TestDataChannel/0 (592 ms) [256/451] RTCStatsTraversalTest.NoReachableConnections (384 ms) [257/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.CorrectCryptoInOfferWithSdesAndGcm/0 (430 ms) [258/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateReliableRtpDataChannelShouldFail/0 (697 ms) [259/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferWithRtpMux/0 (523 ms) [260/451] PeerConnectionSignalingTest/PeerConnectionSignalingTest.SessionVersionIncrementedInSubsequentDifferentAnswer/0 (592 ms) [261/451] PeerConnectionIceTest/PeerConnectionIceTest.IceCredentialsCreateAnswer/0 (647 ms) [262/451] RtpSenderReceiverTest.AddAndDestroyAudioRtpReceiver (518 ms) [263/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/18 (645 ms) [264/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/6 (523 ms) [265/451] PeerConnectionRtpTest/PeerConnectionRtpTest.LegacyObserverNotCalledIfPeerConnectionDereferenced/0 (638 ms) [266/451] SctpDataChannelTest.ClosedWhenSendBufferFull (420 ms) [267/451] PeerConnectionMediaTest/PeerConnectionMediaInvalidMediaTest.FailToSetLocalAnswer/2 (536 ms) [268/451] PeerConnectionJsepTest.SetLocalDescriptionFailsMissingMid (562 ms) [269/451] PeerConnectionRtpTestUnifiedPlan.AddTrackReusesTransceiver (581 ms) [270/451] PeerConnectionRtpTestUnifiedPlan.AddTransceiverHasCorrectInitProperties (480 ms) [271/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/2 (615 ms) [272/451] PeerConnectionFactoryTest.CreatePCUsingNoUsernameInUri (448 ms) [273/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.SetLocalOffer/10 (721 ms) [274/451] ProxyTest.Method0 (373 ms) [275/451] PeerConnectionCryptoTest/PeerConnectionCryptoTest.FailToSetRemoteOfferWithNoFingerprintWhenDtlsOn/0 (579 ms) [276/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/6 (582 ms) [277/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.TestSendOnlyDataChannel/0 (539 ms) [278/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/30 (549 ms) [279/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferFailsWithInvalidOfferToReceiveVideo/0 (463 ms) [280/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationReturnsRangeErrorForBadCandidatePoolSize/0 (546 ms) [281/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateAnswer/2 (648 ms) [282/451] WebRtcSdpTest.SerializeAndDeserializeWithConnectionAddress (361 ms) [283/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateSctpDataChannelWithInvalidIdShouldFail/0 (545 ms) [284/451] SimulcastSdpSerializerTest.Serialize_OnlyReceive (384 ms) [285/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/18 (388 ms) [286/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreateOfferWithAudioOnlyOptions/0 (661 ms) [287/451] DisableSimulcastLayersInSender/RtpSenderReceiverTest.DisableSimulcastLayersWithMediaEngine/2 (509 ms) [288/451] PeerConnectionMediaTest/PeerConnectionMediaOfferDirectionTest.VerifyDirection/6 (634 ms) [289/451] PeerConnectionIceTest/PeerConnectionIceTest.SetLocalDescriptionFailsIfNoIceCredentials/0 (460 ms) [290/451] PeerConnectionSimulcastTests.SimulcastLayersRemovedFromTail (434 ms) [291/451] PeerConnectionSimulcastTests.CanCreateTransceiverWithSimulcast (469 ms) [292/451] PeerConnectionMediaTest/PeerConnectionMediaTest.AnswerHasDifferentDirectionsForAudioVideo/0 (620 ms) [293/451] PeerConnectionMediaTest/PeerConnectionMediaTest.FailToSetLocalDescriptionIfCreateMediaChannelFails/0 (460 ms) [294/451] PeerConnectionSignalingTest/PeerConnectionSignalingStateTest.CreateOffer/10 (536 ms) [295/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationChangesCandidateFilter/0 (568 ms) [296/451] NumberOfSendEncodings/PeerConnectionSimulcastMetricsTests.NumberOfSendEncodingsIsLogged/2 (414 ms) [297/451] RtpParametersConversionTest.ToCricketStreamParamsVecMissingSsrcs (343 ms) [298/451] WebRtcSdpTest.DeserializeBrokenSdp (387 ms) [299/451] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.DataSectionRejectedIfCalleeHasSctpDisabled/0 (520 ms) [300/451] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.CannotEnableBothMediaTransportAndRtpDataChannels/0 (398 ms) [301/451] RTCConfigurationTest.ComparisonOperators (365 ms) [302/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/2 (898 ms) [303/451] PeerConnectionJsepTest.DataOnlyInitialOffer (324 ms) [304/451] JsepSessionDescriptionTest.SerializeSessionDescriptionWithIPv6Only (371 ms) [305/451] PeerConnectionInterfaceTestPlanB.AddTrackAfterAddStream (599 ms) [306/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.RenegotiationNeededForNewRtpDataChannel/0 (351 ms) [307/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.CreatePeerConnectionWithPooledCandidates/0 (484 ms) [308/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.PooledSessionsDiscardedAfterClose/0 (387 ms) [309/451] SctpDataChannelTest.NoMsgSentIfNegotiatedAndNotFromOpenMsg (337 ms) [310/451] PeerConnectionJsepTest.AddingTrackWithAddTransceiverSpecifiesRandomTrackId (376 ms) [311/451] Encrypted/WebRtcSdpExtmapTest.DeserializeSessionDescriptionWithMediaLevelExtmap/1 (233 ms) [312/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/14 (450 ms) [313/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/6 (387 ms) [314/451] RtpTransceiverDirectionTest/EnumerateAllDirectionsTest.TestReversedSwapped/2 (326 ms) [315/451] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.CreateOfferWithNoDataChannelsGivesNoDataSection/0 (506 ms) [316/451] StatsCollectorTest.ExtractDataInfo (314 ms) [317/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetBitrateCurrentLessThanImplicitMin/0 (292 ms) [318/451] RtpSenderReceiverTest.LocalAudioSourceOptionsApplied (213 ms) [319/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/22 (490 ms) [320/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/30 (350 ms) [321/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetConfigurationFailsAfterClose/0 (309 ms) [322/451] DtmfSenderTest.TryInsertDtmfWhenItDoesNotWork (458 ms) [323/451] RtpSenderReceiverTest.VideoSenderCanSetParametersBeforeNegotiation (225 ms) [324/451] PeerConnectionMediaTestPlanB.SimulcastOffer (486 ms) [325/451] JsepSessionDescriptionTest.RemoveCandidateAndSetConnectionAddress (236 ms) [326/451] NumberOfSendEncodings/PeerConnectionSimulcastMetricsTests.NumberOfSendEncodingsIsLogged/6 (364 ms) [327/451] PeerConnectionCryptoTest/PeerConnectionCryptoDtlsCertGenTest.TestCertificateGeneration/22 (291 ms) [328/451] WebRtcSdpTest.DeserializeSerializeRtcpFb (486 ms) [329/451] RTCStatsCollectorTest.CollectRTCOutboundRTPStreamStats_Audio (322 ms) [330/451] HasStream/StatsCollectorTrackTest.GetStatsFromLocalAudioTrack/1 (353 ms) [331/451] WebRtcSdpTest.DeserializeSdpWithSctpDataChannelAndUnusualPortInAttribute (384 ms) [332/451] PeerConnectionMediaTest/PeerConnectionMediaOfferDirectionTest.VerifyDirection/2 (311 ms) [333/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.GetConfigurationAfterCreatePeerConnection/0 (614 ms) [334/451] SimulcastDeserializationErrors/SimulcastSdpSerializerTest.SimulcastDeserializationFailed/10 (317 ms) [335/451] RtpSenderReceiverTest.VideoSenderDetectInvalidScaleResolutionDownBy (284 ms) [336/451] WebRtcSdpTest.DeserializingNegativeBandwidthLimitFails (243 ms) [337/451] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithSendSet/2 (323 ms) [338/451] DtmfSenderTest.InsertDtmfWhileProviderIsDeleted (246 ms) [339/451] RtpTransceiverDirectionTest/EnumerateAllDirectionsTest.TestReversedIdentity/2 (356 ms) [340/451] PeerConnectionMediaTest/PeerConnectionMediaOfferDirectionTest.VerifyDirection/10 (387 ms) [341/451] WebRtcSdpTest.DeserializeSessionDescriptionWithoutCandidates (337 ms) [342/451] PeerConnectionInterfaceTest/PeerConnectionInterfaceTest.SetIceRegatherIntervalRangeWithoutContinualGatheringFails/0 (473 ms) [343/451] RTCStatsCollectorTest.GetStatsWithSenderSelector (212 ms) [344/451] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.NoSctpTransportCreatedIfMediaTransportDataChannelsEnabled/0 (391 ms) [345/451] WebRtcSdpTest.ParseMediaTransportMultipleLines (228 ms) [346/451] RTCStatsCollectorTest.CollectRTCIceCandidatePairStats (325 ms) [347/451] PeerConnectionFactoryTest.LocalRendering (289 ms) [348/451] WebRtcSdpTest.DeserializeInvalidCandidiate (364 ms) [349/451] PeerConnectionRtpTestUnifiedPlan.RemoveTrackChangesDirectionFromSendOnlyToInactive (327 ms) [350/451] RtpSenderReceiverTest.AudioSenderEarlyWarmupSsrcThenTrack (373 ms) [351/451] WebRtcSdpTest.SerializeBundleOnlyAttribute (206 ms) [352/451] PlayoutLatencyTest.SetLatency (401 ms) [353/451] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/3 (231 ms) [354/451] RTCStatsCollectorTest.CollectLocalRTCMediaStreamStatsAndRTCMediaStreamTrackStats_Video (324 ms) [355/451] HasStream/StatsCollectorTrackTest.TrackAndSsrcObjectExistAfterUpdateSsrcStats/1 (351 ms) [356/451] RtpTransceiverDirectionTest/EnumerateAllDirectionsTest.TestIdentity/2 (323 ms) [357/451] SctpUtilsTest.WriteParseOpenMessageWithMaxRetransmits (356 ms) [358/451] PeerConnectionSignalingTest/PeerConnectionSignalingTest.FailToSetNullLocalDescription/0 (600 ms) [359/451] RtpParametersConversionTest.ToRtpCapabilities (331 ms) [360/451] PeerConnectionFactoryTest.CheckRtpSenderAudioCapabilities (288 ms) [361/451] PeerConnectionDataChannelTest/PeerConnectionDataChannelTest.CreateDataChannelWithDtlsDisabledSucceeds/0 (338 ms) [362/451] RtpSenderReceiverTest.VideoSenderMustCallGetParametersBeforeSetParameters (258 ms) [363/451] JsepSessionDescriptionTest/EnumerateAllSdpTypesTest.TestIdentity/1 (253 ms) [364/451] RTCStatsCollectorTest.CachedStatsReports (333 ms) [365/451] SimulcastSdpSerializerTest.Deserialize_SendReceiveReversed (309 ms) [366/451] StatsCollectorTest.ChainedCertificateReportsCreated (390 ms) [367/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/46 (291 ms) [368/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/38 (203 ms) [369/451] RidDescriptionSdpSerializerTest.Serialize_FormatList (241 ms) [370/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/26 (220 ms) [371/451] MediaStreamTest.RemoveTrack (291 ms) [372/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyDirection/34 (221 ms) [373/451] WebRtcSdpTest.SerializeSessionDescriptionEmpty (207 ms) [374/451] IceServerParsingTest.CopyUsernameAndPasswordFromIceServer (233 ms) [375/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/34 (198 ms) [376/451] RtpSenderReceiverTest.SetVideoMinMaxSendBitrate (352 ms) [377/451] WebRtcSdpTest.DeserializeSdpWithSctpDataChannelsWithSctpColonPort (376 ms) [378/451] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithSendSet/6 (243 ms) [379/451] WebRtcSdpTest.ParseMediaTransportIgnoreNonsenseAttributeLines (250 ms) [380/451] RTCStatsCollectorTest.CollectRTCCertificateStatsMultiple (315 ms) [381/451] WebRtcSdpTest.DeserializeSessionDescriptionWithRecvOnlyContent (213 ms) [382/451] WebRtcSdpTest.DeserializeInvalidPortInCandidateAttribute (240 ms) [383/451] RtpSenderReceiverTest.AudioSenderCantSetUnimplementedRtpEncodingParameters (335 ms) [384/451] RtpSenderReceiverTest.AudioSenderDetectTransactionIdModification (358 ms) [385/451] RtpParametersConversionTest.ToRtpEncodingsWithMultipleStreamParams (295 ms) [386/451] RtpSenderReceiverTest.AudioSenderSsrcSetToZero (293 ms) [387/451] RtpSenderReceiverTest.VideoSenderCannotSetFrameEncryptorAfterStop (246 ms) [388/451] FakeAdmTest.PlayoutTest (217 ms) [389/451] WebRtcSdpTest.DeserializeSessionDescriptionWithoutExtmapAllowMixed (256 ms) [390/451] WebRtcSdpTest.SerializeSimulcast_ComplexSerialization (235 ms) [391/451] WebRtcSdpTest.DeserializeSessionDescriptionWithRejectedVideo (374 ms) [392/451] WebRtcSdpTest.SerializeSessionDescriptionWithExtmapEncrypted (207 ms) [393/451] RidDescriptionSdpSerializerTest.Deserialize_WithMultipleFormats (282 ms) [394/451] TrackMediaInfoMapTest.SingleSenderReceiverPerTrackWithMissingSsrc (240 ms) [395/451] RtpSenderReceiverTest.PropagatesVideoTrackContentHintSetBeforeEnabling (278 ms) [396/451] DisableSimulcastLayersInSender/RtpSenderReceiverTest.DisableSimulcastLayersWithoutMediaEngine/2 (411 ms) [397/451] WebRtcSdpTest.SerializeUnifiedPlanSessionDescription (261 ms) [398/451] HasStream/StatsCollectorTrackTest.ReportsFromRemoteTrack/1 (225 ms) [399/451] SctpDataChannelTest.QueuedCloseFlushes (233 ms) [400/451] RtpSenderReceiverTest.VideoSenderSetParametersOldValueFail (202 ms) [401/451] WebRtcSdpTest.DeserializeSimulcastNegative_DuplicateAttribute (217 ms) [402/451] RtpSenderReceiverTest.RemoteVideoTrackState (304 ms) [403/451] RtpSenderReceiverTest.CanNotInsertDtmf (236 ms) [404/451] HasStream/StatsCollectorTrackTest.VideoBandwidthEstimationInfoIsReported/1 (273 ms) [405/451] RTCStatsCollectorTest.StatsReportedOnZeroSsrc (460 ms) [406/451] SctpDataChannelTest.LateCreatedChannelTransitionToOpen (259 ms) [407/451] WebRtcSdpTest.DeserializeDtlsSetupAttribute (214 ms) [408/451] WebRtcSdpTest.SerializeVideoFmtp (235 ms) [409/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/46 (223 ms) [410/451] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/15 (272 ms) [411/451] RtpSenderReceiverTest.AudioSenderInitParametersMovedAfterNegotiation (228 ms) [412/451] SimulcastDeserializationErrors/SimulcastSdpSerializerTest.SimulcastDeserializationFailed/6 (199 ms) [413/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/38 (214 ms) [414/451] WebRtcSdpTest.DeserializeVideoFmtpWithSpace (388 ms) [415/451] ProxyTest.WorkerMethod2 (217 ms) [416/451] JsepSessionDescriptionTest.AddCandidateWithoutMid (221 ms) [417/451] SctpDataChannelTest.NeverOpened (464 ms) [418/451] RtpTransceiverTest.CannotSetChannelOnStoppedTransceiver (201 ms) [419/451] WebRtcSdpTest.SerializeSessionDescriptionWithDataChannelAndBandwidth (368 ms) [420/451] RtpParametersConversionTest.ToCricketFeedbackParam (212 ms) [421/451] RtpSenderReceiverTest.AudioSenderSsrcChanged (212 ms) [422/451] SctpDataChannelTest.ConnectedAfterTransportBecomesAvailable (230 ms) [423/451] WebRtcSdpTest.DataChannelOnlyHasNoMsidSignaling (410 ms) [424/451] RidDescriptionSdpSerializerTest.Serialize_RestrictionWithoutValue (377 ms) [425/451] RidDescriptionSdpSerializerTest.Deserialize_WithFormatAndRestrictions (209 ms) [426/451] TrackMediaInfoMapTest.MultipleMultiSsrcSendersPerTrack (216 ms) [427/451] WebRtcSdpTest.DeserializeCandidateWithDifferentTransport (223 ms) [428/451] WebRtcSdpTest.DeserializeSessionDescriptionWithBundle (305 ms) [429/451] SimulcastDeserializationErrors/SimulcastSdpSerializerTest.SimulcastDeserializationFailed/2 (198 ms) [430/451] HasStream/StatsCollectorTrackTest.RemoteSsrcInfoIsAbsent/1 (204 ms) [431/451] RTCStatsTraversalTest.MultiplePathsToSameNode (252 ms) [432/451] RtpParametersConversionTest.ToCricketCodecInvalidKind (374 ms) [433/451] RtpParametersConversionTest.ToCricketRtpHeaderExtensionsErrors (196 ms) [434/451] SctpDataChannelTest.BlockedWhenSendQueuedDataNoCrash (378 ms) [435/451] JsepSessionDescriptionTest.AddCandidateDuplicates (355 ms) [436/451] VideoTrackTest.SourceStateChangeTrackState (231 ms) [437/451] RidDescriptionDeserializationErrors/RidDescriptionSdpSerializerTest.RidDescriptionDeserializationFailed/7 (210 ms) [438/451] RtpSenderReceiverTest.AudioSenderCannotSetFrameEncryptorAfterStop (416 ms) [439/451] WebRtcSdpTest.SerializeOnlySsrcAttributeMsid (227 ms) [440/451] WebRtcSdpTest.ParseConnectionDataSessionLevelOnly (269 ms) [441/451] WebRtcSdpTest.SerializeSessionDescriptionWithBundle (302 ms) [442/451] RtpParametersConversionTest.ToCricketCodecInvalidRtcpFeedback (210 ms) [443/451] PeerConnectionMediaTest/PeerConnectionMediaAnswerDirectionTest.VerifyRejected/26 (197 ms) [444/451] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithRecvSet/2 (205 ms) [445/451] IceServerParsingTest.VerifyDefaults (204 ms) [446/451] WebRtcSdpTest.SerializeSessionDescriptionWithAudioVideoRejected (266 ms) [447/451] RtpTransceiverDirectionTest/EnumerateAllDirectionsAndBool.TestWithRecvSet/6 (299 ms) [448/451] WebRtcSdpTest.SerializeSessionDescriptionWithH264 (201 ms) [449/451] RtpParametersConversionTest.ToRtcpFeedbackErrors (223 ms) [450/451] SctpSidAllocatorTest.SctpIdReusedForRemovedDataChannel (233 ms) [451/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.VerifyIceStates/0 (59391 ms) [452/451] PeerConnectionIntegrationTest/PeerConnectionIntegrationIceStatesTest.VerifyIceStates/4 (59695 ms) FAILED TESTS (1/451): 11786 ms: .\peerconnection_unittests.exe PeerConnectionIntegrationTest/PeerConnectionIntegrationTest.MediaTransportOfferUpgradeOnTheCallee/0 (try #1) 2019-03-21 15:04:39,253 - root: [DEBUG] Contents of task_output_dir: ['c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb18f2d5a80d10', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb18f97cdb7610', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb19004982ad10', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb190715c1b610'] 2019-03-21 15:04:39,253 - root: [DEBUG] Found shard_json_files: ['c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb18f2d5a80d10\\output.json', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb18f97cdb7610\\output.json', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb19004982ad10\\output.json', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb190715c1b610\\output.json'] 2019-03-21 15:04:39,253 - root: [INFO] merge_cmd: C:\b\s\w\ir\cache\vpython\691c07\Scripts\python.exe C:\b\s\w\ir\cache\builder\src\testing\merge_scripts\standard_isolated_script_merge.py --summary-json c:\b\s\w\ir\tmp\t\tmpqa8tdp.json --task-output-dir c:\b\s\w\ir\tmp\t\tmpyl4hti -o c:\b\s\w\ir\tmp\t\tmpfmmwqi.json c:\b\s\w\ir\tmp\t\tmpyl4hti\43bb18f2d5a80d10\output.json c:\b\s\w\ir\tmp\t\tmpyl4hti\43bb18f97cdb7610\output.json c:\b\s\w\ir\tmp\t\tmpyl4hti\43bb19004982ad10\output.json c:\b\s\w\ir\tmp\t\tmpyl4hti\43bb190715c1b610\output.json Running ['C:\\b\\s\\w\\ir\\cache\\vpython\\691c07\\Scripts\\python.exe', 'C:\\b\\s\\w\\ir\\cache\\builder\\src\\testing\\merge_scripts\\standard_isolated_script_merge.py', '--summary-json', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpqa8tdp.json', '--task-output-dir', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti', '-o', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpfmmwqi.json', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb18f2d5a80d10\\output.json', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb18f97cdb7610\\output.json', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb19004982ad10\\output.json', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb190715c1b610\\output.json'] in None (env: None) Command ['C:\\b\\s\\w\\ir\\cache\\vpython\\691c07\\Scripts\\python.exe', 'C:\\b\\s\\w\\ir\\cache\\builder\\src\\testing\\merge_scripts\\standard_isolated_script_merge.py', '--summary-json', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpqa8tdp.json', '--task-output-dir', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti', '-o', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpfmmwqi.json', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb18f2d5a80d10\\output.json', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb18f97cdb7610\\output.json', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb19004982ad10\\output.json', 'c:\\b\\s\\w\\ir\\tmp\\t\\tmpyl4hti\\43bb190715c1b610\\output.json'] returned exit code 0