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

python -u /b/s/w/ir/kitchen-checkout/build/scripts/tools/runit.py --show-path --with-third-party-lib -- python /b/s/w/ir/kitchen-checkout/build/scripts/slave/recipe_modules/swarming/resources/collect_task.py --verbose -o /b/s/w/ir/tmp/t/tmpLoxgp9.json --task-output-dir /b/s/w/ir/kitchen-workdir/recipe_cleanup/0w23_17 --merge-script /b/s/w/ir/kitchen-checkout/build/scripts/slave/recipe_modules/swarming/resources/noop_merge.py --merge-script-stdout-file /b/s/w/ir/tmp/t/tmpSFpDMQmerge_script_log --merge-additional-args '[]' --summary-json-file /b/s/w/ir/tmp/t/tmpZ8uMv8.json -- swarming collect -server https://chromium-swarm.appspot.com -worker 50 -task-summary-python -task-output-stdout all -verbose -requests-json /b/s/w/ir/tmp/t/tmp8Gu46F.json in dir /b/s/w/ir/kitchen-workdir: allow_subannotations: False base_name: 'apprtcmobile_tests (iPhone 6s iOS 9.3)' cmd: ['python', '-u', '/b/s/w/ir/kitchen-checkout/build/scripts/tools/runit.py', '--show-path', '--with-third-party-lib', '--', 'python', '/b/s/w/ir/kitchen-checkout/build/scripts/slave/recipe_modules/swarming/resources/collect_task.py', '--verbose', '-o', '/b/s/w/ir/tmp/t/tmpLoxgp9.json', '--task-output-dir', '/b/s/w/ir/kitchen-workdir/recipe_cleanup/0w23_17', '--merge-script', '/b/s/w/ir/kitchen-checkout/build/scripts/slave/recipe_modules/swarming/resources/noop_merge.py', '--merge-script-stdout-file', '/b/s/w/ir/tmp/t/tmpSFpDMQmerge_script_log', '--merge-additional-args', '[]', '--summary-json-file', '/b/s/w/ir/tmp/t/tmpZ8uMv8.json', '--', 'swarming', 'collect', '-server', 'https://chromium-swarm.appspot.com', '-worker', '50', '-task-summary-python', '-task-output-stdout', 'all', '-verbose', '-requests-json', '/b/s/w/ir/tmp/t/tmp8Gu46F.json'] env: {} env_prefixes: {'PATH': '/b/s/w/ir/cache/swarming_client'} env_suffixes: {} infra_step: False name: 'apprtcmobile_tests (iPhone 6s iOS 9.3)' nest_level: 0 ok_ret: frozenset([0]) step_test_data: <lambda>(...) trigger_specs: () full environment: Apple_PubSub_Socket_Render: /private/tmp/com.apple.launchd.5SglnwHDmm/Render BOTO_CONFIG: /b/s/w/ir/tmp/gsutil_task/.boto BUILDBUCKET_EXPERIMENTAL: FALSE CIPD_CACHE_DIR: /b/s/cipd_cache/cache CIPD_PROTOCOL: v2 DEVSHELL_CLIENT_PORT: 52257 DOCKER_CONFIG: /b/s/w/ir/tmp/docker_cfg_task DOCKER_TMPDIR: /b/s/w/ir/tmp/docker_tmp_task GIT_CONFIG_NOSYSTEM: 1 GIT_TERMINAL_PROMPT: 0 HOME: /Users/chrome-bot INFRA_GIT_WRAPPER_HOME: /b/s/w/ir/tmp/git_home_task LOGDOG_COORDINATOR_HOST: logs.chromium.org LOGDOG_STREAM_PREFIX: buildbucket/cr-buildbucket.appspot.com/8922149347395586784 LOGDOG_STREAM_PROJECT: webrtc LOGDOG_STREAM_SERVER_PATH: unix:/b/s/w/ir/tmp/ld.sock LOGNAME: chrome-bot LUCI_CONTEXT: /b/s/w/itsP5nFR/luci_context.340516215 MAC_CHROMIUM_TMPDIR: /b/s/w/ir/tmp/t NO_GCE_CHECK: False PATH: /b/s/w/ir/cache/swarming_client:/b/s/w/ir/cipd_bin_packages:/b/s/w/ir/cipd_bin_packages/bin:/b/s/cipd_cache/bin:/opt/local/bin:/opt/local/sbin:/usr/local/sbin:/usr/local/git/bin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin PWD: /b/s/w/ir PYTHONIOENCODING: UTF-8 PYTHONUNBUFFERED: 1 SHELL: /bin/bash SHLVL: 1 SSH_AUTH_SOCK: /private/tmp/com.apple.launchd.WvfuWvdKgg/Listeners SWARMING_BOT_ID: build184-m9 SWARMING_HEADLESS: 1 SWARMING_SERVER: https://chromium-swarm.appspot.com SWARMING_TASK_ID: 42e2871f13706511 TEMP: /b/s/w/ir/tmp/t TEMPDIR: /b/s/w/ir/tmp/t TMP: /b/s/w/ir/tmp/t TMPDIR: /b/s/w/ir/tmp/t USER: chrome-bot VERSIONER_PYTHON_PREFER_32_BIT: no VERSIONER_PYTHON_VERSION: 2.7 VPYTHON_VIRTUALENV_ROOT: /b/s/w/ir/cache/vpython XPC_FLAGS: 0x0 XPC_SERVICE_NAME: 0 _: /b/s/w/ir/cipd_bin_packages/vpython __CF_USER_TEXT_ENCODING: 0x1F4:0x0:0x0 Set PYTHONPATH: /b/s/w/ir/kitchen-checkout/build/scripts:/b/s/w/ir/kitchen-checkout/build/site_config:/b/s/w/ir/kitchen-checkout/build/third_party:/b/s/w/ir/kitchen-checkout/build/third_party/buildbot_8_4p1:/b/s/w/ir/kitchen-checkout/build/third_party/buildbot_slave_8_4:/b/s/w/ir/kitchen-checkout/build/third_party/coverage-3.7.1:/b/s/w/ir/kitchen-checkout/build/third_party/decorator_3_3_1:/b/s/w/ir/kitchen-checkout/build/third_party/google_api_python_client:/b/s/w/ir/kitchen-checkout/build/third_party/httplib2/python2:/b/s/w/ir/kitchen-checkout/build/third_party/infra_libs:/b/s/w/ir/kitchen-checkout/build/third_party/jinja2:/b/s/w/ir/kitchen-checkout/build/third_party/markupsafe:/b/s/w/ir/kitchen-checkout/build/third_party/oauth2client:/b/s/w/ir/kitchen-checkout/build/third_party/pyasn1:/b/s/w/ir/kitchen-checkout/build/third_party/pyasn1-modules:/b/s/w/ir/kitchen-checkout/build/third_party/python-rsa:/b/s/w/ir/kitchen-checkout/build/third_party/requests_2_10_0:/b/s/w/ir/kitchen-checkout/build/third_party/setuptools-0.6c11:/b/s/w/ir/kitchen-checkout/build/third_party/sqlalchemy_0_7_1:/b/s/w/ir/kitchen-checkout/build/third_party/sqlalchemy_migrate_0_7_1:/b/s/w/ir/kitchen-checkout/build/third_party/tempita_0_5:/b/s/w/ir/kitchen-checkout/build/third_party/twisted_10_2:/b/s/w/ir/kitchen-checkout/build/third_party/uritemplate:/b/s/w/ir/kitchen-checkout/build/third_party/site-packages 2019-02-07 12:48:46,164 - root: [DEBUG] Using task_output_dir: '/b/s/w/ir/kitchen-workdir/recipe_cleanup/0w23_17' 2019-02-07 12:48:46,164 - root: [WARNING] task_output_dir '/b/s/w/ir/kitchen-workdir/recipe_cleanup/0w23_17' already exists! 2019-02-07 12:48:46,164 - root: [WARNING] task_output_dir existing content: [] 2019-02-07 12:48:46,164 - root: [INFO] collect_cmd: swarming collect -server https://chromium-swarm.appspot.com -worker 50 -task-summary-python -task-output-stdout all -verbose -requests-json /b/s/w/ir/tmp/t/tmp8Gu46F.json -output-dir /b/s/w/ir/kitchen-workdir/recipe_cleanup/0w23_17 -task-summary-json /b/s/w/ir/tmp/t/tmpZ8uMv8.json [D2019-02-07T12:48:47.731454-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:48:49.220245-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:48:50.835589-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:48:52.447771-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:48:54.020605-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:48:55.802956-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:48:57.498004-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:48:59.329415-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:01.198949-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:03.060514-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:05.093702-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:07.147253-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:09.207389-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:11.335725-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:13.517040-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:15.673329-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:17.890634-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:20.244118-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:22.725692-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:25.279468-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:27.957467-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:30.740036-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:33.665814-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:36.527811-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:39.583131-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:42.700562-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:45.839916-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:49.142593-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:52.631981-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:56.251889-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:49:59.855396-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:50:03.627886-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:50:07.603370-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:50:11.676268-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:50:15.881093-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:50:20.165711-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:50:24.778955-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:50:29.340956-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:50:34.186819-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:50:39.120234-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:50:44.385350-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:50:49.677837-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:50:55.152575-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:51:00.833675-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:51:06.774555-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:51:12.940348-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:51:19.152977-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:51:25.727653-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:51:32.305897-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:51:39.126279-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:51:46.312662-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:51:53.763435-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:52:01.436833-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:52:09.355852-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:52:17.423032-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:52:25.728484-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:52:34.399636-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:52:43.387046-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:52:52.722533-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:53:02.294974-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:53:12.232682-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:53:22.408853-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:53:33.012725-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:53:43.870355-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:53:55.127511-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:54:06.767491-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:54:18.828360-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:54:31.221513-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:54:44.070758-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:54:57.377683-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:55:11.120274-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:55:25.235631-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:55:39.975336-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:55:55.326603-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:56:10.546997-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:56:25.846926-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:56:41.107783-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:56:56.444837-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:57:11.870109-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:57:27.314732-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:57:42.714512-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:57:58.151026-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:58:13.528330-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:58:28.979090-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:58:44.196448-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:58:59.547292-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:59:14.768044-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:59:30.058158-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T12:59:45.409439-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:00:00.780672-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:00:16.144830-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:00:31.511014-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:00:46.916281-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:01:02.319747-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:01:17.795504-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:01:33.202880-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:01:48.557663-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:02:04.023813-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:02:19.474187-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:02:34.954716-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:02:50.202891-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:03:05.538623-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:03:20.892437-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:03:36.244992-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:03:51.698962-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:04:07.366972-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:04:22.761779-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:04:38.139892-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:04:53.605349-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:05:09.021308-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:05:24.318690-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:05:39.662521-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:05:55.032192-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:06:10.333184-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:06:25.672828-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:06:41.057084-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:06:56.396076-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:07:11.751285-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:07:27.149738-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:07:42.508973-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:07:57.872825-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:08:13.194234-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:08:28.454267-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:08:43.828622-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:08:59.194054-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:09:14.473337-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:09:29.776022-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:09:45.039182-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:10:00.455581-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:10:16.238427-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:10:31.877742-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:10:47.261321-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:11:04.263586-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:11:19.573430-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:11:34.922957-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:11:50.283185-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:12:05.665171-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:12:21.071044-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:12:36.415988-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:12:51.825625-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:13:07.108151-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:13:22.404644-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:13:37.778913-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:13:53.184736-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:14:08.672581-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:14:23.936214-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:14:39.328246-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:14:54.613217-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:15:12.126158-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:15:27.840536-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:15:43.186263-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:15:58.532696-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:16:14.026659-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:16:29.373549-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:16:44.742442-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:17:00.134847-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:17:15.562249-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:17:31.125835-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:17:46.525923-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:18:01.888547-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:18:17.227414-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:18:32.604206-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:18:47.807936-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:19:03.163990-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:19:18.524003-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:19:33.815098-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:19:49.076496-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:20:04.403053-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:20:19.807073-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:20:35.322593-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:20:50.610143-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:21:05.939795-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:21:21.306992-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:21:36.666830-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:21:52.003376-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:22:07.465038-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:22:22.824202-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:22:38.161865-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:22:53.647824-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:23:08.992070-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:23:24.276876-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:23:39.631022-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:23:54.964245-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:24:10.309592-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:24:25.624848-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:24:40.973947-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:24:56.433845-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:25:11.803579-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:25:27.312117-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:25:42.792896-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:25:58.228306-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:26:13.970620-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:26:29.329444-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 [D2019-02-07T13:26:44.665846-08:00 46522 0 collect.go:343] Waiting task_id: 42e2898861104610 42e2898861104610: exit 0 [I 2019-02-07 13:24:44] About to install Xcode 10l232m in Xcode.app for ios Device wiped. Using Xcode version 10.0 build 10L232m at /b/s/w/ir/Xcode.app/Contents/Developer User defaults from command line: IDETestRunSpecificationPath = /var/folders/2j/22s2gz0s7hn48k32d47clxf80000gm/T/ED1DD5E9-7B30-4009-9CA8-B2CB0B26D2E4 Testing started on 'iPhone 6s' 2019-02-07 13:24:54.894 xcodebuild[603:6390] IDETestOperationsObserverDebug: Writing diagnostic log for test session to: /Users/chrome-bot/Library/Developer/Xcode/DerivedData/temporary-bontgrowxwyfgnajrbpubqfcilne/Logs/Test/Test-Transient Testing-2019.02.07_13-24-54--0800.xcresult/1_Test/Diagnostics/apprtcmobile_tests_module-046FD62B-E677-4961-91D4-4DEAFE153152/apprtcmobile_tests_module-36BF3F6E-7D01-4E2E-B1E7-2CA4347CD5C8/Session-apprtcmobile_tests_module-2019-02-07_132454-HyNwFm.log 2019-02-07 13:24:54.895 xcodebuild[603:6361] [MT] IDETestOperationsObserverDebug: (3835BAC6-3736-4738-9867-57463E5C9499) Beginning test session apprtcmobile_tests_module-3835BAC6-3736-4738-9867-57463E5C9499 at 2019-02-07 13:24:54.895 with Xcode 10L232m on target <DVTiPhoneSimulator: 0x7f9fa02b3460> { SimDevice: iPhone 6s (575DCC5F-F87D-4E2F-8DCB-48613DFF1334, iOS 9.3, Shutdown) } (9.3 (13E233)) dyld: warning: could not load inserted library '/b/s/w/ir/Xcode.app/Contents/Developer/Platforms/iPhoneSimulator.platform/Developer/Library/PrivateFrameworks/IDEBundleInjection.framework/IDEBundleInjection' because image not found objc[705]: Class RTCDispatcher is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/apprtcmobile_tests. One of the two will be used. Which one is undefined. objc[705]: Class RTCCameraPreviewView is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/apprtcmobile_tests. One of the two will be used. Which one is undefined. objc[705]: Class RTCEncodedImage is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/apprtcmobile_tests. One of the two will be used. Which one is undefined. objc[705]: Class RTCRtpFragmentationHeader is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/apprtcmobile_tests. One of the two will be used. Which one is undefined. objc[705]: Class RTCVideoCapturer is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/apprtcmobile_tests. One of the two will be used. Which one is undefined. objc[705]: Class RTCVideoCodecInfo is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/apprtcmobile_tests. One of the two will be used. Which one is undefined. objc[705]: Class RTCVideoEncoderQpThresholds is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/apprtcmobile_tests. One of the two will be used. Which one is undefined. objc[705]: Class RTCVideoEncoderSettings is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/apprtcmobile_tests. One of the two will be used. Which one is undefined. objc[705]: Class RTCVideoFrame is implemented in both /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/Frameworks/WebRTC.framework/WebRTC and /Users/chrome-bot/Library/Developer/CoreSimulator/Devices/575DCC5F-F87D-4E2F-8DCB-48613DFF1334/data/Containers/Bundle/Application/3C9C7893-C498-4B33-9149-E8C0B8994239/apprtcmobile_tests.app/apprtcmobile_tests. One of the two will be used. Which one is undefined. Test Suite 'All tests' started at 2019-02-07 21:25:16.022 Test Suite 'apprtcmobile_tests.app' started at 2019-02-07 21:25:16.023 Test Suite 'ARDAppClientTest' started at 2019-02-07 21:25:16.024 Test Case '-[ARDAppClientTest testSession]' started. [000:000] [771] (audio_device_module.mm:21): CreateAudioDeviceModule [000:002] [771] (audio_device_module_ios.mm:43): current platform is IOS [000:003] [771] (audio_device_module_ios.mm:44): iPhone Audio APIs will be utilized. [000:022] [771] (audio_processing_impl.cc:424): Capture analyzer activated: 0 Capture post processor activated: 0 Render pre processor activated: 0 [000:025] [771] (webrtc_video_engine.cc:447): WebRtcVideoEngine::WebRtcVideoEngine() [000:028] [771] (webrtc_voice_engine.cc:194): WebRtcVoiceEngine::WebRtcVoiceEngine [000:037] [42499] (webrtc_voice_engine.cc:217): WebRtcVoiceEngine::Init [000:037] [42499] (webrtc_voice_engine.cc:224): Supported send codecs in order of preference: [000:049] [42499] (webrtc_voice_engine.cc:227): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:049] [42499] (webrtc_voice_engine.cc:227): ISAC/16000/1 (103) [000:049] [42499] (webrtc_voice_engine.cc:227): ISAC/32000/1 (104) [000:049] [42499] (webrtc_voice_engine.cc:227): G722/8000/1 (9) [000:049] [42499] (webrtc_voice_engine.cc:227): ILBC/8000/1 (102) [000:049] [42499] (webrtc_voice_engine.cc:227): PCMU/8000/1 (0) [000:049] [42499] (webrtc_voice_engine.cc:227): PCMA/8000/1 (8) [000:049] [42499] (webrtc_voice_engine.cc:227): CN/32000/1 (106) [000:049] [42499] (webrtc_voice_engine.cc:227): CN/16000/1 (105) [000:049] [42499] (webrtc_voice_engine.cc:227): CN/8000/1 (13) [000:049] [42499] (webrtc_voice_engine.cc:227): telephone-event/48000/1 (110) [000:049] [42499] (webrtc_voice_engine.cc:227): telephone-event/32000/1 (112) [000:049] [42499] (webrtc_voice_engine.cc:227): telephone-event/16000/1 (113) [000:049] [42499] (webrtc_voice_engine.cc:227): telephone-event/8000/1 (126) [000:049] [42499] (webrtc_voice_engine.cc:230): Supported recv codecs in order of preference: [000:050] [42499] (webrtc_voice_engine.cc:233): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:050] [42499] (webrtc_voice_engine.cc:233): ISAC/16000/1 (103) [000:050] [42499] (webrtc_voice_engine.cc:233): ISAC/32000/1 (104) [000:050] [42499] (webrtc_voice_engine.cc:233): G722/8000/1 (9) [000:050] [42499] (webrtc_voice_engine.cc:233): ILBC/8000/1 (102) [000:050] [42499] (webrtc_voice_engine.cc:233): PCMU/8000/1 (0) [000:050] [42499] (webrtc_voice_engine.cc:233): PCMA/8000/1 (8) [000:050] [42499] (webrtc_voice_engine.cc:233): CN/32000/1 (106) [000:050] [42499] (webrtc_voice_engine.cc:233): CN/16000/1 (105) [000:050] [42499] (webrtc_voice_engine.cc:233): CN/8000/1 (13) [000:050] [42499] (webrtc_voice_engine.cc:233): telephone-event/48000/1 (110) [000:050] [42499] (webrtc_voice_engine.cc:233): telephone-event/32000/1 (112) [000:050] [42499] (webrtc_voice_engine.cc:233): telephone-event/16000/1 (113) [000:050] [42499] (webrtc_voice_engine.cc:233): telephone-event/8000/1 (126) [000:050] [42499] (audio_device_module_ios.mm:68): Init [000:051] [42499] (audio_device_buffer.cc:61): AudioDeviceBuffer::ctor [000:052] [42499] (audio_device_ios.mm:116): AudioDeviceIOS::ctor<NSThread: 0x7faf37449940>{number = 3, name = (null)} [000:052] [42499] (audio_device_module_ios.mm:48): AttachAudioBuffer [000:052] [42499] (audio_device_ios.mm:132): AudioDeviceIOS::AttachAudioBuffer [000:052] [42499] (audio_device_ios.mm:139): AudioDeviceIOS::Init [000:052] [42499] (audio_device_ios.mm:84): LogDeviceInfo [000:052] [42499] (audio_device_ios.mm:86): system name: iPhone OS [000:052] [42499] (audio_device_ios.mm:87): system version: 9.3 [000:052] [42499] (audio_device_ios.mm:88): device type: iPhone [000:053] [42499] (audio_device_ios.mm:89): device name: x86_64 [000:053] [42499] (audio_device_ios.mm:90): process name: apprtcmobile_tests [000:053] [42499] (audio_device_ios.mm:91): process ID: 705 [000:053] [42499] (audio_device_ios.mm:92): OS version: Version 9.3 (Build 13E233) [000:053] [42499] (audio_device_ios.mm:93): processing cores: 8 [000:053] [42499] (audio_device_ios.mm:94): low power mode: 0 [000:053] [42499] (audio_device_ios.mm:96): TARGET_IPHONE_SIMULATOR is defined [000:053] [42499] (audio_device_ios.mm:98): DeviceIsSimulator: 1 [000:053] [42499] (audio_device_ios.mm:648): AudioDeviceIOS::UpdateAudioDevicebuffer [000:054] [42499] (audio_device_buffer.cc:184): SetPlayoutSampleRate(48000) [000:054] [42499] (audio_device_buffer.cc:204): SetPlayoutChannels(1) [000:054] [42499] (audio_device_buffer.cc:178): SetRecordingSampleRate(48000) [000:054] [42499] (audio_device_buffer.cc:198): SetRecordingChannels(1) [000:055] [42499] (audio_device_module_ios.mm:405): SetPlayoutDevice(0) [000:055] [42499] (audio_device_ios.mm:992): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetPlayoutDevice(uint16_t): Not implemented [000:055] [42499] (audio_device_module_ios.mm:107): InitSpeaker [000:055] [42499] (audio_device_module_ios.mm:309): StereoPlayoutIsAvailable [000:055] [42499] (audio_device_module_ios.mm:316): output: 0 [000:055] [42499] (audio_device_module_ios.mm:321): SetStereoPlayout(0) [000:055] [42499] (audio_device_ios.mm:1045): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetStereoPlayout(bool): Not implemented [000:055] [42499] (audio_device_module_ios.mm:329): stereo playout is not supported [000:055] [42499] (adm_helpers.cc:57): Failed to set stereo playout mode. [000:055] [42499] (audio_device_module_ios.mm:467): SetRecordingDevice(0) [000:055] [42499] (audio_device_ios.mm:1094): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetRecordingDevice(uint16_t): Not implemented [000:055] [42499] (audio_device_module_ios.mm:113): InitMicrophone [000:055] [42499] (audio_device_module_ios.mm:276): StereoRecordingIsAvailable [000:056] [42499] (audio_device_module_ios.mm:283): output: 0 [000:056] [42499] (audio_device_module_ios.mm:288): SetStereoRecording(0) [000:056] [42499] (adm_helpers.cc:77): Failed to set stereo recording mode. [000:059] [42499] (audio_device_module_ios.mm:580): RegisterAudioCallback [000:059] [42499] (audio_device_buffer.cc:79): RegisterAudioCallback [000:059] [42499] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 1, agc: 1, ns: 1, hf: 1, swap: 0, audio_jitter_buffer_max_packets: 50, 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] [42499] (webrtc_voice_engine.cc:331): Always disable AEC on iOS. Use built-in instead. [000:059] [42499] (webrtc_voice_engine.cc:358): Always disable NS on iOS. Use built-in instead. [000:059] [42499] (webrtc_voice_engine.cc:369): Always disable AGC on iOS. Use built-in instead. [000:059] [42499] (audio_device_module_ios.mm:596): BuiltInAECIsAvailable [000:059] [42499] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [000:059] [42499] (audio_device_module_ios.mm:599): output: 0 [000:059] [42499] (audio_processing_impl.cc:697): Highpass filter activated: 0 [000:060] [42499] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [000:060] [42499] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [000:060] [42499] (apm_helpers.cc:90): Echo control set to 0 with mode 0 [000:060] [42499] (audio_device_module_ios.mm:612): BuiltInAGCIsAvailable [000:060] [42499] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [000:060] [42499] (audio_device_module_ios.mm:615): output: 0 [000:060] [42499] (apm_helpers.cc:79): AGC set to 0 with mode 2 [000:060] [42499] (audio_device_module_ios.mm:628): BuiltInNSIsAvailable [000:060] [42499] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [000:060] [42499] (audio_device_module_ios.mm:631): output: 0 [000:060] [42499] (audio_processing_impl.cc:697): Highpass filter activated: 0 [000:060] [42499] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [000:060] [42499] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [000:060] [42499] (apm_helpers.cc:104): NS set to 0 [000:060] [42499] (webrtc_voice_engine.cc:466): Stereo swapping enabled? 0 [000:061] [42499] (webrtc_voice_engine.cc:471): NetEq capacity is 50 [000:061] [42499] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:061] [42499] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:061] [42499] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:061] [42499] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:061] [42499] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:061] [42499] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:061] [42499] (webrtc_voice_engine.cc:536): Typing detection is enabled? 0 [000:061] [42499] (audio_processing_impl.cc:697): Highpass filter activated: 1 [000:061] [42499] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [000:061] [42499] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 (RTCLogging.mm:33): (ARDAppClient.m:270 -[ARDAppClient connectToRoomWithId:settings:isLoopback:]_block_invoke): Joined room:testRoom on room server. [000:068] [771] (openssl_identity.cc:44): Making key pair [000:252] [771] (openssl_identity.cc:92): Returning key pair [000:253] [771] (openssl_certificate.cc:58): Making certificate for WebRTC [000:261] [771] (openssl_certificate.cc:108): Returning certificate [000:263] [771] (RTCCertificate.mm:63): CERT PEM [000:263] [771] (RTCCertificate.mm:64): -----BEGIN CERTIFICATE----- MIIBnjCCAQegAwIBAgIJAOCasLUd/OxwMA0GCSqGSIb3DQEBCwUAMBExDzANBgNV BAMMBldlYlJUQzAeFw0xOTAyMDYyMTI1MTZaFw0xOTAyMDcyMTI2NTZaMBExDzAN BgNVBAMMBldlYlJUQzCBnzANBgkqhkiG9w0BAQEFAAOBjQAwgYkCgYEAr89vYF+2 PGxwz+4mramo+Y+GBat5vQhPeT+Yr6IuVCzVj6Ac9Ub+CIJFA5aKOy6DJH3tRbRt 5UNYgSUhttZ2W0dsmPuvspjmnxQue1Lxv4EyDjrkVvFFPi1hcMyhdKNjao/vDcCK ocDqlXEjBEIvCPXxDcPoiOniNAQcFWc9t+0CAwEAATANBgkqhkiG9w0BAQsFAAOB gQByZNJtpu2DyG907ogW3V/68erqWRy1i04cyROzvrt18qK41+Er6FFfgHDmMVGe 733oaCOa0yxE4400UxuPuFbbKcUJK+saoLMQZtAddEaIUvtDIvP3z7gpbky+Ngdm aPHD+zRlgBcykZhisVh2B4MsBtBJR0SQ3UinoX5XYYYkFg== -----END CERTIFICATE----- [000:264] [771] (RTCConfiguration.mm:213): Have configured cert - using it. [000:265] [771] (RTCConfiguration.mm:218): Created cert from PEM strings. [000:273] [42499] (rtc_event_log_impl.cc:70): Creating legacy encoder for RTC event log. [000:274] [42499] (peer_connection_factory.cc:450): Using default network controller factory [000:279] [42499] (bitrate_prober.cc:64): Bandwidth probing enabled, set to inactive [000:284] [42499] (paced_sender.cc:399): ProcessThreadAttached 0x3744f990 [000:284] [42499] (cpu_info.cc:49): Available number of cores: 8 [000:289] [42499] (aimd_rate_control.cc:101): Using aimd rate control with back off factor 0.85 [000:289] [42499] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating. [000:317] [42243] (message_queue.cc:518): Message took 51ms to dispatch. Posted from: CreatePeerConnection@../../api/peer_connection_factory_proxy.h:34 [000:317] [42243] (webrtc_session_description_factory.cc:314): Using certificate supplied to the constructor. [000:321] [42243] (peer_connection.cc:1314): Adding audio transceiver in response to a call to AddTrack. (RTCLogging.mm:33): (ARDAppClient.m:396 -[ARDAppClient peerConnectionShouldNegotiate:]): WARNING: Renegotiation needed but unimplemented. [000:325] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x7faf34416760): created DTMF sender: RTCDtmfSender { remainingTones: duration: 0.100000 sec interToneGap: 0.050000 sec } [000:326] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf37306110): created sender: RTCRtpSender { senderId: ARDAMSa0 } [000:328] [42243] (peer_connection.cc:1314): Adding video transceiver in response to a call to AddTrack. (RTCLogging.mm:33): (ARDAppClient.m:396 -[ARDAppClient peerConnectionShouldNegotiate:]): WARNING: Renegotiation needed but unimplemented. [000:328] [42243] (rtp_sender.cc:561): Tried to get DTMF sender from video sender. [000:328] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf372055c0): created sender: RTCRtpSender { senderId: ARDAMSv0 } [000:329] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x7faf37452c40): created DTMF sender: RTCDtmfSender { remainingTones: duration: 0.100000 sec interToneGap: 0.050000 sec } [000:329] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf37209af0): created sender: RTCRtpSender { senderId: ARDAMSa0 } [000:330] [771] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x7faf37454d50): created receiver: RTCRtpReceiver { receiverId: b213d12b-dc73-4a90-ae02-3e5edac76902 } [000:330] [771] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x7faf37209510): created transceiver: RTCRtpTransceiver { sender: RTCRtpSender { senderId: ARDAMSa0 } receiver: RTCRtpReceiver { receiverId: b213d12b-dc73-4a90-ae02-3e5edac76902 } } [000:330] [42243] (rtp_sender.cc:561): Tried to get DTMF sender from video sender. [000:330] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf37309060): created sender: RTCRtpSender { senderId: ARDAMSv0 } [000:330] [771] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x7faf3730bef0): created receiver: RTCRtpReceiver { receiverId: c8decf54-dee3-48ff-ba49-4c0fad656b99 } [000:331] [771] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x7faf3730c130): created transceiver: RTCRtpTransceiver { sender: RTCRtpSender { senderId: ARDAMSv0 } receiver: RTCRtpReceiver { receiverId: c8decf54-dee3-48ff-ba49-4c0fad656b99 } } [000:335] [42243] (media_session.cc:333): Duplicate id found. Reassigning from 102 to 127 [000:336] [42243] (media_session.cc:333): Duplicate id found. Reassigning from 104 to 125 [000:336] [42243] (media_session.cc:333): Duplicate id found. Reassigning from 106 to 124 [000:336] [42243] (media_session.cc:333): Duplicate id found. Reassigning from 103 to 123 [000:336] [42243] (media_session.cc:333): Duplicate id found. Reassigning from 105 to 122 [000:343] [42499] (rtc_event_log_impl.cc:185): Starting WebRTC event log. (Timestamp, UTC) = (127814541, 1549574716387026). [000:345] [771] (audio_device_module.mm:21): CreateAudioDeviceModule [000:345] [771] (audio_device_module_ios.mm:43): current platform is IOS [000:345] [771] (audio_device_module_ios.mm:44): iPhone Audio APIs will be utilized. [000:345] [771] (audio_processing_impl.cc:424): Capture analyzer activated: 0 Capture post processor activated: 0 Render pre processor activated: 0 [000:346] [771] (webrtc_video_engine.cc:447): WebRtcVideoEngine::WebRtcVideoEngine() [000:346] [771] (webrtc_voice_engine.cc:194): WebRtcVoiceEngine::WebRtcVoiceEngine [000:346] [22787] (webrtc_voice_engine.cc:217): WebRtcVoiceEngine::Init [000:346] [22787] (webrtc_voice_engine.cc:224): Supported send codecs in order of preference: [000:346] [22787] (webrtc_voice_engine.cc:227): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:346] [22787] (webrtc_voice_engine.cc:227): ISAC/16000/1 (103) [000:346] [22787] (webrtc_voice_engine.cc:227): ISAC/32000/1 (104) [000:346] [22787] (webrtc_voice_engine.cc:227): G722/8000/1 (9) [000:346] [22787] (webrtc_voice_engine.cc:227): ILBC/8000/1 (102) [000:346] [22787] (webrtc_voice_engine.cc:227): PCMU/8000/1 (0) [000:346] [22787] (webrtc_voice_engine.cc:227): PCMA/8000/1 (8) [000:346] [22787] (webrtc_voice_engine.cc:227): CN/32000/1 (106) [000:346] [22787] (webrtc_voice_engine.cc:227): CN/16000/1 (105) [000:346] [22787] (webrtc_voice_engine.cc:227): CN/8000/1 (13) [000:346] [22787] (webrtc_voice_engine.cc:227): telephone-event/48000/1 (110) [000:346] [22787] (webrtc_voice_engine.cc:227): telephone-event/32000/1 (112) [000:346] [22787] (webrtc_voice_engine.cc:227): telephone-event/16000/1 (113) [000:346] [22787] (webrtc_voice_engine.cc:227): telephone-event/8000/1 (126) [000:346] [22787] (webrtc_voice_engine.cc:230): Supported recv codecs in order of preference: [000:347] [22787] (webrtc_voice_engine.cc:233): opus/48000/2 { minptime=10 useinbandfec=1 } (111) [000:347] [22787] (webrtc_voice_engine.cc:233): ISAC/16000/1 (103) [000:347] [22787] (webrtc_voice_engine.cc:233): ISAC/32000/1 (104) [000:347] [22787] (webrtc_voice_engine.cc:233): G722/8000/1 (9) [000:347] [22787] (webrtc_voice_engine.cc:233): ILBC/8000/1 (102) [000:347] [22787] (webrtc_voice_engine.cc:233): PCMU/8000/1 (0) [000:347] [22787] (webrtc_voice_engine.cc:233): PCMA/8000/1 (8) [000:347] [22787] (webrtc_voice_engine.cc:233): CN/32000/1 (106) [000:347] [22787] (webrtc_voice_engine.cc:233): CN/16000/1 (105) [000:347] [22787] (webrtc_voice_engine.cc:233): CN/8000/1 (13) [000:347] [22787] (webrtc_voice_engine.cc:233): telephone-event/48000/1 (110) [000:347] [22787] (webrtc_voice_engine.cc:233): telephone-event/32000/1 (112) [000:347] [22787] (webrtc_voice_engine.cc:233): telephone-event/16000/1 (113) [000:347] [22787] (webrtc_voice_engine.cc:233): telephone-event/8000/1 (126) [000:347] [22787] (audio_device_module_ios.mm:68): Init [000:347] [22787] (audio_device_buffer.cc:61): AudioDeviceBuffer::ctor [000:347] [22787] (audio_device_ios.mm:116): AudioDeviceIOS::ctor<NSThread: 0x7faf3720afa0>{number = 4, name = (null)} [000:347] [22787] (audio_device_module_ios.mm:48): AttachAudioBuffer [000:347] [22787] (audio_device_ios.mm:132): AudioDeviceIOS::AttachAudioBuffer [000:347] [22787] (audio_device_ios.mm:139): AudioDeviceIOS::Init [000:347] [22787] (audio_device_ios.mm:84): LogDeviceInfo [000:347] [22787] (audio_device_ios.mm:86): system name: iPhone OS [000:347] [22787] (audio_device_ios.mm:87): system version: 9.3 [000:347] [22787] (audio_device_ios.mm:88): device type: iPhone [000:347] [22787] (audio_device_ios.mm:89): device name: x86_64 [000:347] [22787] (audio_device_ios.mm:90): process name: apprtcmobile_tests [000:347] [22787] (audio_device_ios.mm:91): process ID: 705 [000:348] [22787] (audio_device_ios.mm:92): OS version: Version 9.3 (Build 13E233) [000:348] [22787] (audio_device_ios.mm:93): processing cores: 8 [000:348] [22787] (audio_device_ios.mm:94): low power mode: 0 [000:348] [22787] (audio_device_ios.mm:96): TARGET_IPHONE_SIMULATOR is defined [000:348] [22787] (audio_device_ios.mm:98): DeviceIsSimulator: 1 [000:348] [22787] (audio_device_ios.mm:648): AudioDeviceIOS::UpdateAudioDevicebuffer [000:348] [22787] (audio_device_buffer.cc:184): SetPlayoutSampleRate(48000) [000:348] [22787] (audio_device_buffer.cc:204): SetPlayoutChannels(1) [000:348] [22787] (audio_device_buffer.cc:178): SetRecordingSampleRate(48000) [000:348] [22787] (audio_device_buffer.cc:198): SetRecordingChannels(1) [000:348] [22787] (audio_device_module_ios.mm:405): SetPlayoutDevice(0) [000:348] [22787] (audio_device_ios.mm:992): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetPlayoutDevice(uint16_t): Not implemented [000:348] [22787] (audio_device_module_ios.mm:107): InitSpeaker [000:348] [22787] (audio_device_module_ios.mm:309): StereoPlayoutIsAvailable [000:348] [22787] (audio_device_module_ios.mm:316): output: 0 [000:348] [22787] (audio_device_module_ios.mm:321): SetStereoPlayout(0) [000:348] [22787] (audio_device_ios.mm:1045): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetStereoPlayout(bool): Not implemented [000:348] [22787] (audio_device_module_ios.mm:329): stereo playout is not supported [000:348] [22787] (adm_helpers.cc:57): Failed to set stereo playout mode. [000:348] [22787] (audio_device_module_ios.mm:467): SetRecordingDevice(0) [000:348] [22787] (audio_device_ios.mm:1094): virtual int32_t webrtc::ios_adm::AudioDeviceIOS::SetRecordingDevice(uint16_t): Not implemented [000:348] [22787] (audio_device_module_ios.mm:113): InitMicrophone [000:348] [22787] (audio_device_module_ios.mm:276): StereoRecordingIsAvailable [000:348] [22787] (audio_device_module_ios.mm:283): output: 0 [000:348] [22787] (audio_device_module_ios.mm:288): SetStereoRecording(0) [000:348] [22787] (adm_helpers.cc:77): Failed to set stereo recording mode. [000:348] [22787] (audio_device_module_ios.mm:580): RegisterAudioCallback [000:349] [22787] (audio_device_buffer.cc:79): RegisterAudioCallback [000:349] [22787] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 1, agc: 1, ns: 1, hf: 1, swap: 0, audio_jitter_buffer_max_packets: 50, 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, } [000:349] [22787] (webrtc_voice_engine.cc:331): Always disable AEC on iOS. Use built-in instead. [000:349] [22787] (webrtc_voice_engine.cc:358): Always disable NS on iOS. Use built-in instead. [000:349] [22787] (webrtc_voice_engine.cc:369): Always disable AGC on iOS. Use built-in instead. [000:349] [22787] (audio_device_module_ios.mm:596): BuiltInAECIsAvailable [000:349] [22787] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [000:349] [22787] (audio_device_module_ios.mm:599): output: 0 [000:349] [22787] (audio_processing_impl.cc:697): Highpass filter activated: 0 [000:350] [22787] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [000:350] [22787] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [000:350] [22787] (apm_helpers.cc:90): Echo control set to 0 with mode 0 [000:350] [22787] (audio_device_module_ios.mm:612): BuiltInAGCIsAvailable [000:350] [22787] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [000:350] [22787] (audio_device_module_ios.mm:615): output: 0 [000:350] [22787] (apm_helpers.cc:79): AGC set to 0 with mode 2 [000:350] [22787] (audio_device_module_ios.mm:628): BuiltInNSIsAvailable [000:350] [22787] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [000:350] [22787] (audio_device_module_ios.mm:631): output: 0 [000:350] [22787] (audio_processing_impl.cc:697): Highpass filter activated: 0 [000:350] [22787] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [000:350] [22787] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [000:350] [22787] (apm_helpers.cc:104): NS set to 0 [000:350] [22787] (webrtc_voice_engine.cc:466): Stereo swapping enabled? 0 [000:350] [22787] (webrtc_voice_engine.cc:471): NetEq capacity is 50 [000:350] [22787] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:350] [22787] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:350] [22787] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:350] [22787] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:350] [22787] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:350] [22787] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:350] [22787] (webrtc_voice_engine.cc:536): Typing detection is enabled? 0 [000:350] [22787] (audio_processing_impl.cc:697): Highpass filter activated: 1 [000:351] [22787] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [000:351] [22787] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 (RTCLogging.mm:33): (ARDAppClient.m:270 -[ARDAppClient connectToRoomWithId:settings:isLoopback:]_block_invoke): Joined room:testRoom on room server. [000:351] [771] (openssl_identity.cc:44): Making key pair [000:628] [771] (openssl_identity.cc:92): Returning key pair [000:628] [771] (openssl_certificate.cc:58): Making certificate for WebRTC [000:631] [771] (openssl_certificate.cc:108): Returning certificate [000:631] [771] (RTCCertificate.mm:63): CERT PEM [000:631] [771] (RTCCertificate.mm:64): -----BEGIN CERTIFICATE----- MIIBnjCCAQegAwIBAgIJAP7KGdpEp6bIMA0GCSqGSIb3DQEBCwUAMBExDzANBgNV BAMMBldlYlJUQzAeFw0xOTAyMDYyMTI1MTZaFw0xOTAyMDcyMTI2NTZaMBExDzAN BgNVBAMMBldlYlJUQzCBnzANBgkqhkiG9w0BAQEFAAOBjQAwgYkCgYEAoWDf+L2x b7yuP1J3CIHm2ZBFTdab+BMcDtNd4kfLwXuTHHI1z4CxyCNtOVqXdc3YStKUdKQB eKKZdU3iORu6V3fnku0SD+Vr+cJMMrbbfor8yjXdpJrmagbjf0gvgCmchsjoUdkW ze3ld9H9DcrkVzug7M2xHU6dF6fEfdoa1GsCAwEAATANBgkqhkiG9w0BAQsFAAOB gQBPU0OnOl5VdAAqK5WxFPy5DpCG237mAslZ9oq3NT750jlPyrnlUB9qIqx6i4+r 6ZYbMZYxAYPwrmbz375sBBDo2sFI6EcipzqQinS4yFSrOEbyI4MSdWt7YivHscxX I+y/TDE35uH4ebmGOYI8j7xFXWaBUZnl7LJwFDERy2c68w== -----END CERTIFICATE----- [000:631] [771] (RTCConfiguration.mm:213): Have configured cert - using it. [000:633] [771] (RTCConfiguration.mm:218): Created cert from PEM strings. [000:634] [22787] (rtc_event_log_impl.cc:70): Creating legacy encoder for RTC event log. [000:634] [22787] (peer_connection_factory.cc:450): Using default network controller factory [000:634] [22787] (bitrate_prober.cc:64): Bandwidth probing enabled, set to inactive [000:634] [22787] (paced_sender.cc:399): ProcessThreadAttached 0x3451ff50 [000:634] [22787] (aimd_rate_control.cc:101): Using aimd rate control with back off factor 0.85 [000:634] [22787] (remote_bitrate_estimator_single_stream.cc:71): RemoteBitrateEstimatorSingleStream: Instantiating. [000:635] [41987] (webrtc_session_description_factory.cc:314): Using certificate supplied to the constructor. [000:636] [41987] (peer_connection.cc:1314): Adding audio transceiver in response to a call to AddTrack. (RTCLogging.mm:33): (ARDAppClient.m:396 -[ARDAppClient peerConnectionShouldNegotiate:]): WARNING: Renegotiation needed but unimplemented. [000:637] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x7faf372269a0): created DTMF sender: RTCDtmfSender { remainingTones: duration: 0.100000 sec interToneGap: 0.050000 sec } [000:637] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf3470e3f0): created sender: RTCRtpSender { senderId: ARDAMSa0 } [000:637] [41987] (peer_connection.cc:1314): Adding video transceiver in response to a call to AddTrack. (RTCLogging.mm:33): (ARDAppClient.m:396 -[ARDAppClient peerConnectionShouldNegotiate:]): WARNING: Renegotiation needed but unimplemented. [000:637] [41987] (rtp_sender.cc:561): Tried to get DTMF sender from video sender. [000:638] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf3441b120): created sender: RTCRtpSender { senderId: ARDAMSv0 } [000:638] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x7faf3730cb20): created DTMF sender: RTCDtmfSender { remainingTones: duration: 0.100000 sec interToneGap: 0.050000 sec } [000:638] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf3441b0e0): created sender: RTCRtpSender { senderId: ARDAMSa0 } [000:638] [771] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x7faf34701160): created receiver: RTCRtpReceiver { receiverId: 9810a9db-a4fb-4db4-9675-5a1d69df7fef } [000:639] [771] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x7faf3441c550): created transceiver: RTCRtpTransceiver { sender: RTCRtpSender { senderId: ARDAMSa0 } receiver: RTCRtpReceiver { receiverId: 9810a9db-a4fb-4db4-9675-5a1d69df7fef } } [000:639] [41987] (rtp_sender.cc:561): Tried to get DTMF sender from video sender. [000:639] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf3730d540): created sender: RTCRtpSender { senderId: ARDAMSv0 } [000:639] [771] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x7faf345216c0): created receiver: RTCRtpReceiver { receiverId: b2e291d8-140c-4aeb-908d-7e863de2bbcc } [000:639] [771] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x7faf3730d960): created transceiver: RTCRtpTransceiver { sender: RTCRtpSender { senderId: ARDAMSv0 } receiver: RTCRtpReceiver { receiverId: b2e291d8-140c-4aeb-908d-7e863de2bbcc } } [000:640] [22787] (rtc_event_log_impl.cc:185): Starting WebRTC event log. (Timestamp, UTC) = (128111578, 1549574716684063). [000:643] [771] (webrtc_sdp.cc:3151): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:645] [771] (webrtc_sdp.cc:3151): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:653] [22531] (p2p_transport_channel.cc:489): Set backup connection ping interval to 25000 milliseconds. [000:653] [22531] (p2p_transport_channel.cc:498): Set ICE receiving timeout to 2500 milliseconds [000:653] [22531] (p2p_transport_channel.cc:505): Set ping most likely connection to 0 [000:653] [22531] (p2p_transport_channel.cc:512): Set stable_writable_connection_ping_interval to 2500 [000:653] [22531] (p2p_transport_channel.cc:525): Set presume writable when fully relayed to 0 [000:653] [22531] (p2p_transport_channel.cc:534): Set regather_on_failed_networks_interval to 300000 [000:653] [22531] (p2p_transport_channel.cc:553): Set receiving_switching_delay to 1000 [000:654] [22531] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0 [000:654] [22531] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 3745d430 [000:655] [22531] (p2p_transport_channel.cc:489): Set backup connection ping interval to 25000 milliseconds. [000:655] [22531] (p2p_transport_channel.cc:498): Set ICE receiving timeout to 2500 milliseconds [000:655] [22531] (p2p_transport_channel.cc:505): Set ping most likely connection to 0 [000:655] [22531] (p2p_transport_channel.cc:512): Set stable_writable_connection_ping_interval to 2500 [000:655] [22531] (p2p_transport_channel.cc:525): Set presume writable when fully relayed to 0 [000:655] [22531] (p2p_transport_channel.cc:534): Set regather_on_failed_networks_interval to 300000 [000:655] [22531] (p2p_transport_channel.cc:553): Set receiving_switching_delay to 1000 [000:655] [22531] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 1 transport 0 [000:655] [22531] (dtls_srtp_transport.cc:66): Setting RTP Transport on 1 transport 3747fe50 [000:656] [22531] (p2p_transport_channel.cc:426): Set ICE ufrag: fAzx pwd: QilkW90Q9UtzhCKlhakNY9r/ on transport 0 [000:656] [22531] (p2p_transport_channel.cc:426): Set ICE ufrag: fAzx pwd: QilkW90Q9UtzhCKlhakNY9r/ on transport 1 [000:657] [42499] (webrtc_voice_engine.cc:1476): Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [000:657] [42499] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [000:657] [42499] (webrtc_voice_engine.cc:331): Always disable AEC on iOS. Use built-in instead. [000:657] [42499] (webrtc_voice_engine.cc:358): Always disable NS on iOS. Use built-in instead. [000:657] [42499] (webrtc_voice_engine.cc:369): Always disable AGC on iOS. Use built-in instead. [000:657] [42499] (audio_device_module_ios.mm:596): BuiltInAECIsAvailable [000:657] [42499] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [000:657] [42499] (audio_device_module_ios.mm:599): output: 0 [000:657] [42499] (audio_processing_impl.cc:697): Highpass filter activated: 1 [000:657] [42499] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [000:657] [42499] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [000:657] [42499] (apm_helpers.cc:90): Echo control set to 0 with mode 0 [000:657] [42499] (audio_device_module_ios.mm:612): BuiltInAGCIsAvailable [000:657] [42499] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [000:657] [42499] (audio_device_module_ios.mm:615): output: 0 [000:657] [42499] (apm_helpers.cc:79): AGC set to 0 with mode 2 [000:657] [42499] (audio_device_module_ios.mm:628): BuiltInNSIsAvailable [000:657] [42499] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [000:658] [42499] (audio_device_module_ios.mm:631): output: 0 [000:658] [42499] (apm_helpers.cc:104): NS set to 0 [000:658] [42499] (webrtc_voice_engine.cc:471): NetEq capacity is 50 [000:658] [42499] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:658] [42499] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:658] [42499] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:658] [42499] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:658] [42499] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:658] [42499] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:658] [42499] (webrtc_voice_engine.cc:536): Typing detection is enabled? 0 [000:658] [42499] (audio_processing_impl.cc:697): Highpass filter activated: 1 [000:658] [42499] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [000:658] [42499] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [000:659] [42499] (webrtc_voice_engine.cc:1494): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [000:660] [42499] (channel.cc:149): Created channel for 0 [000:661] [42499] (channel.cc:217): BaseChannel::Init_w, media_transport=0 [000:662] [42499] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [000:662] [42499] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [000:662] [42499] (webrtc_video_engine.cc:459): CreateMediaChannel. Options: VideoOptions {} [000:663] [42499] (channel.cc:149): Created channel for 1 [000:664] [42499] (channel.cc:217): BaseChannel::Init_w, media_transport=0 [000:664] [42499] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [000:664] [42499] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [000:664] [42243] (peer_connection.cc:3981): Session: 1660230460349573763 Old state: kStable New state: kHaveLocalOffer (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 1 [000:664] [42499] (channel.cc:875): Setting local voice description [000:665] [42499] (webrtc_voice_engine.cc:1312): 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: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}]} [000:665] [42499] (webrtc_voice_engine.cc:1504): Setting receive voice codecs. [000:666] [42499] (webrtc_voice_engine.cc:1775): AddSendStream: {id:ARDAMSa0;ssrcs:[1056030614];ssrc_groups:;cname:gz57yMrR3CAwNWnn;stream_ids:ARDAMS;} [000:671] [42499] (neteq_impl.cc:119): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=50, min_delay_ms=0, enable_fast_accelerate=false, enable_muted_state=false, enable_rtx_handling=false [000:673] [42499] (audio_coding_module.cc:333): Created [000:677] [42499] (channel_send.cc:701): Not setting media_transport_ rate observers. [000:678] [42499] (audio_send_stream.cc:146): AudioSendStream: 1056030614 [000:678] [42499] (audio_send_stream.cc:226): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1056030614, extmap-allow-mixed: false, extensions: [], c_name: gz57yMrR3CAwNWnn}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: <unset>} [000:679] [42499] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [000:679] [42499] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [000:679] [42499] (channel.cc:680): Add send stream ssrc: 1056030614 [000:680] [42499] (channel.cc:867): Changing voice state, recv=0 send=0 [000:680] [42499] (channel.cc:1013): Setting local video description [000:680] [42499] (webrtc_video_engine.cc:925): SetRecvParameters: {codecs: [VideoCodec[96:H264], VideoCodec[97:rtx], VideoCodec[98:H264], VideoCodec[99:rtx], VideoCodec[100:VP8], VideoCodec[101:rtx], VideoCodec[127:VP9], VideoCodec[123:rtx], VideoCodec[125:red], VideoCodec[122:rtx], VideoCodec[124:ulpfec]], extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 10}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 12}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}]} [000:681] [42499] (webrtc_video_engine.cc:940): Changing recv codecs from {VideoCodec[96:H264], VideoCodec[98:H264], VideoCodec[100:VP8], VideoCodec[102:VP9]} to {VideoCodec[96:H264], VideoCodec[98:H264], VideoCodec[100:VP8], VideoCodec[127:VP9]} [000:681] [42499] (webrtc_video_engine.cc:1044): AddSendStream: {id:ARDAMSv0;ssrcs:[362193604,3849333516];ssrc_groups:{semantics:FID;ssrcs:[362193604,3849333516]};cname:gz57yMrR3CAwNWnn;stream_ids:ARDAMS;} [000:682] [42499] (webrtc_video_engine.cc:1080): SetLocalSsrc on all the receive streams because we added a send stream. [000:682] [42499] (channel.cc:680): Add send stream ssrc: 362193604 [000:682] [42499] (channel.cc:1000): Changing video state, send=0 [000:682] [42243] (peer_connection.cc:5297): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport. [000:683] [42499] (webrtc_voice_engine.cc:1476): Setting voice channel options: AudioOptions {} [000:683] [42499] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [000:683] [42499] (webrtc_voice_engine.cc:331): Always disable AEC on iOS. Use built-in instead. [000:683] [42499] (webrtc_voice_engine.cc:358): Always disable NS on iOS. Use built-in instead. [000:683] [42499] (webrtc_voice_engine.cc:369): Always disable AGC on iOS. Use built-in instead. [000:683] [42499] (audio_device_module_ios.mm:596): BuiltInAECIsAvailable [000:683] [42499] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [000:683] [42499] (audio_device_module_ios.mm:599): output: 0 [000:683] [42499] (audio_processing_impl.cc:697): Highpass filter activated: 1 [000:683] [42499] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [000:683] [42499] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [000:683] [42499] (apm_helpers.cc:90): Echo control set to 0 with mode 0 [000:683] [42499] (audio_device_module_ios.mm:612): BuiltInAGCIsAvailable [000:683] [42499] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [000:683] [42499] (audio_device_module_ios.mm:615): output: 0 [000:683] [42499] (apm_helpers.cc:79): AGC set to 0 with mode 2 [000:683] [42499] (audio_device_module_ios.mm:628): BuiltInNSIsAvailable [000:683] [42499] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [000:683] [42499] (audio_device_module_ios.mm:631): output: 0 [000:683] [42499] (apm_helpers.cc:104): NS set to 0 [000:683] [42499] (webrtc_voice_engine.cc:471): NetEq capacity is 50 [000:683] [42499] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:683] [42499] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:683] [42499] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:683] [42499] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:683] [42499] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:683] [42499] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:683] [42499] (webrtc_voice_engine.cc:536): Typing detection is enabled? 0 [000:683] [42499] (audio_processing_impl.cc:697): Highpass filter activated: 1 [000:684] [42499] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [000:684] [42499] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [000:684] [42499] (webrtc_voice_engine.cc:1494): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [000:684] [42499] (webrtc_video_engine.cc:1003): SetVideoSend (ssrc= 362193604, options: VideoOptions {noise reduction: false, is_screencast : false, }, source = (source)) [000:684] [22531] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [000:684] [22531] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 1 [000:685] [771] (webrtc_sdp.cc:3151): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:685] [771] (webrtc_sdp.cc:3151): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:686] [22531] (basic_port_allocator.cc:108): Filtered out ignored networks: [000:686] [22531] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [000:686] [22531] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [000:686] [22531] (basic_port_allocator.cc:851): Network manager has started [000:686] [22531] (basic_port_allocator.cc:108): Filtered out ignored networks: [000:686] [22531] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [000:686] [22531] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [000:686] [22531] (basic_port_allocator.cc:851): Network manager has started [000:686] [22531] (basic_port_allocator.cc:108): Filtered out ignored networks: [000:686] [22531] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [000:686] [22531] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [000:686] [22531] (basic_port_allocator.cc:766): Allocate ports on 1 networks [000:687] [22531] (basic_port_allocator.cc:108): Filtered out ignored networks: [000:687] [22531] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [000:687] [22531] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [000:687] [22531] (basic_port_allocator.cc:766): Allocate ports on 1 networks [000:687] [22531] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Udp [000:687] [22279] (p2p_transport_channel.cc:489): Set backup connection ping interval to 25000 milliseconds. [000:687] [22279] (p2p_transport_channel.cc:498): Set ICE receiving timeout to 2500 milliseconds [000:687] [22531] (port.cc:318): Port[3800d400::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10 [000:687] [22279] (p2p_transport_channel.cc:505): Set ping most likely connection to 0 [000:688] [22279] (p2p_transport_channel.cc:512): Set stable_writable_connection_ping_interval to 2500 [000:688] [22279] (p2p_transport_channel.cc:525): Set presume writable when fully relayed to 0 [000:688] [22279] (p2p_transport_channel.cc:534): Set regather_on_failed_networks_interval to 300000 [000:688] [22279] (p2p_transport_channel.cc:553): Set receiving_switching_delay to 1000 [000:688] [22279] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0 [000:688] [22279] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 34522700 [000:688] [22279] (p2p_transport_channel.cc:489): Set backup connection ping interval to 25000 milliseconds. [000:688] [22279] (p2p_transport_channel.cc:498): Set ICE receiving timeout to 2500 milliseconds [000:688] [22279] (p2p_transport_channel.cc:505): Set ping most likely connection to 0 [000:688] [22279] (p2p_transport_channel.cc:512): Set stable_writable_connection_ping_interval to 2500 [000:688] [22279] (p2p_transport_channel.cc:525): Set presume writable when fully relayed to 0 [000:688] [22279] (p2p_transport_channel.cc:534): Set regather_on_failed_networks_interval to 300000 [000:688] [22279] (p2p_transport_channel.cc:553): Set receiving_switching_delay to 1000 [000:688] [22279] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 1 transport 0 [000:688] [22279] (dtls_srtp_transport.cc:66): Setting RTP Transport on 1 transport 34523ad0 [000:688] [22279] (p2p_transport_channel.cc:437): Received remote ICE parameters: ufrag=fAzx, renomination enabled [000:688] [22279] (p2p_transport_channel.cc:437): Received remote ICE parameters: ufrag=fAzx, renomination enabled [000:688] [22531] (basic_port_allocator.cc:875): Adding allocated port for 0 [000:689] [22531] (basic_port_allocator.cc:894): Port[3800d400:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator [000:689] [22787] (webrtc_voice_engine.cc:1476): Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [000:689] [22787] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [000:689] [22787] (webrtc_voice_engine.cc:331): Always disable AEC on iOS. Use built-in instead. [000:689] [22787] (webrtc_voice_engine.cc:358): Always disable NS on iOS. Use built-in instead. [000:689] [22787] (webrtc_voice_engine.cc:369): Always disable AGC on iOS. Use built-in instead. [000:689] [22787] (audio_device_module_ios.mm:596): BuiltInAECIsAvailable [000:689] [22787] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [000:689] [22787] (audio_device_module_ios.mm:599): output: 0 [000:689] [22787] (audio_processing_impl.cc:697): Highpass filter activated: 1 [000:689] [22531] (basic_port_allocator.cc:912): Port[3800d400:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:3009622470:1:udp:2122260223:192.168.144.89:57441:local::0:fAzx:QilkW90Q9UtzhCKlhakNY9r/:1:10:0] [000:689] [22531] (basic_port_allocator.cc:940): Port[3800d400:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready. [000:689] [22531] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [000:689] [22787] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [000:689] [22531] (p2p_transport_channel.cc:790): Port[3800d400:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0 [000:689] [22787] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [000:689] [22787] (apm_helpers.cc:90): Echo control set to 0 with mode 0 [000:689] [22787] (audio_device_module_ios.mm:612): BuiltInAGCIsAvailable [000:689] [22787] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [000:689] [22787] (audio_device_module_ios.mm:615): output: 0 [000:689] [22787] (apm_helpers.cc:79): AGC set to 0 with mode 2 [000:689] [22531] (jsep_transport_controller.cc:1297): 0 Transport 1 state changed. Check if state is complete. [000:690] [22787] (audio_device_module_ios.mm:628): BuiltInNSIsAvailable [000:690] [22787] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [000:690] [22787] (audio_device_module_ios.mm:631): output: 0 [000:690] [22787] (apm_helpers.cc:104): NS set to 0 [000:690] [22787] (webrtc_voice_engine.cc:471): NetEq capacity is 50 [000:690] [22787] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:690] [22531] (basic_port_allocator.cc:1017): Port[3800d400:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates. [000:690] [22787] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 1 [000:690] [22531] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Udp [000:690] [22787] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:690] [22531] (port.cc:318): Port[35027000::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10 [000:690] [22787] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:690] [22531] (basic_port_allocator.cc:875): Adding allocated port for 1 [000:690] [22787] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:690] [22531] (basic_port_allocator.cc:894): Port[35027000:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator [000:690] [22787] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:690] [22787] (webrtc_voice_engine.cc:536): Typing detection is enabled? 0 [000:691] [22787] (audio_processing_impl.cc:697): Highpass filter activated: 1 [000:691] [22531] (basic_port_allocator.cc:912): Port[35027000:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:3009622470:1:udp:2122260223:192.168.144.89:54833:local::0:fAzx:QilkW90Q9UtzhCKlhakNY9r/:1:10:0] [000:691] [22531] (basic_port_allocator.cc:940): Port[35027000:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready. [000:691] [22531] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [000:691] [22531] (p2p_transport_channel.cc:790): Port[35027000:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0 [000:691] [22787] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [000:691] [22531] (jsep_transport_controller.cc:1297): 1 Transport 1 state changed. Check if state is complete. [000:691] [22787] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [000:691] [22787] (webrtc_voice_engine.cc:1494): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [000:691] [22787] (channel.cc:149): Created channel for 0 [000:691] [22531] (basic_port_allocator.cc:1017): Port[35027000:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates. [000:692] [22787] (channel.cc:217): BaseChannel::Init_w, media_transport=0 [000:692] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [000:692] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [000:692] [22787] (webrtc_video_engine.cc:459): CreateMediaChannel. Options: VideoOptions {} [000:692] [22787] (channel.cc:149): Created channel for 1 [000:693] [22787] (channel.cc:217): BaseChannel::Init_w, media_transport=0 [000:693] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [000:693] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [000:693] [41987] (peer_connection.cc:3981): Session: 8089159972302902786 Old state: kStable New state: kHaveRemoteOffer (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 3 [000:693] [22787] (channel.cc:927): Setting remote voice description [000:693] [22787] (webrtc_voice_engine.cc:1267): 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: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: 0, options: AudioOptions {}} [000:694] [22787] (webrtc_voice_engine.cc:1693): Recreate all the receive streams because the send codec has changed. [000:694] [22787] (webrtc_voice_engine.cc:2115): WebRtcVoiceMediaChannel::SetMaxSendBitrate. [000:694] [22787] (webrtc_voice_engine.cc:1476): Setting voice channel options: AudioOptions {} [000:694] [22787] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [000:694] [22787] (webrtc_voice_engine.cc:331): Always disable AEC on iOS. Use built-in instead. [000:694] [22787] (webrtc_voice_engine.cc:358): Always disable NS on iOS. Use built-in instead. [000:694] [22787] (webrtc_voice_engine.cc:369): Always disable AGC on iOS. Use built-in instead. [000:694] [22787] (audio_device_module_ios.mm:596): BuiltInAECIsAvailable [000:694] [22787] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [000:694] [22787] (audio_device_module_ios.mm:599): output: 0 [000:694] [22787] (audio_processing_impl.cc:697): Highpass filter activated: 1 [000:694] [22787] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [000:694] [22787] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [000:694] [22787] (apm_helpers.cc:90): Echo control set to 0 with mode 0 [000:694] [22787] (audio_device_module_ios.mm:612): BuiltInAGCIsAvailable [000:694] [22787] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [000:694] [22787] (audio_device_module_ios.mm:615): output: 0 [000:694] [22787] (apm_helpers.cc:79): AGC set to 0 with mode 2 [000:694] [22787] (audio_device_module_ios.mm:628): BuiltInNSIsAvailable [000:694] [22787] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [000:695] [22787] (audio_device_module_ios.mm:631): output: 0 [000:695] [22787] (apm_helpers.cc:104): NS set to 0 [000:695] [22787] (webrtc_voice_engine.cc:471): NetEq capacity is 50 [000:695] [22787] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [000:695] [22787] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [000:695] [22787] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [000:695] [22787] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [000:695] [22787] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [000:695] [22787] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [000:695] [22787] (webrtc_voice_engine.cc:536): Typing detection is enabled? 0 [000:695] [22787] (audio_processing_impl.cc:697): Highpass filter activated: 1 [000:695] [22787] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [000:695] [22787] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [000:695] [22787] (webrtc_voice_engine.cc:1494): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [000:695] [22787] (webrtc_voice_engine.cc:1840): AddRecvStream: {id:ARDAMSa0;ssrcs:[1056030614];ssrc_groups:;cname:gz57yMrR3CAwNWnn;stream_ids:ARDAMS;} [000:697] [22787] (neteq_impl.cc:119): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=50, min_delay_ms=0, enable_fast_accelerate=false, enable_muted_state=true, enable_rtx_handling=false [000:697] [22787] (audio_coding_module.cc:333): Created [000:697] [22787] (audio_receive_stream.cc:112): AudioReceiveStream: 1056030614 [000:697] [22787] (audio_receive_stream.cc:344): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 1056030614, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS} [000:697] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [000:697] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [000:698] [22787] (channel.cc:723): Add remote ssrc: 1056030614 [000:698] [22787] (channel.cc:867): Changing voice state, recv=0 send=0 [000:698] [22787] (channel.cc:1065): Setting remote video description [000:698] [22787] (webrtc_video_engine.cc:663): SetSendParameters: {codecs: [VideoCodec[96:H264], VideoCodec[97:rtx], VideoCodec[98:H264], VideoCodec[99:rtx], VideoCodec[100:VP8], VideoCodec[101:rtx], VideoCodec[127:VP9], VideoCodec[123:rtx], VideoCodec[125:red], VideoCodec[122:rtx], VideoCodec[124:ulpfec]], conference_mode: no, extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 10}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 12}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: 1} [000:699] [22787] (webrtc_video_engine.cc:672): Using codec: VideoCodec[96:H264] [000:699] [22787] (webrtc_video_engine.cc:724): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. [000:699] [22787] (webrtc_video_engine.cc:1145): AddRecvStream: {id:ARDAMSv0;ssrcs:[362193604,3849333516];ssrc_groups:{semantics:FID;ssrcs:[362193604,3849333516]};cname:gz57yMrR3CAwNWnn;stream_ids:ARDAMS;} [000:708] [22787] (video_receive_stream.cc:203): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 102, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 362193604, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 106, red_type: 104, rtx_ssrc: 3849333516, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 103 (pt) -> 102 (apt), 105 (pt) -> 104 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0} [000:709] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [000:709] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [000:713] [22787] (channel.cc:723): Add remote ssrc: 362193604 [000:714] [22787] (channel.cc:1000): Changing video state, send=0 [000:714] [41987] (peer_connection.cc:5297): Local and Remote descriptions must be applied to get the SSL Role of the SCTP transport. [000:714] [41987] (peer_connection.cc:2593): Processing the MSIDs for MID=0 (streams=[ARDAMS]). [000:714] [41987] (peer_connection.cc:2604): Processing the addition of a remote track for MID=0. [000:714] [22787] (webrtc_voice_engine.cc:1953): SetOutputVolume() to 1 for recv stream with ssrc 1056030614 [000:715] [41987] (peer_connection.cc:2593): Processing the MSIDs for MID=1 (streams=[ARDAMS]). [000:715] [41987] (peer_connection.cc:2604): Processing the addition of a remote track for MID=1. [000:715] [22787] (webrtc_video_engine.cc:1275): SetSink: ssrc:362193604 (ptr) [000:715] [41987] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x7faf347216e0): created DTMF sender: RTCDtmfSender { remainingTones: duration: 0.100000 sec interToneGap: 0.050000 sec } [000:715] [41987] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf347213c0): created sender: RTCRtpSender { senderId: ARDAMSa0 } [000:715] [41987] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x7faf3471cbc0): created receiver: RTCRtpReceiver { receiverId: 9810a9db-a4fb-4db4-9675-5a1d69df7fef } [000:715] [41987] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x7faf34721440): created transceiver: RTCRtpTransceiver { sender: RTCRtpSender { senderId: ARDAMSa0 } receiver: RTCRtpReceiver { receiverId: 9810a9db-a4fb-4db4-9675-5a1d69df7fef } } (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving audio on track 9810a9db-a4fb-4db4-9675-5a1d69df7fef. [000:715] [41987] (rtp_sender.cc:561): Tried to get DTMF sender from video sender. [000:716] [41987] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf3471d180): created sender: RTCRtpSender { senderId: ARDAMSv0 } [000:716] [41987] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x7faf3471f060): created receiver: RTCRtpReceiver { receiverId: b2e291d8-140c-4aeb-908d-7e863de2bbcc } [000:716] [41987] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x7faf34721430): created transceiver: RTCRtpTransceiver { sender: RTCRtpSender { senderId: ARDAMSv0 } receiver: RTCRtpReceiver { receiverId: b2e291d8-140c-4aeb-908d-7e863de2bbcc } } (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving video on track b2e291d8-140c-4aeb-908d-7e863de2bbcc. (RTCLogging.mm:33): (ARDAppClient.m:381 -[ARDAppClient peerConnection:didAddStream:]): Stream with 1 video tracks and 1 audio tracks was added. [000:717] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x7faf37486360): created DTMF sender: RTCDtmfSender { remainingTones: duration: 0.100000 sec interToneGap: 0.050000 sec } [000:717] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf374857c0): created sender: RTCRtpSender { senderId: ARDAMSa0 } [000:717] [42243] (rtp_sender.cc:561): Tried to get DTMF sender from video sender. [000:717] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf34516610): created sender: RTCRtpSender { senderId: ARDAMSv0 } [000:720] [41987] (peer_connection.cc:6018): 0 is not ready to use the remote candidate because the local or remote description is not set. [000:720] [41987] (peer_connection.cc:6018): 1 is not ready to use the remote candidate because the local or remote description is not set. [000:720] [41987] (peer_connection.cc:1987): CreateAnswer: offer_to_receive_audio is not supported with Unified Plan semantics. Use the RtpTransceiver API instead. [000:720] [41987] (peer_connection.cc:1992): CreateAnswer: offer_to_receive_video is not supported with Unified Plan semantics. Use the RtpTransceiver API instead. [000:721] [41987] (peer_connection.cc:5324): Local and Remote descriptions must be applied to get the SSL Role of the session. [000:721] [41987] (peer_connection.cc:5324): Local and Remote descriptions must be applied to get the SSL Role of the session. [000:724] [771] (webrtc_sdp.cc:3151): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:724] [771] (webrtc_sdp.cc:3151): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [000:725] [22279] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0 [000:725] [22279] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 34522700 [000:726] [22279] (p2p_transport_channel.cc:426): Set ICE ufrag: K4UY pwd: Wl/yP3Ek+aQWfSEvScC5SdYh on transport 0 [000:726] [22279] (dtls_transport.cc:370): DtlsTransport[0|1|__]: DTLS setup complete. [000:727] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [000:727] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [000:727] [41987] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv. [000:727] [41987] (rtp_transceiver.cc:169): Changing transceiver (MID=1) current direction from <not set> to kSendRecv. [000:727] [22787] (channel.cc:557): Channel enabled [000:727] [22787] (channel.cc:867): Changing voice state, recv=0 send=0 [000:727] [22787] (channel.cc:557): Channel enabled [000:727] [22787] (channel.cc:1000): Changing video state, send=0 [000:727] [41987] (peer_connection.cc:3981): Session: 8089159972302902786 Old state: kHaveRemoteOffer New state: kStable (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 0 [000:728] [22787] (channel.cc:875): Setting local voice description [000:728] [22787] (webrtc_voice_engine.cc:1312): 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: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}]} [000:728] [22787] (webrtc_voice_engine.cc:1504): Setting receive voice codecs. [000:728] [22787] (audio_receive_stream.cc:344): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 1056030614, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS} [000:729] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [000:729] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [000:729] [22787] (audio_receive_stream.cc:135): ~AudioReceiveStream: 1056030614 [000:729] [22787] (neteq_impl.cc:119): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=50, min_delay_ms=0, enable_fast_accelerate=false, enable_muted_state=true, enable_rtx_handling=false [000:729] [22787] (audio_coding_module.cc:333): Created [000:729] [22787] (audio_receive_stream.cc:112): AudioReceiveStream: 1056030614 [000:729] [22787] (audio_receive_stream.cc:344): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 1056030614, local_ssrc: 4195875351, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: [{uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS} [000:730] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [000:730] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [000:730] [22787] (webrtc_voice_engine.cc:1775): AddSendStream: {id:ARDAMSa0;ssrcs:[1694263004];ssrc_groups:;cname:TpAYwgBWogJE85ER;stream_ids:ARDAMS;} [000:730] [22787] (neteq_impl.cc:119): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=50, min_delay_ms=0, enable_fast_accelerate=false, enable_muted_state=false, enable_rtx_handling=false [000:730] [22787] (audio_coding_module.cc:333): Created [000:730] [22787] (channel_send.cc:701): Not setting media_transport_ rate observers. [000:730] [22787] (audio_send_stream.cc:146): AudioSendStream: 1694263004 [000:731] [22787] (audio_send_stream.cc:226): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1694263004, extmap-allow-mixed: false, extensions: [{uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], c_name: TpAYwgBWogJE85ER}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, cng_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {minptime: 10, useinbandfec: 1}}}} [000:736] [22787] (audio_encoder_opus.cc:772): Set Opus bitrate to 32000 bps. [000:736] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [000:736] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [000:736] [22787] (audio_receive_stream.cc:344): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 1056030614, local_ssrc: 1694263004, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: [{uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS} [000:736] [22787] (channel.cc:680): Add send stream ssrc: 1694263004 [000:736] [22787] (audio_device_module_ios.mm:542): Playing [000:737] [22787] (audio_device_module_ios.mm:479): InitPlayout [000:737] [22787] (audio_device_module_ios.mm:505): PlayoutIsInitialized [000:737] [22787] (audio_device_ios.mm:184): AudioDeviceIOS::InitPlayout [000:737] [22787] (audio_device_ios.mm:845): AudioDeviceIOS::InitPlayOrRecord [000:746] [22531] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Relay [000:746] [22531] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Relay [000:763] [22787] (RTCLogging.mm:33): (RTCAudioSession.mm:99 -[RTCAudioSession initWithAudioSession:]): RTCAudioSession (0x7faf34409450): init. [000:763] [22787] (RTCLogging.mm:33): (audio_device_ios.mm:810 ConfigureAudioSession): Configuring audio session. [000:763] [22787] (RTCLogging.mm:33): (RTCAudioSession.mm:699 -[RTCAudioSession configureWebRTCSession:]): Configuring audio session for WebRTC. [000:765] [22787] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:64 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set category to: AVAudioSessionCategoryPlayAndRecord [000:765] [22787] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:75 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set mode to: AVAudioSessionModeVoiceChat [000:765] [22787] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:103 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred sample rate to: 48000.00 [000:765] [22787] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:116 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred IO buffer duration to: 0.020000 [000:765] [22787] (RTCLogging.mm:33): (RTCAudioSession.mm:624 -[RTCAudioSession incrementActivationCount]): Incrementing activation count. [000:765] [22787] (RTCLogging.mm:33): (RTCAudioSession.mm:377 -[RTCAudioSession setActive:error:]): Number of current activations: 1 [000:765] [22787] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:145 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set input number of channels to: 1 [000:765] [22787] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:158 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set output number of channels to: 1 [000:765] [22787] (RTCLogging.mm:33): (audio_device_ios.mm:821 ConfigureAudioSession): Configured audio session. [000:766] [22787] (audio_device_ios.mm:664): AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession [000:766] [22787] (RTCLogging.mm:33): (audio_device_ios.mm:669 SetupAudioBuffersForActiveAudioSession): RTCAudioSession: { category: AVAudioSessionCategoryPlayAndRecord categoryOptions: 4 mode: AVAudioSessionModeVoiceChat isActive: 1 sampleRate: 48000.00 IOBufferDuration: 0.010667 outputNumberOfChannels: 2 inputNumberOfChannels: 2 outputLatency: 0.010000 inputLatency: 0.010000 outputVolume: 1.000000 } [000:766] [22787] (audio_device_ios.mm:700): frames per I/O buffer: 512 [000:766] [22787] (audio_device_ios.mm:701): bytes per I/O buffer: 1024 [000:766] [22787] (audio_device_ios.mm:648): AudioDeviceIOS::UpdateAudioDevicebuffer [000:766] [22787] (audio_device_buffer.cc:184): SetPlayoutSampleRate(48000) [000:766] [22787] (audio_device_buffer.cc:204): SetPlayoutChannels(1) [000:766] [22787] (audio_device_buffer.cc:178): SetRecordingSampleRate(48000) [000:766] [22787] (audio_device_buffer.cc:198): SetRecordingChannels(1) [000:766] [22787] (fine_audio_buffer.cc:32): FineAudioBuffer [000:766] [22787] (fine_audio_buffer.cc:34): playout_samples_per_channel_10ms: 480 [000:766] [22787] (fine_audio_buffer.cc:36): playout_channels: 1 [000:766] [22787] (fine_audio_buffer.cc:39): record_samples_per_channel_10ms: 480 [000:766] [22787] (fine_audio_buffer.cc:41): record_channels: 1 [000:766] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:196 Initialize): Initializing audio unit with sample rate: 48000.000000 [000:766] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:43 LogStreamDescription): AudioStreamBasicDescription: { mSampleRate: 48000.00 formatIDString: lpcm mFormatFlags: 0xC mBytesPerPacket: 2 mFramesPerPacket: 1 mBytesPerFrame: 2 mChannelsPerFrame: 1 mBitsPerChannel: 16 mReserved: 4294934608 } 2019-02-07 13:25:16.823 apprtcmobile_tests[705:7792] 21:25:16.822 WARNING: 40: ERROR: couldn't get default input device, ID = 0, err = 0! 2019-02-07 13:25:16.824 apprtcmobile_tests[705:7792] 21:25:16.824 WARNING: 40: ERROR: couldn't get default output device, ID = 0, err = 0! 2019-02-07 13:25:16.825 apprtcmobile_tests[705:7792] 21:25:16.824 ERROR: 708: Error finding valid input or output devices! 2019-02-07 13:25:16.826 apprtcmobile_tests[705:7792] 21:25:16.826 ERROR: 316: error -66680 2019-02-07 13:25:16.826 apprtcmobile_tests[705:7772] 21:25:16.826 ERROR: 316: error -66680 2019-02-07 13:25:16.827 apprtcmobile_tests[705:7792] 21:25:16.827 ERROR: 316: error -66680 2019-02-07 13:25:16.827 apprtcmobile_tests[705:7772] 21:25:16.827 ERROR: 109: * * * NULL AQIONode object 2019-02-07 13:25:16.827 apprtcmobile_tests[705:7786] 21:25:16.827 ERROR: >aurioc> 807: failed: -10851 (enable 3, outf< 1 ch, 48000 Hz, Int16> inf< 1 ch, 48000 Hz, Int16>) [000:784] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851. [000:784] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again... 2019-02-07 13:25:16.828 apprtcmobile_tests[705:7772] 21:25:16.828 ERROR: 772: Can't make UISound Renderer [000:797] [22531] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Tcp [000:797] [22531] (port.cc:318): Port[37824a00::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10 [000:797] [22531] (basic_port_allocator.cc:875): Adding allocated port for 0 [000:797] [22531] (basic_port_allocator.cc:894): Port[37824a00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator [000:798] [22531] (basic_port_allocator.cc:912): Port[37824a00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:4259368246:1:tcp:1518280447:192.168.144.89:49250:local::0:fAzx:QilkW90Q9UtzhCKlhakNY9r/:1:10:0] [000:798] [22531] (basic_port_allocator.cc:940): Port[37824a00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready. [000:798] [22531] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [000:798] [22531] (p2p_transport_channel.cc:790): Port[37824a00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0 [000:798] [22531] (basic_port_allocator.cc:1017): Port[37824a00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates. [000:798] [22531] (basic_port_allocator.cc:1119): All candidates gathered for 0:1:0 [000:798] [22531] (p2p_transport_channel.cc:845): P2PTransportChannel: 0, component 1 gathering complete [000:798] [22531] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Tcp [000:798] [22531] (port.cc:318): Port[37822600::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10 [000:798] [22531] (basic_port_allocator.cc:875): Adding allocated port for 1 [000:798] [22531] (basic_port_allocator.cc:894): Port[37822600:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator [000:798] [22531] (basic_port_allocator.cc:912): Port[37822600:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:4259368246:1:tcp:1518280447:192.168.144.89:49251:local::0:fAzx:QilkW90Q9UtzhCKlhakNY9r/:1:10:0] [000:798] [22531] (basic_port_allocator.cc:940): Port[37822600:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready. [000:798] [22531] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [000:798] [22531] (p2p_transport_channel.cc:790): Port[37822600:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0 [000:798] [22531] (basic_port_allocator.cc:1017): Port[37822600:1:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates. [000:799] [22531] (basic_port_allocator.cc:1119): All candidates gathered for 1:1:0 [000:799] [22531] (p2p_transport_channel.cc:845): P2PTransportChannel: 1, component 1 gathering complete (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 2 2019-02-07 13:25:16.933 apprtcmobile_tests[705:7792] 21:25:16.933 ERROR: 316: error -66680 2019-02-07 13:25:16.933 apprtcmobile_tests[705:7792] 21:25:16.933 ERROR: 316: error -66680 2019-02-07 13:25:16.933 apprtcmobile_tests[705:7786] 21:25:16.933 ERROR: >aurioc> 807: failed: -10851 (enable 3, outf< 1 ch, 48000 Hz, Int16> inf< 1 ch, 48000 Hz, Int16>) [000:890] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851. [000:890] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again... 2019-02-07 13:25:17.034 apprtcmobile_tests[705:7792] 21:25:17.034 ERROR: 316: error -66680 2019-02-07 13:25:17.034 apprtcmobile_tests[705:7792] 21:25:17.034 ERROR: 316: error -66680 2019-02-07 13:25:17.034 apprtcmobile_tests[705:7786] 21:25:17.034 ERROR: >aurioc> 807: failed: -10851 (enable 3, outf< 1 ch, 48000 Hz, Int16> inf< 1 ch, 48000 Hz, Int16>) [000:991] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851. [000:991] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again... 2019-02-07 13:25:17.140 apprtcmobile_tests[705:7792] 21:25:17.140 ERROR: 316: error -66680 2019-02-07 13:25:17.140 apprtcmobile_tests[705:7792] 21:25:17.140 ERROR: 316: error -66680 2019-02-07 13:25:17.141 apprtcmobile_tests[705:7786] 21:25:17.141 ERROR: >aurioc> 807: failed: -10851 (enable 3, outf< 1 ch, 48000 Hz, Int16> inf< 1 ch, 48000 Hz, Int16>) [001:097] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851. [001:097] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again... 2019-02-07 13:25:17.242 apprtcmobile_tests[705:7792] 21:25:17.241 ERROR: 316: error -66680 2019-02-07 13:25:17.242 apprtcmobile_tests[705:7792] 21:25:17.242 ERROR: 316: error -66680 2019-02-07 13:25:17.242 apprtcmobile_tests[705:7786] 21:25:17.242 ERROR: >aurioc> 807: failed: -10851 (enable 3, outf< 1 ch, 48000 Hz, Int16> inf< 1 ch, 48000 Hz, Int16>) [001:199] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851. [001:199] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:242 Initialize): Too many initialization attempts. [001:199] [22787] (audio_device_module_ios.mm:485): output: 0 [001:199] [22787] (audio_device_module_ios.mm:517): StartPlayout [001:199] [22787] (audio_device_module_ios.mm:542): Playing [001:199] [22787] (audio_device_buffer.cc:96): StartPlayout [001:199] [22787] (audio_device_ios.mm:226): AudioDeviceIOS::StartPlayout [001:199] [22787] (audio_device_module_ios.mm:524): output: 0 [001:200] [22787] (channel.cc:867): Changing voice state, recv=1 send=0 [001:200] [22787] (message_queue.cc:518): Message took 472ms to dispatch. Posted from: SetLocalContent@../../pc/channel.cc:298 [001:200] [22787] (channel.cc:1013): Setting local video description [001:200] [22787] (webrtc_video_engine.cc:925): SetRecvParameters: {codecs: [VideoCodec[96:H264], VideoCodec[97:rtx], VideoCodec[98:H264], VideoCodec[99:rtx], VideoCodec[100:VP8], VideoCodec[101:rtx], VideoCodec[127:VP9], VideoCodec[123:rtx], VideoCodec[125:red], VideoCodec[122:rtx], VideoCodec[124:ulpfec]], extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 10}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 12}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}]} [001:201] [22787] (webrtc_video_engine.cc:940): Changing recv codecs from {VideoCodec[96:H264], VideoCodec[98:H264], VideoCodec[100:VP8], VideoCodec[102:VP9]} to {VideoCodec[96:H264], VideoCodec[98:H264], VideoCodec[100:VP8], VideoCodec[127:VP9]} [001:201] [22787] (webrtc_video_engine.cc:2373): MaybeRecreateWebRtcFlexfecStream (recv) because of SetRecvParameters [001:201] [22787] (webrtc_video_engine.cc:2378): RecreateWebRtcVideoStream (recv) because of SetRecvParameters [001:201] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [001:201] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [001:201] [22787] (video_receive_stream.cc:252): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 102, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 362193604, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 106, red_type: 104, rtx_ssrc: 3849333516, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 103 (pt) -> 102 (apt), 105 (pt) -> 104 (apt), }, extensions: []}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0} [001:202] [22787] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [001:202] [22787] (receive_statistics_proxy.cc:487): Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [001:203] [22787] (video_receive_stream.cc:203): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 127, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 362193604, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 124, red_type: 125, rtx_ssrc: 3849333516, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 122 (pt) -> 125 (apt), 123 (pt) -> 127 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 10}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 12}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0} [001:203] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [001:203] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [001:204] [22787] (webrtc_video_engine.cc:1044): AddSendStream: {id:ARDAMSv0;ssrcs:[115403994,848457161];ssrc_groups:{semantics:FID;ssrcs:[115403994,848457161]};cname:TpAYwgBWogJE85ER;stream_ids:ARDAMS;} [001:204] [22787] (webrtc_video_engine.cc:1744): RecreateWebRtcStream (send) because of SetCodec. [001:206] [22787] (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 [001:207] [8711] (rtp_video_sender.cc:463): Transmitting payload type without picture ID using NACK+ULPFEC is a waste of bandwidth since ULPFEC packets also have to be retransmitted. Disabling ULPFEC. [001:208] [8711] (video_send_stream_impl.cc:250): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [115403994], 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: 10}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 12}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 124, red_payload_type: 125, red_rtx_payload_type: 122}, payload_name: H264, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [848457161], payload_type: 97}, c_name: TpAYwgBWogJE85ER}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [001:208] [8711] (video_send_stream_impl.cc:276): ERROR: Initial encoder max bitrate = -1 which is <= 0! [001:208] [8711] (video_send_stream_impl.cc:278): Using default encoder max bitrate = 10 Mbps [001:209] [8711] (video_stream_encoder.cc:502): ConfigureEncoder requested. [001:209] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [001:209] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [001:209] [22787] (video_send_stream.cc:159): VideoSendStream::Stop [001:209] [22787] (webrtc_video_engine.cc:1080): SetLocalSsrc on all the receive streams because we added a send stream. [001:210] [22787] (webrtc_video_engine.cc:2314): RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc=115403994 [001:210] [8711] (video_send_stream_impl.cc:423): VideoSendStream::Stop [001:210] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [001:210] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [001:210] [22787] (video_receive_stream.cc:252): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 127, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 362193604, local_ssrc: 1, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 124, red_type: 125, rtx_ssrc: 3849333516, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 122 (pt) -> 125 (apt), 123 (pt) -> 127 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 10}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 12}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0} [001:211] [22787] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [001:211] [22787] (receive_statistics_proxy.cc:487): Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [001:212] [22787] (video_receive_stream.cc:203): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 127, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 362193604, local_ssrc: 115403994, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 124, red_type: 125, rtx_ssrc: 3849333516, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 122 (pt) -> 125 (apt), 123 (pt) -> 127 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 10}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 12}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0} [001:212] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [001:212] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [001:212] [22787] (channel.cc:680): Add send stream ssrc: 115403994 [001:213] [22787] (video_send_stream.cc:159): VideoSendStream::Stop [001:213] [22787] (channel.cc:1000): Changing video state, send=0 [001:213] [8711] (video_send_stream_impl.cc:423): VideoSendStream::Stop [001:213] [41987] (peer_connection.cc:3911): Changing IceConnectionState 0 => 1 (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 1 [001:213] [22279] (jsep_transport_controller.cc:319): Not adding candidate because the JsepTransport doesn't exist. Ignore it. [001:213] [41987] (peer_connection.cc:5303): Non-rejected SCTP m= section is needed to get the SSL Role of the SCTP transport. [001:213] [22787] (webrtc_voice_engine.cc:1476): Setting voice channel options: AudioOptions {} [001:213] [22787] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [001:213] [22787] (webrtc_voice_engine.cc:331): Always disable AEC on iOS. Use built-in instead. [001:213] [22787] (webrtc_voice_engine.cc:358): Always disable NS on iOS. Use built-in instead. [001:213] [22787] (webrtc_voice_engine.cc:369): Always disable AGC on iOS. Use built-in instead. [001:213] [22787] (audio_device_module_ios.mm:596): BuiltInAECIsAvailable [001:213] [22787] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [001:213] [22787] (audio_device_module_ios.mm:599): output: 0 [001:214] [22787] (audio_processing_impl.cc:697): Highpass filter activated: 1 [001:214] [22787] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [001:214] [22787] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [001:214] [22787] (apm_helpers.cc:90): Echo control set to 0 with mode 0 [001:214] [22787] (audio_device_module_ios.mm:612): BuiltInAGCIsAvailable [001:214] [22787] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [001:214] [22787] (audio_device_module_ios.mm:615): output: 0 [001:214] [22787] (apm_helpers.cc:79): AGC set to 0 with mode 2 [001:214] [22787] (audio_device_module_ios.mm:628): BuiltInNSIsAvailable [001:214] [22787] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [001:214] [22787] (audio_device_module_ios.mm:631): output: 0 [001:214] [22787] (apm_helpers.cc:104): NS set to 0 [001:214] [22787] (webrtc_voice_engine.cc:471): NetEq capacity is 50 [001:214] [22787] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [001:214] [22787] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [001:214] [22787] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [001:214] [22787] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [001:214] [22787] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [001:214] [22787] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [001:214] [22787] (webrtc_voice_engine.cc:536): Typing detection is enabled? 0 [001:214] [22787] (audio_processing_impl.cc:697): Highpass filter activated: 1 [001:214] [22787] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [001:214] [22787] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [001:214] [22787] (webrtc_voice_engine.cc:1494): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [001:215] [22787] (webrtc_video_engine.cc:1003): SetVideoSend (ssrc= 115403994, options: VideoOptions {noise reduction: false, is_screencast : false, }, source = (source)) [001:215] [8711] (video_stream_encoder.cc:502): ConfigureEncoder requested. [001:215] [22279] (basic_port_allocator.cc:339): Start getting ports with prune_turn_ports disabled [001:216] [22279] (basic_port_allocator.cc:108): Filtered out ignored networks: [001:216] [22279] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [001:216] [22279] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [001:216] [22279] (basic_port_allocator.cc:851): Network manager has started [001:216] [22279] (basic_port_allocator.cc:108): Filtered out ignored networks: [001:216] [22279] (basic_port_allocator.cc:110): Net[lo0:::1/128:Loopback:id=3] [001:216] [41987] (message_queue.cc:518): Message took 491ms to dispatch. Posted from: SetLocalDescription@../../api/peer_connection_proxy.h:100 [001:216] [22279] (basic_port_allocator.cc:110): Net[lo0:127.0.0.0/8:Loopback:id=2] [001:216] [22279] (basic_port_allocator.cc:766): Allocate ports on 1 networks (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 1 [001:216] [22279] (basic_port_allocator.cc:1318): Net[en0:192.168.144.0/22:Wifi:id=1]: Allocation Phase=Udp [001:216] [22279] (port.cc:318): Port[38838e00::1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port created with network cost 10 [001:216] [22279] (basic_port_allocator.cc:875): Adding allocated port for 0 [001:216] [771] (webrtc_sdp.cc:3151): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [001:216] [22279] (basic_port_allocator.cc:894): Port[38838e00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Added port to allocator [001:216] [22279] (basic_port_allocator.cc:912): Port[38838e00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Gathered candidate: Cand[:3009622470:1:udp:2122260223:192.168.144.89:52008:local::0:K4UY:Wl/yP3Ek+aQWfSEvScC5SdYh:1:10:0] [001:216] [22279] (basic_port_allocator.cc:940): Port[38838e00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port ready. [001:216] [22279] (physical_socket_server.cc:553): Socket::OPT_DSCP not supported. [001:216] [22279] (p2p_transport_channel.cc:790): Port[38838e00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: SetOption(5, 0) failed: 0 [001:217] [22279] (port.cc:1126): Conn[38839400:0:Net[en0:192.168.144.0/22:Wifi:id=1]:+UhPDYRv:1:0:local:udp:192.168.144.89:52008->GXGegV20:1:2122260223:local:udp:192.168.144.89:57441|C--W|-|0|0|9115038255631187454|-]: Connection created [001:217] [771] (webrtc_sdp.cc:3151): Ignored line: a=rtcp:9 IN IP4 0.0.0.0 [001:217] [22279] (p2p_transport_channel.cc:1255): Channel[0|1|__]: Created connection with origin: 2, total: 1 [001:217] [22279] (p2p_transport_channel.cc:1905): Channel[0|1|__]: Transport channel state changed from 0 to 2 [001:217] [22279] (jsep_transport_controller.cc:1297): 0 Transport 1 state changed. Check if state is complete. [001:217] [22279] (jsep_transport_controller.cc:1297): 0 Transport 1 state changed. Check if state is complete. [001:217] [22279] (p2p_transport_channel.cc:1467): Channel[0|1|__]: Have a pingable connection for the first time; starting to ping. (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 1 [001:218] [22279] (basic_port_allocator.cc:1017): Port[38838e00:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Port completed gathering candidates. [001:219] [22279] (port.cc:1756): Conn[38839400:0:Net[en0:192.168.144.0/22:Wifi:id=1]:+UhPDYRv:1:0:local:udp:192.168.144.89:52008->GXGegV20:1:2122260223:local:udp:192.168.144.89:57441|C--W|-|0|0|9115038255631187454|-]: Sent STUN ping, id=796a6b76756d495a4434576d, use_candidate=0, nomination=0 [001:219] [22531] (port.cc:528): Received STUN ping id=796a6b76756d495a4434576d from unknown address 192.168.144.89:52008 [001:219] [22531] (port.cc:1126): Conn[35024000:0:Net[en0:192.168.144.0/22:Wifi:id=1]:cmIshCi3:1:0:local:udp:192.168.144.89:57441->WcuOGs0o:1:1853824767:prflx:udp:192.168.144.89:52008|C--W|-|0|0|7962116751024340479|-]: Connection created [001:219] [22531] (p2p_transport_channel.cc:979): Adding connection from peer reflexive candidate: Cand[:1408031891:1:udp:1853824767:192.168.144.89:52008:prflx::0:K4UY::1:10:0] [001:220] [22531] (port.cc:853): Port[3800d400:0:1:0:local:Net[en0:192.168.144.0/22:Wifi:id=1]]: Sent STUN ping response, to=192.168.144.89:52008, id=796a6b76756d495a4434576d [001:220] [22531] (p2p_transport_channel.cc:1905): Channel[0|1|R_]: Transport channel state changed from 0 to 2 [001:220] [22531] (jsep_transport_controller.cc:1297): 0 Transport 1 state changed. Check if state is complete. [001:220] [22279] (port.cc:1704): Conn[38839400:0:Net[en0:192.168.144.0/22:Wifi:id=1]:+UhPDYRv:1:0:local:udp:192.168.144.89:52008->GXGegV20:1:2122260223:local:udp:192.168.144.89:57441|C--I|-|0|0|9115038255631187454|-]: Received STUN ping response, id=796a6b76756d495a4434576d, code=0, rtt=2, pings_since_last_response=796a6b76756d495a4434576d [001:220] [22531] (jsep_transport_controller.cc:1297): 0 Transport 1 state changed. Check if state is complete. [001:220] [22279] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed [001:220] [22279] (p2p_transport_channel.cc:1843): Channel[0|1|__]: New selected connection: Conn[38839400:0:Net[en0:192.168.144.0/22:Wifi:id=1]:+UhPDYRv:1:0:local:udp:192.168.144.89:52008->GXGegV20:1:2122260223:local:udp:192.168.144.89:57441|CRWS|S|0|0|9115038255631187454|2] [001:220] [22279] (channel.cc:381): Network route was changed. [001:220] [22279] (channel.cc:381): Network route was changed. [001:220] [22279] (dtls_transport.cc:811): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 2 [001:220] [22279] (openssl_stream_adapter.cc:784): BeginSSL with peer. [001:223] [42499] (webrtc_video_engine.cc:1322): Call stats: 128694, {send_bw_bps: 0, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1} [001:224] [22531] (dtls_srtp_transport.cc:61): Setting RTCP Transport on 0 transport 0 [001:224] [22531] (dtls_srtp_transport.cc:66): Setting RTP Transport on 0 transport 3745d430 [001:224] [22531] (p2p_transport_channel.cc:437): Received remote ICE parameters: ufrag=K4UY, renomination enabled [001:224] [22531] (dtls_transport.cc:370): DtlsTransport[0|1|__]: DTLS setup complete. [001:224] [42499] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [001:224] [42499] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [001:225] [22531] (p2p_transport_channel.cc:1467): Channel[0|1|R_]: Have a pingable connection for the first time; starting to ping. [001:225] [22531] (p2p_transport_channel.cc:2431): Selecting connection for triggered check: Conn[35024000:0:Net[en0:192.168.144.0/22:Wifi:id=1]:cmIshCi3:1:0:local:udp:192.168.144.89:57441->WcuOGs0o:1:1853824767:prflx:udp:192.168.144.89:52008|CR-W|-|0|0|7962116751024340479|-] [001:225] [42499] (channel.cc:557): Channel enabled [001:225] [22531] (port.cc:1756): Conn[35024000:0:Net[en0:192.168.144.0/22:Wifi:id=1]:cmIshCi3:1:0:local:udp:192.168.144.89:57441->WcuOGs0o:1:1853824767:prflx:udp:192.168.144.89:52008|CR-W|-|0|0|7962116751024340479|-]: Sent STUN ping, id=4d2f4a2b34505675412b6853, use_candidate=0, nomination=0 [001:225] [42499] (channel.cc:867): Changing voice state, recv=1 send=0 [001:225] [42499] (channel.cc:557): Channel enabled [001:225] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client enter_early_data [001:226] [42499] (channel.cc:1000): Changing video state, send=0 [001:226] [42243] (peer_connection.cc:3981): Session: 1660230460349573763 Old state: kHaveLocalOffer New state: kStable [001:226] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client read_hello_verify_request [001:226] [22531] (dtls_transport.cc:546): DtlsTransport[0|1|__]: Packet received before DTLS started. [001:226] [22279] (openssl_adapter.cc:794): SSL_connect:error in TLS client read_hello_verify_request (RTCLogging.mm:33): (ARDAppClient.m:374 -[ARDAppClient peerConnection:didChangeSignalingState:]): Signaling state changed: 0 [001:226] [22531] (dtls_transport.cc:555): DtlsTransport[0|1|__]: Caching DTLS ClientHello packet until DTLS is started. [001:226] [22279] (dtls_transport.cc:698): DtlsTransport[0|1|__]: DtlsTransport: Started DTLS handshake [001:226] [42499] (channel.cc:927): Setting remote voice description [001:226] [22279] (srtp_transport.cc:363): The params in SRTP transport are reset. [001:226] [42499] (webrtc_voice_engine.cc:1267): 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: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: 0, options: AudioOptions {}} [001:226] [22279] (basic_port_allocator.cc:1119): All candidates gathered for 0:1:0 [001:226] [22279] (p2p_transport_channel.cc:845): P2PTransportChannel: 0, component 1 gathering complete [001:226] [42499] (audio_send_stream.cc:226): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1056030614, extmap-allow-mixed: false, extensions: [], c_name: gz57yMrR3CAwNWnn}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, cng_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {minptime: 10, useinbandfec: 1}}}} (RTCLogging.mm:33): (ARDAppClient.m:414 -[ARDAppClient peerConnection:didChangeIceGatheringState:]): ICE gathering state changed: 2 [001:227] [22531] (port.cc:1704): Conn[35024000:0:Net[en0:192.168.144.0/22:Wifi:id=1]:cmIshCi3:1:0:local:udp:192.168.144.89:57441->WcuOGs0o:1:1853824767:prflx:udp:192.168.144.89:52008|CR-I|-|0|0|7962116751024340479|-]: Received STUN ping response, id=4d2f4a2b34505675412b6853, code=0, rtt=2, pings_since_last_response=4d2f4a2b34505675412b6853 [001:227] [42499] (audio_encoder_opus.cc:772): Set Opus bitrate to 32000 bps. [001:227] [22531] (p2p_transport_channel.cc:272): Switching selected connection due to: candidate pair state changed [001:227] [42499] (webrtc_voice_engine.cc:1693): Recreate all the receive streams because the send codec has changed. [001:227] [22531] (p2p_transport_channel.cc:1843): Channel[0|1|R_]: New selected connection: Conn[35024000:0:Net[en0:192.168.144.0/22:Wifi:id=1]:cmIshCi3:1:0:local:udp:192.168.144.89:57441->WcuOGs0o:1:1853824767:prflx:udp:192.168.144.89:52008|CRWS|S|0|0|7962116751024340479|2] [001:227] [22531] (channel.cc:381): Network route was changed. [001:228] [22531] (channel.cc:381): Network route was changed. [001:228] [42499] (audio_send_stream.cc:226): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1056030614, extmap-allow-mixed: false, extensions: [{uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], c_name: gz57yMrR3CAwNWnn}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, cng_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {minptime: 10, useinbandfec: 1}}}} [001:228] [22531] (dtls_transport.cc:811): DtlsTransport[0|1|__]: configuring DTLS handshake timeout 50 based on ICE RTT 2 [001:228] [22531] (openssl_stream_adapter.cc:784): BeginSSL with peer. [001:228] [42499] (audio_send_stream.cc:226): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1056030614, extmap-allow-mixed: false, extensions: [{uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], c_name: gz57yMrR3CAwNWnn}, rtcp_report_interval_ms: 5000, send_transport: (Transport), media_transport: null, min_bitrate_bps: -1, max_bitrate_bps: -1, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, cng_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {minptime: 10, useinbandfec: 1}}}} [001:228] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_hello [001:228] [22531] (openssl_adapter.cc:794): SSL_accept:error in TLS server read_client_hello [001:228] [42499] (webrtc_voice_engine.cc:2115): WebRtcVoiceMediaChannel::SetMaxSendBitrate. [001:229] [22531] (dtls_transport.cc:698): DtlsTransport[0|1|__]: DtlsTransport: Started DTLS handshake [001:229] [42499] (webrtc_voice_engine.cc:1476): Setting voice channel options: AudioOptions {} [001:229] [42499] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [001:229] [22531] (srtp_transport.cc:363): The params in SRTP transport are reset. [001:229] [42499] (webrtc_voice_engine.cc:331): Always disable AEC on iOS. Use built-in instead. [001:229] [22531] (dtls_transport.cc:704): DtlsTransport[0|1|__]: Handling cached DTLS ClientHello packet. [001:229] [42499] (webrtc_voice_engine.cc:358): Always disable NS on iOS. Use built-in instead. [001:229] [42499] (webrtc_voice_engine.cc:369): Always disable AGC on iOS. Use built-in instead. [001:229] [42499] (audio_device_module_ios.mm:596): BuiltInAECIsAvailable [001:229] [42499] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [001:229] [42499] (audio_device_module_ios.mm:599): output: 0 [001:229] [42499] (audio_processing_impl.cc:697): Highpass filter activated: 1 [001:229] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server select_certificate [001:229] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server select_parameters [001:230] [42499] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [001:230] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_hello [001:230] [42499] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [001:230] [42499] (apm_helpers.cc:90): Echo control set to 0 with mode 0 [001:230] [42499] (audio_device_module_ios.mm:612): BuiltInAGCIsAvailable [001:230] [42499] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [001:230] [42499] (audio_device_module_ios.mm:615): output: 0 [001:230] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_certificate [001:230] [42499] (apm_helpers.cc:79): AGC set to 0 with mode 2 [001:230] [42499] (audio_device_module_ios.mm:628): BuiltInNSIsAvailable [001:230] [42499] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [001:230] [42499] (audio_device_module_ios.mm:631): output: 0 [001:230] [42499] (apm_helpers.cc:104): NS set to 0 [001:230] [42499] (webrtc_voice_engine.cc:471): NetEq capacity is 50 [001:230] [42499] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [001:230] [42499] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [001:230] [42499] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [001:230] [42499] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [001:230] [42499] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [001:230] [42499] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [001:230] [42499] (webrtc_voice_engine.cc:536): Typing detection is enabled? 0 [001:230] [42499] (audio_processing_impl.cc:697): Highpass filter activated: 1 [001:230] [42499] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [001:230] [42499] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [001:230] [42499] (webrtc_voice_engine.cc:1494): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [001:230] [42499] (webrtc_voice_engine.cc:1840): AddRecvStream: {id:ARDAMSa0;ssrcs:[1694263004];ssrc_groups:;cname:TpAYwgBWogJE85ER;stream_ids:ARDAMS;} [001:231] [42499] (neteq_impl.cc:119): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=50, min_delay_ms=0, enable_fast_accelerate=false, enable_muted_state=true, enable_rtx_handling=false [001:231] [42499] (audio_coding_module.cc:333): Created [001:231] [42499] (audio_receive_stream.cc:112): AudioReceiveStream: 1694263004 [001:231] [42499] (audio_receive_stream.cc:344): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 1694263004, local_ssrc: 1056030614, transport_cc: on, nack: {rtp_history_ms: 0}, extensions: [{uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}]}, rtcp_send_transport: (Transport), media_transport: null, sync_group: ARDAMS} [001:231] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_key_exchange [001:231] [42499] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [001:231] [42499] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [001:231] [42499] (audio_device_module_ios.mm:542): Playing [001:231] [42499] (audio_device_module_ios.mm:479): InitPlayout [001:231] [42499] (audio_device_module_ios.mm:505): PlayoutIsInitialized [001:231] [42499] (audio_device_ios.mm:184): AudioDeviceIOS::InitPlayout [001:231] [42499] (audio_device_ios.mm:845): AudioDeviceIOS::InitPlayOrRecord [001:232] [42499] (RTCLogging.mm:33): (audio_device_ios.mm:810 ConfigureAudioSession): Configuring audio session. [001:232] [42499] (RTCLogging.mm:33): (RTCAudioSession.mm:699 -[RTCAudioSession configureWebRTCSession:]): Configuring audio session for WebRTC. [001:232] [42499] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:116 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set preferred IO buffer duration to: 0.020000 [001:232] [42499] (RTCLogging.mm:33): (RTCAudioSession.mm:624 -[RTCAudioSession incrementActivationCount]): Incrementing activation count. [001:232] [42499] (RTCLogging.mm:33): (RTCAudioSession.mm:377 -[RTCAudioSession setActive:error:]): Number of current activations: 2 [001:232] [42499] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:145 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set input number of channels to: 1 [001:232] [42499] (RTCLogging.mm:33): (RTCAudioSession+Configuration.mm:158 -[RTCAudioSession(Configuration) setConfiguration:active:shouldSetActive:error:]): Set output number of channels to: 1 [001:232] [42499] (RTCLogging.mm:33): (audio_device_ios.mm:821 ConfigureAudioSession): Configured audio session. [001:232] [42499] (audio_device_ios.mm:664): AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession [001:232] [42499] (RTCLogging.mm:33): (audio_device_ios.mm:669 SetupAudioBuffersForActiveAudioSession): RTCAudioSession: { category: AVAudioSessionCategoryPlayAndRecord categoryOptions: 4 mode: AVAudioSessionModeVoiceChat isActive: 1 sampleRate: 48000.00 IOBufferDuration: 0.010667 outputNumberOfChannels: 2 inputNumberOfChannels: 2 outputLatency: 0.010000 inputLatency: 0.010000 outputVolume: 1.000000 } [001:233] [42499] (audio_device_ios.mm:700): frames per I/O buffer: 512 [001:233] [42499] (audio_device_ios.mm:701): bytes per I/O buffer: 1024 [001:233] [42499] (audio_device_ios.mm:648): AudioDeviceIOS::UpdateAudioDevicebuffer [001:233] [42499] (audio_device_buffer.cc:184): SetPlayoutSampleRate(48000) [001:233] [42499] (audio_device_buffer.cc:204): SetPlayoutChannels(1) [001:233] [42499] (audio_device_buffer.cc:178): SetRecordingSampleRate(48000) [001:233] [42499] (audio_device_buffer.cc:198): SetRecordingChannels(1) [001:233] [42499] (fine_audio_buffer.cc:32): FineAudioBuffer [001:233] [42499] (fine_audio_buffer.cc:34): playout_samples_per_channel_10ms: 480 [001:233] [42499] (fine_audio_buffer.cc:36): playout_channels: 1 [001:233] [42499] (fine_audio_buffer.cc:39): record_samples_per_channel_10ms: 480 [001:233] [42499] (fine_audio_buffer.cc:41): record_channels: 1 [001:233] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:196 Initialize): Initializing audio unit with sample rate: 48000.000000 [001:233] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:43 LogStreamDescription): AudioStreamBasicDescription: { mSampleRate: 48000.00 formatIDString: lpcm mFormatFlags: 0xC mBytesPerPacket: 2 mFramesPerPacket: 1 mBytesPerFrame: 2 mChannelsPerFrame: 1 mBitsPerChannel: 16 mReserved: 4294934608 } 2019-02-07 13:25:17.277 apprtcmobile_tests[705:7792] 21:25:17.277 ERROR: 316: error -66680 2019-02-07 13:25:17.277 apprtcmobile_tests[705:7792] 21:25:17.277 ERROR: 316: error -66680 2019-02-07 13:25:17.277 apprtcmobile_tests[705:7780] 21:25:17.277 ERROR: >aurioc> 807: failed: -10851 (enable 3, outf< 1 ch, 48000 Hz, Int16> inf< 1 ch, 48000 Hz, Int16>) [001:234] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851. [001:234] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again... [001:235] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_hello_done [001:235] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_certificate [001:235] [22531] (openssl_adapter.cc:794): SSL_accept:error in TLS server read_client_certificate [001:235] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_hello [001:235] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_certificate [001:235] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client read_certificate_status [001:235] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client verify_server_certificate [001:236] [22279] (openssl_stream_adapter.cc:1055): Accepted peer certificate. [001:236] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_key_exchange [001:236] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client read_certificate_request [001:236] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_hello_done [001:236] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_certificate [001:236] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_key_exchange [001:237] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_certificate_verify [001:240] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client send_client_finished [001:241] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client finish_flight [001:241] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client read_session_ticket [001:241] [22279] (openssl_adapter.cc:794): SSL_connect:error in TLS client read_session_ticket [001:241] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server verify_client_certificate [001:242] [22531] (openssl_stream_adapter.cc:1055): Accepted peer certificate. [001:242] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_key_exchange [001:242] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_certificate_verify [001:243] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server read_change_cipher_spec [001:243] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server process_change_cipher_spec [001:243] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server read_next_proto [001:243] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server read_channel_id [001:243] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server read_client_finished [001:243] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server send_server_finished [001:243] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server finish_server_handshake [001:243] [22531] (openssl_adapter.cc:784): SSL_accept:TLS server done [001:243] [22531] (dtls_transport.cc:630): DtlsTransport[0|1|__]: DTLS handshake complete. [001:244] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client process_change_cipher_spec [001:244] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client read_server_finished [001:244] [22531] (jsep_transport_controller.cc:1234): Transport 0 writability changed to 1. [001:244] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client finish_client_handshake [001:244] [22279] (openssl_adapter.cc:784): SSL_connect:TLS client done [001:244] [22531] (dtls_srtp_transport.cc:219): Extracting keys from transport: 0 [001:244] [22279] (dtls_transport.cc:630): DtlsTransport[0|1|__]: DTLS handshake complete. (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 2 [001:244] [22279] (jsep_transport_controller.cc:1234): Transport 0 writability changed to 1. [001:244] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=up [001:244] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Up [001:244] [22279] (dtls_srtp_transport.cc:219): Extracting keys from transport: 0 [001:244] [41987] (peer_connection.cc:5850): Changing to ICE connected state because all transports are writable. [001:244] [41987] (peer_connection.cc:3911): Changing IceConnectionState 1 => 2 [001:244] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=up [001:244] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Up (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 2 [001:244] [12583] (rtp_transport_controller_send.cc:477): Creating fallback congestion controller [001:245] [12583] (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 [001:245] [12583] (aimd_rate_control.cc:101): Using aimd rate control with back off factor 0.85 [001:245] [12583] (delay_based_bwe.cc:104): Using Trendline filter for delay change estimation with window size 20 [001:246] [12583] (delay_based_bwe.cc:290): BWE Setting start bitrate to: 300 kbps [001:246] [12583] (probe_controller.cc:210): Measured bitrate: 300000 Minimum to probe further: 1260000 [001:246] [12583] (bitrate_prober.cc:111): Probe cluster (bitrate:min bytes:min packets): (900000:1687:5) [001:246] [12583] (bitrate_prober.cc:111): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5) [001:246] [12583] (bitrate_allocator.cc:115): Current BWE 300000 [001:251] [22279] (srtp_transport.cc:308): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [001:251] [22531] (srtp_transport.cc:308): SRTP activated with negotiated parameters: send cipher_suite 1 recv cipher_suite 1 [001:251] [22279] (channel.cc:587): Channel writable (0) for the first time [001:251] [22531] (channel.cc:587): Channel writable (0) for the first time [001:251] [22531] (channel.cc:587): Channel writable (1) for the first time [001:251] [22279] (channel.cc:587): Channel writable (1) for the first time [001:251] [22787] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [001:251] [22787] (webrtc_voice_engine.cc:331): Always disable AEC on iOS. Use built-in instead. [001:252] [22787] (webrtc_voice_engine.cc:358): Always disable NS on iOS. Use built-in instead. [001:252] [22787] (webrtc_voice_engine.cc:369): Always disable AGC on iOS. Use built-in instead. [001:252] [22787] (audio_device_module_ios.mm:596): BuiltInAECIsAvailable [001:252] [22787] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [001:252] [22787] (audio_device_module_ios.mm:599): output: 0 [001:252] [22787] (audio_processing_impl.cc:697): Highpass filter activated: 1 [001:252] [22787] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [001:252] [22787] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [001:252] [22787] (apm_helpers.cc:90): Echo control set to 0 with mode 0 [001:252] [22787] (audio_device_module_ios.mm:612): BuiltInAGCIsAvailable [001:252] [22787] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [001:252] [22787] (audio_device_module_ios.mm:615): output: 0 [001:252] [22787] (apm_helpers.cc:79): AGC set to 0 with mode 2 [001:252] [22787] (audio_device_module_ios.mm:628): BuiltInNSIsAvailable [001:252] [22787] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [001:252] [22787] (audio_device_module_ios.mm:631): output: 0 [001:252] [22787] (apm_helpers.cc:104): NS set to 0 [001:252] [22787] (webrtc_voice_engine.cc:471): NetEq capacity is 50 [001:252] [22787] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [001:252] [22787] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [001:252] [22787] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [001:252] [22787] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [001:252] [22787] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [001:252] [22787] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [001:252] [22787] (webrtc_voice_engine.cc:536): Typing detection is enabled? 0 [001:252] [22787] (audio_processing_impl.cc:697): Highpass filter activated: 1 [001:253] [22787] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [001:253] [22787] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [001:253] [22787] (audio_device_module_ios.mm:511): RecordingIsInitialized [001:253] [22787] (audio_device_module_ios.mm:573): Recording [001:253] [22787] (audio_device_module_ios.mm:492): InitRecording [001:253] [22787] (audio_device_module_ios.mm:511): RecordingIsInitialized [001:253] [22787] (audio_device_module_ios.mm:548): StartRecording [001:253] [22787] (audio_device_module_ios.mm:573): Recording [001:253] [22787] (audio_device_buffer.cc:115): StartRecording [001:253] [22787] (audio_device_ios.mm:279): AudioDeviceIOS::StartRecording [001:253] [22787] (audio_device_module_ios.mm:555): output: 0 [001:253] [22787] (channel.cc:867): Changing voice state, recv=1 send=1 [001:253] [22787] (video_send_stream.cc:132): VideoSendStream::UpdateActiveSimulcastLayers [001:253] [12583] (video_send_stream_impl.cc:364): VideoSendStream::UpdateActiveSimulcastLayers [001:253] [12583] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps [001:254] [22787] (channel.cc:1000): Changing video state, send=1 [001:254] [8711] (video_stream_encoder.cc:1151): Video suspend state changed to: not suspended 2019-02-07 13:25:17.381 apprtcmobile_tests[705:7792] 21:25:17.381 ERROR: 316: error -66680 2019-02-07 13:25:17.381 apprtcmobile_tests[705:7792] 21:25:17.381 ERROR: 316: error -66680 2019-02-07 13:25:17.382 apprtcmobile_tests[705:7780] 21:25:17.382 ERROR: >aurioc> 807: failed: -10851 (enable 3, outf< 1 ch, 48000 Hz, Int16> inf< 1 ch, 48000 Hz, Int16>) [001:338] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851. [001:338] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again... 2019-02-07 13:25:17.487 apprtcmobile_tests[705:7792] 21:25:17.487 ERROR: 316: error -66680 2019-02-07 13:25:17.488 apprtcmobile_tests[705:7792] 21:25:17.488 ERROR: 316: error -66680 2019-02-07 13:25:17.488 apprtcmobile_tests[705:7780] 21:25:17.488 ERROR: >aurioc> 807: failed: -10851 (enable 3, outf< 1 ch, 48000 Hz, Int16> inf< 1 ch, 48000 Hz, Int16>) [001:445] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851. [001:445] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again... 2019-02-07 13:25:17.593 apprtcmobile_tests[705:7792] 21:25:17.593 ERROR: 316: error -66680 2019-02-07 13:25:17.594 apprtcmobile_tests[705:7792] 21:25:17.594 ERROR: 316: error -66680 2019-02-07 13:25:17.594 apprtcmobile_tests[705:7780] 21:25:17.594 ERROR: >aurioc> 807: failed: -10851 (enable 3, outf< 1 ch, 48000 Hz, Int16> inf< 1 ch, 48000 Hz, Int16>) [001:551] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851. [001:551] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:245 Initialize): Pause 100ms and try audio unit initialization again... 2019-02-07 13:25:17.699 apprtcmobile_tests[705:7792] 21:25:17.699 ERROR: 316: error -66680 2019-02-07 13:25:17.699 apprtcmobile_tests[705:7792] 21:25:17.699 ERROR: 316: error -66680 2019-02-07 13:25:17.699 apprtcmobile_tests[705:7780] 21:25:17.699 ERROR: >aurioc> 807: failed: -10851 (enable 3, outf< 1 ch, 48000 Hz, Int16> inf< 1 ch, 48000 Hz, Int16>) [001:656] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:238 Initialize): Failed to initialize the Voice Processing I/O unit. Error=-10851. [001:656] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:242 Initialize): Too many initialization attempts. [001:656] [42499] (audio_device_module_ios.mm:485): output: 0 [001:656] [42499] (audio_device_module_ios.mm:517): StartPlayout [001:656] [42499] (audio_device_module_ios.mm:542): Playing [001:656] [42499] (audio_device_buffer.cc:96): StartPlayout [001:656] [42499] (audio_device_ios.mm:226): AudioDeviceIOS::StartPlayout [001:656] [42499] (audio_device_module_ios.mm:524): output: 0 [001:656] [42499] (channel.cc:723): Add remote ssrc: 1694263004 [001:657] [42499] (webrtc_voice_engine.cc:310): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, } [001:657] [42499] (webrtc_voice_engine.cc:331): Always disable AEC on iOS. Use built-in instead. [001:657] [42499] (webrtc_voice_engine.cc:358): Always disable NS on iOS. Use built-in instead. [001:657] [42499] (webrtc_voice_engine.cc:369): Always disable AGC on iOS. Use built-in instead. [001:657] [42499] (audio_device_module_ios.mm:596): BuiltInAECIsAvailable [001:657] [42499] (audio_device_generic.cc:17): virtual bool webrtc::AudioDeviceGeneric::BuiltInAECIsAvailable() const: Not supported on this platform [001:657] [42499] (audio_device_module_ios.mm:599): output: 0 [001:657] [42499] (audio_processing_impl.cc:697): Highpass filter activated: 1 [001:657] [42499] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [001:657] [42499] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [001:657] [42499] (apm_helpers.cc:90): Echo control set to 0 with mode 0 [001:657] [42499] (audio_device_module_ios.mm:612): BuiltInAGCIsAvailable [001:657] [42499] (audio_device_generic.cc:27): virtual bool webrtc::AudioDeviceGeneric::BuiltInAGCIsAvailable() const: Not supported on this platform [001:657] [42499] (audio_device_module_ios.mm:615): output: 0 [001:657] [42499] (apm_helpers.cc:79): AGC set to 0 with mode 2 [001:657] [42499] (audio_device_module_ios.mm:628): BuiltInNSIsAvailable [001:657] [42499] (audio_device_generic.cc:37): virtual bool webrtc::AudioDeviceGeneric::BuiltInNSIsAvailable() const: Not supported on this platform [001:657] [42499] (audio_device_module_ios.mm:631): output: 0 [001:657] [42499] (apm_helpers.cc:104): NS set to 0 [001:658] [42499] (webrtc_voice_engine.cc:471): NetEq capacity is 50 [001:658] [42499] (webrtc_voice_engine.cc:477): NetEq fast mode? 0 [001:658] [42499] (webrtc_voice_engine.cc:483): NetEq minimum delay is 0 [001:658] [42499] (webrtc_voice_engine.cc:489): NetEq handle reordered packets? 0 [001:658] [42499] (webrtc_voice_engine.cc:500): Delay agnostic aec is enabled? 0 [001:658] [42499] (webrtc_voice_engine.cc:510): Extended filter aec is enabled? 0 [001:658] [42499] (webrtc_voice_engine.cc:520): Experimental ns is enabled? 0 [001:658] [42499] (webrtc_voice_engine.cc:536): Typing detection is enabled? 0 [001:658] [42499] (audio_processing_impl.cc:697): Highpass filter activated: 1 [001:658] [42499] (audio_processing_impl.cc:711): Gain Controller 2 activated: 0 [001:658] [42499] (audio_processing_impl.cc:713): Pre-amplifier activated: 0 [001:658] [42499] (audio_device_module_ios.mm:511): RecordingIsInitialized [001:658] [42499] (audio_device_module_ios.mm:573): Recording [001:658] [42499] (audio_device_module_ios.mm:492): InitRecording [001:658] [42499] (audio_device_module_ios.mm:511): RecordingIsInitialized [001:658] [42499] (audio_device_module_ios.mm:548): StartRecording [001:658] [42499] (audio_device_module_ios.mm:573): Recording [001:658] [42499] (audio_device_buffer.cc:115): StartRecording [001:658] [42499] (audio_device_ios.mm:279): AudioDeviceIOS::StartRecording [001:658] [42499] (audio_device_module_ios.mm:555): output: 0 [001:658] [42499] (channel.cc:867): Changing voice state, recv=1 send=1 [001:659] [42499] (message_queue.cc:518): Message took 433ms to dispatch. Posted from: SetRemoteContent@../../pc/channel.cc:307 [001:659] [42499] (channel.cc:1065): Setting remote video description [001:659] [42499] (webrtc_video_engine.cc:663): SetSendParameters: {codecs: [VideoCodec[96:H264], VideoCodec[97:rtx], VideoCodec[98:H264], VideoCodec[99:rtx], VideoCodec[100:VP8], VideoCodec[101:rtx], VideoCodec[127:VP9], VideoCodec[123:rtx], VideoCodec[125:red], VideoCodec[122:rtx], VideoCodec[124:ulpfec]], conference_mode: no, extensions: [{uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 10}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 12}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}], extmap-allow-mixed: false, max_bandwidth_bps: -1, mid: 1} [001:659] [42499] (webrtc_video_engine.cc:672): Using codec: VideoCodec[96:H264] [001:660] [42499] (webrtc_video_engine.cc:1744): RecreateWebRtcStream (send) because of SetCodec. [001:660] [42499] (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 [001:660] [12583] (rtp_video_sender.cc:463): Transmitting payload type without picture ID using NACK+ULPFEC is a waste of bandwidth since ULPFEC packets also have to be retransmitted. Disabling ULPFEC. [001:660] [12583] (video_send_stream_impl.cc:250): VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [362193604], 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: 10}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 12}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 124, red_payload_type: 125, red_rtx_payload_type: 122}, payload_name: H264, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [3849333516], payload_type: 97}, c_name: gz57yMrR3CAwNWnn}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [001:660] [12583] (video_send_stream_impl.cc:276): ERROR: Initial encoder max bitrate = -1 which is <= 0! [001:660] [12583] (video_send_stream_impl.cc:278): Using default encoder max bitrate = 10 Mbps [001:661] [42499] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [001:661] [12583] (video_stream_encoder.cc:502): ConfigureEncoder requested. [001:661] [42499] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [001:661] [42499] (video_send_stream.cc:159): VideoSendStream::Stop [001:661] [42499] (webrtc_video_engine.cc:724): SetFeedbackOptions on all the receive streams because the send codec or RTCP mode has changed. [001:661] [8711] (video_send_stream_impl.cc:423): VideoSendStream::Stop [001:661] [42499] (webrtc_video_engine.cc:1145): AddRecvStream: {id:ARDAMSv0;ssrcs:[115403994,848457161];ssrc_groups:{semantics:FID;ssrcs:[115403994,848457161]};cname:TpAYwgBWogJE85ER;stream_ids:ARDAMS;} [001:662] [42499] (video_receive_stream.cc:203): VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 127, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 115403994, local_ssrc: 362193604, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 124, red_type: 125, rtx_ssrc: 848457161, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 122 (pt) -> 125 (apt), 123 (pt) -> 127 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 10}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 12}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0} [001:663] [42499] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [001:663] [42499] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [001:663] [42499] (channel.cc:723): Add remote ssrc: 115403994 [001:663] [42499] (video_send_stream.cc:132): VideoSendStream::UpdateActiveSimulcastLayers [001:663] [12583] (video_send_stream_impl.cc:364): VideoSendStream::UpdateActiveSimulcastLayers [001:663] [12583] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 10000000bps [001:663] [42499] (channel.cc:1000): Changing video state, send=1 [001:664] [42499] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=up [001:664] [42499] (rtp_transport_controller_send.cc:266): SignalNetworkState Up [001:664] [42243] (peer_connection.cc:3911): Changing IceConnectionState 0 => 1 [001:664] [42499] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=up [001:664] [42499] (rtp_transport_controller_send.cc:266): SignalNetworkState Up [001:664] [8711] (rtp_transport_controller_send.cc:477): Creating fallback congestion controller (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 1 [001:664] [42243] (peer_connection.cc:5303): Non-rejected SCTP m= section is needed to get the SSL Role of the SCTP transport. [001:664] [42243] (peer_connection.cc:2593): Processing the MSIDs for MID=0 (streams=[ARDAMS]). [001:664] [42499] (channel.cc:867): Changing voice state, recv=1 send=1 [001:664] [8711] (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 [001:664] [42243] (peer_connection.cc:2604): Processing the addition of a remote track for MID=0. [001:664] [42499] (video_send_stream.cc:132): VideoSendStream::UpdateActiveSimulcastLayers [001:664] [8711] (aimd_rate_control.cc:101): Using aimd rate control with back off factor 0.85 [001:664] [42243] (rtp_transceiver.cc:169): Changing transceiver (MID=0) current direction from <not set> to kSendRecv. [001:664] [8711] (delay_based_bwe.cc:104): Using Trendline filter for delay change estimation with window size 20 [001:664] [8711] (delay_based_bwe.cc:290): BWE Setting start bitrate to: 300 kbps [001:664] [8711] (probe_controller.cc:210): Measured bitrate: 300000 Minimum to probe further: 1260000 [001:664] [8711] (bitrate_prober.cc:111): Probe cluster (bitrate:min bytes:min packets): (900000:1687:5) [001:664] [8711] (bitrate_prober.cc:111): Probe cluster (bitrate:min bytes:min packets): (1800000:3375:5) [001:664] [8711] (bitrate_allocator.cc:115): Current BWE 300000 [001:665] [12583] (video_stream_encoder.cc:1151): Video suspend state changed to: not suspended [001:665] [8711] (video_send_stream_impl.cc:364): VideoSendStream::UpdateActiveSimulcastLayers [001:665] [42499] (channel.cc:1000): Changing video state, send=1 [001:665] [42499] (webrtc_voice_engine.cc:1953): SetOutputVolume() to 1 for recv stream with ssrc 1694263004 [001:665] [42243] (peer_connection.cc:2593): Processing the MSIDs for MID=1 (streams=[ARDAMS]). [001:665] [42243] (peer_connection.cc:2604): Processing the addition of a remote track for MID=1. [001:665] [42243] (rtp_transceiver.cc:169): Changing transceiver (MID=1) current direction from <not set> to kSendRecv. [001:665] [42499] (webrtc_video_engine.cc:1275): SetSink: ssrc:115403994 (ptr) [001:665] [42243] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x7faf346c6490): created DTMF sender: RTCDtmfSender { remainingTones: duration: 0.100000 sec interToneGap: 0.050000 sec } [001:665] [42243] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf346c5f20): created sender: RTCRtpSender { senderId: ARDAMSa0 } [001:665] [42243] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x7faf346c7f10): created receiver: RTCRtpReceiver { receiverId: b213d12b-dc73-4a90-ae02-3e5edac76902 } [001:665] [42243] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x7faf346c6590): created transceiver: RTCRtpTransceiver { sender: RTCRtpSender { senderId: ARDAMSa0 } receiver: RTCRtpReceiver { receiverId: b213d12b-dc73-4a90-ae02-3e5edac76902 } } (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving audio on track b213d12b-dc73-4a90-ae02-3e5edac76902. [001:666] [42243] (rtp_sender.cc:561): Tried to get DTMF sender from video sender. [001:666] [42243] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf346ca4d0): created sender: RTCRtpSender { senderId: ARDAMSv0 } [001:666] [42243] (RTCLogging.mm:33): (RTCRtpReceiver.mm:120 -[RTCRtpReceiver initWithFactory:nativeRtpReceiver:]): RTCRtpReceiver(0x7faf346cb560): created receiver: RTCRtpReceiver { receiverId: c8decf54-dee3-48ff-ba49-4c0fad656b99 } [001:666] [42243] (RTCLogging.mm:33): (RTCRtpTransceiver.mm:136 -[RTCRtpTransceiver initWithFactory:nativeRtpTransceiver:]): RTCRtpTransceiver(0x7faf346c6590): created transceiver: RTCRtpTransceiver { sender: RTCRtpSender { senderId: ARDAMSv0 } receiver: RTCRtpReceiver { receiverId: c8decf54-dee3-48ff-ba49-4c0fad656b99 } } (RTCLogging.mm:33): (ARDAppClient.m:387 -[ARDAppClient peerConnection:didStartReceivingOnTransceiver:]): Now receiving video on track c8decf54-dee3-48ff-ba49-4c0fad656b99. (RTCLogging.mm:33): (ARDAppClient.m:381 -[ARDAppClient peerConnection:didAddStream:]): Stream with 1 video tracks and 1 audio tracks was added. [001:666] [42243] (message_queue.cc:518): Message took 447ms to dispatch. Posted from: SetRemoteDescription@../../api/peer_connection_proxy.h:104 (RTCLogging.mm:33): (ARDAppClient.m:409 -[ARDAppClient peerConnection:didChangeConnectionState:]): ICE+DTLS state changed: 2 [001:666] [42243] (peer_connection.cc:5856): Changing to ICE completed state because all transports are complete. [001:666] [42243] (peer_connection.cc:3911): Changing IceConnectionState 1 => 2 (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 2 [001:666] [42243] (peer_connection.cc:3911): Changing IceConnectionState 2 => 3 (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 3 [001:667] [771] (RTCLogging.mm:33): (RTCDtmfSender.mm:70 -[RTCDtmfSender initWithNativeDtmfSender:]): RTCDtmfSender(0x7faf3712a4b0): created DTMF sender: RTCDtmfSender { remainingTones: duration: 0.100000 sec interToneGap: 0.050000 sec } [001:667] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf3712a490): created sender: RTCRtpSender { senderId: ARDAMSa0 } [001:667] [41987] (rtp_sender.cc:561): Tried to get DTMF sender from video sender. [001:667] [771] (RTCLogging.mm:33): (RTCRtpSender.mm:107 -[RTCRtpSender initWithFactory:nativeRtpSender:]): RTCRtpSender(0x7faf346cb110): created sender: RTCRtpSender { senderId: ARDAMSv0 } [001:669] [22279] (jsep_transport_controller.cc:319): Not adding candidate because the JsepTransport doesn't exist. Ignore it. [001:672] [22787] (rtc_event_log_impl.cc:210): Stopping WebRTC event log. [001:672] [22787] (rtc_event_log_impl.cc:227): WebRTC event log successfully stopped. [001:674] [22787] (webrtc_video_engine.cc:2087): VideoSendStream stats: 129145, {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: 115403994, width: 0, height: 0, key: 0, delta: 0, total_bps: 0, retransmit_bps: 0, avg_delay_ms: 0, max_delay_ms: 0, cum_loss: 0, max_ext_seq: 0, nack: 0, fir: 0, pli: 0} [001:674] [22787] (webrtc_video_engine.cc:2536): VideoReceiveStream stats: 129145, {ssrc: 362193604, 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} [001:674] [22787] (webrtc_video_engine.cc:1322): Call stats: 129145, {send_bw_bps: 300000, recv_bw_bps: 0, max_pad_bps: 0, pacer_delay_ms: 0, rtt_ms: -1} [001:675] [41987] (peer_connection.cc:3981): Session: 8089159972302902786 Old state: kStable New state: kClosed [001:675] [22787] (audio_device_module_ios.mm:562): StopRecording [001:675] [22787] (audio_device_ios.mm:299): AudioDeviceIOS::StopRecording [001:675] [22787] (audio_device_buffer.cc:151): StopRecording [001:675] [22787] (audio_device_buffer.cc:174): total recording time: 422 [001:675] [22787] (audio_device_module_ios.mm:566): output: 0 [001:675] [22787] (webrtc_voice_engine.cc:1953): SetOutputVolume() to 0 for recv stream with ssrc 1056030614 [001:675] [22787] (webrtc_video_engine.cc:1003): SetVideoSend (ssrc= 115403994, options: nullptr, source = nullptr) [001:676] [22787] (webrtc_video_engine.cc:1275): SetSink: ssrc:362193604 nullptr [001:676] [22787] (channel.cc:567): Channel disabled [001:676] [22787] (video_send_stream.cc:159): VideoSendStream::Stop [001:676] [22787] (channel.cc:1000): Changing video state, send=0 [001:676] [8711] (video_send_stream_impl.cc:423): VideoSendStream::Stop [001:676] [8711] (rtp_rtcp_impl.cc:398): Failed to send RTCP BYE [001:676] [8711] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps [001:676] [22787] (video_send_stream.cc:159): VideoSendStream::Stop [001:676] [8711] (video_send_stream_impl.cc:423): VideoSendStream::Stop [001:676] [12583] (video_stream_encoder.cc:1151): Video suspend state changed to: suspended [001:676] [12583] (video_send_stream_impl.cc:423): VideoSendStream::Stop [001:676] [12583] (video_send_stream_impl.cc:338): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [115403994], 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: 10}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 12}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 124, red_payload_type: 125, red_rtx_payload_type: 122}, payload_name: H264, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [848457161], payload_type: 97}, c_name: TpAYwgBWogJE85ER}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [001:677] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=up [001:677] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Up [001:678] [22787] (send_statistics_proxy.cc:654): Frames encoded 0 WebRTC.Video.DroppedFrames.Capturer 0 WebRTC.Video.DroppedFrames.EncoderQueue 0 WebRTC.Video.DroppedFrames.Encoder 0 WebRTC.Video.DroppedFrames.Ratelimiter 0 [001:678] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=up [001:678] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Up [001:678] [22787] (video_receive_stream.cc:252): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 127, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 362193604, local_ssrc: 115403994, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 124, red_type: 125, rtx_ssrc: 3849333516, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 122 (pt) -> 125 (apt), 123 (pt) -> 127 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 10}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 12}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0} [001:678] [22787] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [001:679] [22787] (receive_statistics_proxy.cc:487): Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [001:679] [22787] (channel.cc:167): Destroyed channel: 1 [001:679] [22787] (channel.cc:567): Channel disabled [001:679] [22787] (audio_device_module_ios.mm:531): StopPlayout [001:679] [22787] (audio_device_ios.mm:248): AudioDeviceIOS::StopPlayout [001:679] [22787] (audio_device_ios.mm:889): AudioDeviceIOS::ShutdownPlayOrRecord [001:679] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:331 Stop): Stopping audio unit. [001:679] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:338 Stop): Stopped audio unit [001:679] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:347 Uninitialize): Unintializing audio unit. [001:679] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:354 Uninitialize): Uninitialized audio unit. [001:679] [22787] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:459 DisposeAudioUnit): Disposing audio unit. [001:680] [22787] (RTCLogging.mm:33): (RTCAudioSession.mm:196 -[RTCAudioSession removeDelegate:]): Removing delegate: (0x7faf372260d0) [001:680] [22787] (RTCLogging.mm:33): (audio_device_ios.mm:830 UnconfigureAudioSession): Unconfiguring audio session. [001:680] [22787] (RTCLogging.mm:33): (RTCAudioSession.mm:762 -[RTCAudioSession unconfigureWebRTCSession:]): Unconfiguring audio session for WebRTC. [001:681] [22787] (RTCLogging.mm:33): (RTCAudioSession.mm:629 -[RTCAudioSession decrementActivationCount]): Decrementing activation count. [001:681] [22787] (RTCLogging.mm:33): (RTCAudioSession.mm:377 -[RTCAudioSession setActive:error:]): Number of current activations: 1 [001:681] [22787] (RTCLogging.mm:33): (audio_device_ios.mm:841 UnconfigureAudioSession): Unconfigured audio session. [001:681] [22787] (RTCLogging.mm:33): (audio_device_ios.mm:270 StopPlayout): Average number of playout callbacks between glitches: 100000 [001:681] [22787] (audio_device_buffer.cc:137): StopPlayout [001:681] [22787] (audio_device_buffer.cc:143): total playout time: 482 [001:681] [22787] (audio_device_module_ios.mm:535): output: 0 [001:681] [22787] (channel.cc:867): Changing voice state, recv=0 send=0 [001:681] [22787] (webrtc_voice_engine.cc:1814): RemoveSendStream: 1694263004 [001:681] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=up [001:681] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Up [001:681] [22787] (audio_send_stream.cc:180): ~AudioSendStream: 1694263004 [001:681] [22787] (webrtc_voice_engine.cc:1891): RemoveRecvStream: 1056030614 [001:681] [22787] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [001:681] [22787] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [001:681] [22787] (audio_receive_stream.cc:135): ~AudioReceiveStream: 1056030614 [001:681] [8711] (paced_sender.cc:94): PacedSender paused. [001:681] [8711] (control_handler.cc:97): Bitrate estimate state changed, BWE: 300 kbps. [001:682] [22787] (channel.cc:167): Destroyed channel: 0 [001:682] [22279] (openssl_stream_adapter.cc:902): Cleanup [001:682] [22279] (openssl_adapter.cc:787): SSL3 alert write:warning:close notify [001:682] [22531] (openssl_adapter.cc:787): SSL3 alert read:warning:close notify [001:682] [22531] (openssl_stream_adapter.cc:902): Cleanup [001:682] [22531] (openssl_adapter.cc:787): SSL3 alert write:warning:close notify [001:682] [22531] (dtls_transport.cc:651): DtlsTransport[0|1|_W]: DTLS transport closed [001:682] [22531] (jsep_transport_controller.cc:1234): Transport 0 writability changed to 0. [001:682] [42243] (peer_connection.cc:3911): Changing IceConnectionState 3 => 5 (RTCLogging.mm:33): (ARDAppClient.m:401 -[ARDAppClient peerConnection:didChangeIceConnectionState:]): ICE state changed: 5 [001:682] [22531] (channel.cc:600): Channel not writable (0) [001:682] [22531] (channel.cc:600): Channel not writable (1) [001:682] [22531] (srtp_transport.cc:363): The params in SRTP transport are reset. [001:683] [42499] (channel.cc:867): Changing voice state, recv=1 send=1 [001:683] [22787] (paced_sender.cc:399): ProcessThreadAttached 0x0 [001:683] [22787] (paced_sender.cc:399): ProcessThreadAttached 0x0 [001:683] [42499] (video_send_stream.cc:132): VideoSendStream::UpdateActiveSimulcastLayers [001:683] [8711] (video_send_stream_impl.cc:364): VideoSendStream::UpdateActiveSimulcastLayers [001:683] [42499] (channel.cc:1000): Changing video state, send=1 [001:683] [22787] (rtc_event_log_impl.cc:210): Stopping WebRTC event log. [001:683] [22787] (rtc_event_log_impl.cc:227): WebRTC event log successfully stopped. [001:683] [41987] (peer_connection.cc:6577): Usage signature is 4088 [001:684] [41987] (peer_connection.cc:860): Session: 8089159972302902786 is destroyed. [001:684] [41987] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue. [001:684] [22787] (webrtc_video_engine.cc:451): WebRtcVideoEngine::~WebRtcVideoEngine [001:684] [22787] (webrtc_voice_engine.cc:203): WebRtcVoiceEngine::~WebRtcVoiceEngine [001:684] [22787] (audio_device_module_ios.mm:531): StopPlayout [001:684] [22787] (audio_device_ios.mm:248): AudioDeviceIOS::StopPlayout [001:685] [22787] (audio_device_module_ios.mm:535): output: 0 [001:685] [22787] (audio_device_module_ios.mm:562): StopRecording [001:685] [22787] (audio_device_ios.mm:299): AudioDeviceIOS::StopRecording [001:685] [22787] (audio_device_module_ios.mm:566): output: 0 [001:685] [22787] (audio_device_module_ios.mm:580): RegisterAudioCallback [001:685] [22787] (audio_device_buffer.cc:79): RegisterAudioCallback [001:685] [22787] (audio_device_module_ios.mm:91): Terminate [001:685] [22787] (audio_device_ios.mm:167): AudioDeviceIOS::Terminate [001:685] [22787] (audio_device_ios.mm:248): AudioDeviceIOS::StopPlayout [001:685] [22787] (audio_device_ios.mm:299): AudioDeviceIOS::StopRecording [001:686] [22787] (audio_device_module_ios.mm:54): ~AudioDeviceModuleIOS [001:686] [22787] (audio_device_buffer.cc:73): AudioDeviceBuffer::~dtor [001:686] [22787] (audio_device_ios.mm:126): AudioDeviceIOS::~dtor<NSThread: 0x7faf3720afa0>{number = 4, name = (null)} [001:686] [22787] (audio_device_ios.mm:167): AudioDeviceIOS::Terminate [001:686] [22787] (fine_audio_buffer.cc:46): ~FineAudioBuffer Test Case '-[ARDAppClientTest testSession]' passed (1.707 seconds). Test Suite 'ARDAppClientTest' passed at 2019-02-07 21:25:17.732. Executed 1 test, with 0 failures (0 unexpected) in 1.707 (1.708) seconds Test Suite 'ARDFileCaptureControllerTests' started at 2019-02-07 21:25:17.732 Test Case '-[ARDFileCaptureControllerTests testCaptureIsStarted]' started. Test Case '-[ARDFileCaptureControllerTests testCaptureIsStarted]' passed (0.001 seconds). Test Case '-[ARDFileCaptureControllerTests testCaptureIsStoped]' started. Test Case '-[ARDFileCaptureControllerTests testCaptureIsStoped]' passed (0.001 seconds). Test Suite 'ARDFileCaptureControllerTests' passed at 2019-02-07 21:25:17.736. Executed 2 tests, with 0 failures (0 unexpected) in 0.002 (0.004) seconds Test Suite 'ARDSettingsModelTests' started at 2019-02-07 21:25:17.737 Test Case '-[ARDSettingsModelTests testConstraintComponentIsNilWhenInvalidConstraintString]' started. Test Case '-[ARDSettingsModelTests testConstraintComponentIsNilWhenInvalidConstraintString]' passed (0.002 seconds). Test Case '-[ARDSettingsModelTests testHeightConstraintFromStore]' started. Test Case '-[ARDSettingsModelTests testHeightConstraintFromStore]' passed (0.002 seconds). Test Case '-[ARDSettingsModelTests testRetrievingSetting]' started. Test Case '-[ARDSettingsModelTests testRetrievingSetting]' passed (0.001 seconds). Test Case '-[ARDSettingsModelTests testReturningDefaultCallOption]' started. Test Case '-[ARDSettingsModelTests testReturningDefaultCallOption]' passed (0.001 seconds). Test Case '-[ARDSettingsModelTests testStoringAudioSetting]' started. Test Case '-[ARDSettingsModelTests testStoringAudioSetting]' passed (0.001 seconds). Test Case '-[ARDSettingsModelTests testStoringInvalidConstraintReturnsNo]' started. Test Case '-[ARDSettingsModelTests testStoringInvalidConstraintReturnsNo]' passed (0.001 seconds). Test Case '-[ARDSettingsModelTests testWidthConstraintFromStore]' started. Test Case '-[ARDSettingsModelTests testWidthConstraintFromStore]' passed (0.002 seconds). Test Suite 'ARDSettingsModelTests' passed at 2019-02-07 21:25:17.750. Executed 7 tests, with 0 failures (0 unexpected) in 0.010 (0.013) seconds Test Suite 'apprtcmobile_tests.app' passed at 2019-02-07 21:25:17.751. Executed 10 tests, with 0 failures (0 unexpected) in 1.719 (1.728) seconds Test Suite 'All tests' passed at 2019-02-07 21:25:17.752. Executed 10 tests, with 0 failures (0 unexpected) in 1.719 (1.730) seconds [001:709] [42499] (rtc_event_log_impl.cc:210): Stopping WebRTC event log. [001:710] [42499] (rtc_event_log_impl.cc:227): WebRTC event log successfully stopped. [001:712] [42243] (peer_connection.cc:3981): Session: 1660230460349573763 Old state: kStable New state: kClosed [001:712] [42499] (audio_device_module_ios.mm:562): StopRecording [001:712] [42499] (audio_device_ios.mm:299): AudioDeviceIOS::StopRecording [001:712] [42499] (audio_device_buffer.cc:151): StopRecording [001:712] [42499] (audio_device_buffer.cc:174): total recording time: 54 [001:712] [42499] (audio_device_module_ios.mm:566): output: 0 [001:712] [42499] (webrtc_voice_engine.cc:1953): SetOutputVolume() to 0 for recv stream with ssrc 1694263004 [001:712] [42499] (webrtc_video_engine.cc:1003): SetVideoSend (ssrc= 362193604, options: nullptr, source = nullptr) [001:713] [42499] (webrtc_video_engine.cc:1275): SetSink: ssrc:115403994 nullptr [001:713] [42499] (channel.cc:567): Channel disabled [001:713] [42499] (video_send_stream.cc:159): VideoSendStream::Stop [001:713] [42499] (channel.cc:1000): Changing video state, send=0 [001:713] [17411] (video_send_stream_impl.cc:423): VideoSendStream::Stop [001:713] [17411] (rtp_rtcp_impl.cc:398): Failed to send RTCP BYE [001:713] [17411] (bitrate_allocator.cc:245): UpdateAllocationLimits : total_requested_min_bitrate: 0bps, total_requested_padding_bitrate: 0bps, total_requested_max_bitrate: 0bps [001:713] [17411] (video_stream_encoder.cc:1151): Video suspend state changed to: suspended [001:713] [42499] (video_send_stream.cc:159): VideoSendStream::Stop [001:713] [17411] (video_send_stream_impl.cc:423): VideoSendStream::Stop [001:713] [12583] (video_send_stream_impl.cc:423): VideoSendStream::Stop [001:713] [12583] (video_send_stream_impl.cc:338): ~VideoSendStreamInternal: {encoder_settings: { experiment_cpu_load_estimator: off}}, rtp: {ssrcs: [362193604], 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: 10}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 12}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}], nack: {rtp_history_ms: 1000}, ulpfec: {ulpfec_payload_type: 124, red_payload_type: 125, red_rtx_payload_type: 122}, payload_name: H264, payload_type: 96, flexfec: {payload_type: -1, ssrc: 0, protected_media_ssrcs: []}, rtx: {ssrcs: [3849333516], payload_type: 97}, c_name: gz57yMrR3CAwNWnn}, rtcp_report_interval_ms: 1000, send_transport: (Transport), media_transport: nullptr, render_delay_ms: 0, target_delay_ms: 0, suspend_below_min_bitrate: off} [001:714] [42499] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=up [001:714] [42499] (rtp_transport_controller_send.cc:266): SignalNetworkState Up [001:714] [42499] (send_statistics_proxy.cc:654): Frames encoded 0 WebRTC.Video.DroppedFrames.Capturer 0 WebRTC.Video.DroppedFrames.EncoderQueue 0 WebRTC.Video.DroppedFrames.Encoder 0 WebRTC.Video.DroppedFrames.Ratelimiter 0 [001:714] [42499] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=up [001:714] [42499] (rtp_transport_controller_send.cc:266): SignalNetworkState Up [001:714] [42499] (video_receive_stream.cc:252): ~VideoReceiveStream: {decoders: [{payload_type: 96, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 640c1f}}, {payload_type: 98, payload_name: H264, codec_params: {level-asymmetry-allowed: 1packetization-mode: 1profile-level-id: 42e01f}}, {payload_type: 100, payload_name: VP8, codec_params: {}}, {payload_type: 127, payload_name: VP9, codec_params: {}}], rtp: {remote_ssrc: 115403994, local_ssrc: 362193604, rtcp_mode: RtcpMode::kReducedSize, rtcp_xr: {receiver_reference_time_report: off}, remb: on, transport_cc: on, nack: {rtp_history_ms: 1000}, ulpfec_payload_type: 124, red_type: 125, rtx_ssrc: 848457161, rtx_payload_types: {97 (pt) -> 96 (apt), 99 (pt) -> 98 (apt), 101 (pt) -> 100 (apt), 122 (pt) -> 125 (apt), 123 (pt) -> 127 (apt), }, extensions: [{uri: http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07, id: 10}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 5}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/color-space, id: 12}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/playout-delay, id: 6}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-content-type, id: 7}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/video-timing, id: 8}, {uri: urn:3gpp:video-orientation, id: 4}, {uri: urn:ietf:params:rtp-hdrext:sdes:mid, id: 9}, {uri: urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id, id: 14}, {uri: urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id, id: 13}, {uri: urn:ietf:params:rtp-hdrext:toffset, id: 2}]}, renderer: (renderer), render_delay_ms: 10, sync_group: ARDAMS, target_delay_ms: 0} [001:715] [42499] (video_render_frames.cc:47): WebRTC.Video.DroppedFrames.RenderQueue 0 [001:715] [42499] (receive_statistics_proxy.cc:487): Frames decoded 0 WebRTC.Video.DroppedFrames.Receiver 0 [001:715] [42499] (channel.cc:167): Destroyed channel: 1 [001:715] [42499] (channel.cc:567): Channel disabled [001:715] [42499] (audio_device_module_ios.mm:531): StopPlayout [001:715] [42499] (audio_device_ios.mm:248): AudioDeviceIOS::StopPlayout [001:715] [42499] (audio_device_ios.mm:889): AudioDeviceIOS::ShutdownPlayOrRecord [001:715] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:331 Stop): Stopping audio unit. [001:715] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:338 Stop): Stopped audio unit [001:715] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:347 Uninitialize): Unintializing audio unit. [001:715] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:354 Uninitialize): Uninitialized audio unit. [001:715] [42499] (RTCLogging.mm:33): (voice_processing_audio_unit.mm:459 DisposeAudioUnit): Disposing audio unit. [001:716] [42499] (RTCLogging.mm:33): (RTCAudioSession.mm:196 -[RTCAudioSession removeDelegate:]): Removing delegate: (0x7faf37043690) [001:716] [42499] (RTCLogging.mm:33): (audio_device_ios.mm:830 UnconfigureAudioSession): Unconfiguring audio session. [001:716] [42499] (RTCLogging.mm:33): (RTCAudioSession.mm:762 -[RTCAudioSession unconfigureWebRTCSession:]): Unconfiguring audio session for WebRTC. [001:716] [42499] (RTCLogging.mm:33): (RTCAudioSession.mm:629 -[RTCAudioSession decrementActivationCount]): Decrementing activation count. [001:716] [42499] (RTCLogging.mm:33): (RTCAudioSession.mm:377 -[RTCAudioSession setActive:error:]): Number of current activations: 0 [001:716] [42499] (RTCLogging.mm:33): (audio_device_ios.mm:841 UnconfigureAudioSession): Unconfigured audio session. [001:716] [42499] (RTCLogging.mm:33): (audio_device_ios.mm:270 StopPlayout): Average number of playout callbacks between glitches: 100000 [001:716] [42499] (audio_device_buffer.cc:137): StopPlayout [001:716] [42499] (audio_device_buffer.cc:143): total playout time: 60 [001:716] [42499] (audio_device_module_ios.mm:535): output: 0 [001:716] [42499] (channel.cc:867): Changing voice state, recv=0 send=0 [001:716] [42499] (webrtc_voice_engine.cc:1814): RemoveSendStream: 1056030614 [001:716] [42499] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=up [001:716] [42499] (rtp_transport_controller_send.cc:266): SignalNetworkState Up [001:716] [42499] (audio_send_stream.cc:180): ~AudioSendStream: 1056030614 [001:716] [42499] (webrtc_voice_engine.cc:1891): RemoveRecvStream: 1694263004 [001:716] [42499] (call.cc:1095): UpdateAggregateNetworkState: aggregate_state=down [001:716] [42499] (rtp_transport_controller_send.cc:266): SignalNetworkState Down [001:716] [42499] (audio_receive_stream.cc:135): ~AudioReceiveStream: 1694263004 [001:716] [17411] (paced_sender.cc:94): PacedSender paused. [001:716] [17411] (control_handler.cc:97): Bitrate estimate state changed, BWE: 300 kbps. [001:717] [42499] (channel.cc:167): Destroyed channel: 0 [001:717] [22531] (openssl_stream_adapter.cc:902): Cleanup [001:717] [42499] (paced_sender.cc:399): ProcessThreadAttached 0x0 [001:717] [42499] (paced_sender.cc:399): ProcessThreadAttached 0x0 [001:717] [42499] (rtc_event_log_impl.cc:210): Stopping WebRTC event log. [001:718] [42499] (rtc_event_log_impl.cc:227): WebRTC event log successfully stopped. [001:718] [42243] (peer_connection.cc:6577): Usage signature is 4088 [001:718] [42243] (peer_connection.cc:860): Session: 1660230460349573763 is destroyed. [001:718] [42243] (dtmf_sender.cc:216): The Dtmf provider is deleted. Clear the sending queue. [001:719] [42499] (webrtc_video_engine.cc:451): WebRtcVideoEngine::~WebRtcVideoEngine [001:719] [42499] (webrtc_voice_engine.cc:203): WebRtcVoiceEngine::~WebRtcVoiceEngine [001:719] [42499] (audio_device_module_ios.mm:531): StopPlayout [001:719] [42499] (audio_device_ios.mm:248): AudioDeviceIOS::StopPlayout [001:719] [42499] (audio_device_module_ios.mm:535): output: 0 [001:719] [42499] (audio_device_module_ios.mm:562): StopRecording [001:719] [42499] (audio_device_ios.mm:299): AudioDeviceIOS::StopRecording [001:719] [42499] (audio_device_module_ios.mm:566): output: 0 [001:719] [42499] (audio_device_module_ios.mm:580): RegisterAudioCallback [001:719] [42499] (audio_device_buffer.cc:79): RegisterAudioCallback [001:719] [42499] (audio_device_module_ios.mm:91): Terminate [001:719] [42499] (audio_device_ios.mm:167): AudioDeviceIOS::Terminate [001:719] [42499] (audio_device_ios.mm:248): AudioDeviceIOS::StopPlayout [001:719] [42499] (audio_device_ios.mm:299): AudioDeviceIOS::StopRecording [001:720] [42499] (audio_device_module_ios.mm:54): ~AudioDeviceModuleIOS [001:720] [42499] (audio_device_buffer.cc:73): AudioDeviceBuffer::~dtor [001:720] [42499] (audio_device_ios.mm:126): AudioDeviceIOS::~dtor<NSThread: 0x7faf37449940>{number = 3, name = (null)} [001:720] [42499] (audio_device_ios.mm:167): AudioDeviceIOS::Terminate [001:721] [42499] (fine_audio_buffer.cc:46): ~FineAudioBuffer Test session results and logs: /Users/chrome-bot/Library/Developer/Xcode/DerivedData/temporary-bontgrowxwyfgnajrbpubqfcilne/Logs/Test/Test-Transient Testing-2019.02.07_13-24-54--0800.xcresult 2019-02-07 13:25:18.097 xcodebuild[603:6361] [MT] IDETestOperationsObserverDebug: 23.784 elapsed -- Testing started completed. 2019-02-07 13:25:18.097 xcodebuild[603:6361] [MT] IDETestOperationsObserverDebug: 0.000 sec, +0.000 sec -- start 2019-02-07 13:25:18.097 xcodebuild[603:6361] [MT] IDETestOperationsObserverDebug: 23.784 sec, +23.784 sec -- end ** TEST EXECUTE SUCCEEDED ** Waiting for test process to terminate. Test process terminated. Device wiped. /b/s/w/ir/src/out/Debug-iphonesimulator/iossim returned 0 2019-02-07 13:26:45,363 - root: [DEBUG] Contents of task_output_dir: ['/b/s/w/ir/kitchen-workdir/recipe_cleanup/0w23_17/42e2898861104610'] 2019-02-07 13:26:45,364 - root: [WARNING] Expected output.json file missing: set(['/b/s/w/ir/kitchen-workdir/recipe_cleanup/0w23_17/42e2898861104610/output.json']) Found: [] Expected: ['/b/s/w/ir/kitchen-workdir/recipe_cleanup/0w23_17/42e2898861104610/output.json'] 2019-02-07 13:26:45,364 - root: [WARNING] No shard json files found in task_output_dir: '/b/s/w/ir/kitchen-workdir/recipe_cleanup/0w23_17' Found ['/b/s/w/ir/kitchen-workdir/recipe_cleanup/0w23_17/42e2898861104610'] 2019-02-07 13:26:45,364 - root: [DEBUG] Found shard_json_files: ['/b/s/w/ir/kitchen-workdir/recipe_cleanup/0w23_17/42e2898861104610/output.json'] 2019-02-07 13:26:45,364 - root: [INFO] merge_cmd: /b/s/w/ir/cache/vpython/5b0713/bin/python /b/s/w/ir/kitchen-checkout/build/scripts/slave/recipe_modules/swarming/resources/noop_merge.py --summary-json /b/s/w/ir/tmp/t/tmpZ8uMv8.json --task-output-dir /b/s/w/ir/kitchen-workdir/recipe_cleanup/0w23_17 -o /b/s/w/ir/tmp/t/tmpLoxgp9.json Running ['/b/s/w/ir/cache/vpython/5b0713/bin/python', '/b/s/w/ir/kitchen-checkout/build/scripts/slave/recipe_modules/swarming/resources/noop_merge.py', '--summary-json', '/b/s/w/ir/tmp/t/tmpZ8uMv8.json', '--task-output-dir', '/b/s/w/ir/kitchen-workdir/recipe_cleanup/0w23_17', '-o', '/b/s/w/ir/tmp/t/tmpLoxgp9.json'] in None (env: None) Command ['/b/s/w/ir/cache/vpython/5b0713/bin/python', '/b/s/w/ir/kitchen-checkout/build/scripts/slave/recipe_modules/swarming/resources/noop_merge.py', '--summary-json', '/b/s/w/ir/tmp/t/tmpZ8uMv8.json', '--task-output-dir', '/b/s/w/ir/kitchen-workdir/recipe_cleanup/0w23_17', '-o', '/b/s/w/ir/tmp/t/tmpLoxgp9.json'] returned exit code 0