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Good day. Yes, WebRTC specification strongly requires data encryption. That's why WebRTC signaling works only via secure websocket connection (except localhost, for debugging purposes) RTSP supports basic authentication only, so yes, credentials should be passed in URL. But you can hide...

Good day. That's the problem. You should set an actual RTMP publisher version rtmp_flash_ver_subscriber=LNX 76.219.189.0 disable RTMP stream name prefix rtmp_transponder_stream_name_prefix= and use full RTMP URL rtmp_transponder_full_url=true A final set of server parameters that should work...

Good day. Please clarify: do you have the only entry node per CDN (node A)? In this case, when node A is restarted, you should either restart other nodes to enter CDN again, or try to reset all CDN connections on all the nodes using REST API: Connection reset using REST API This is default...

The issue was fixed in WebSDK build 2.0.217. Please update @flashphoner/websdk package and check.

This should work in any environment. But all the filesystems are virtual in docker, so any disk I/O operations may be slower, and this is impredictable. In docker environment, we recoddems to inscrease a maximum snapshot taking interval, for example twice snapshot_taking_interval_ms=6000 Also...

Yes, the parameters are applied to REST API query. The CPU is not a bottleneck in this case, but disk I/O is. You can set up RAM drive and assign snapshots dir to it: snapshot_auto_dir=/ramdrive to prevent disk I/O issues.

No. Video Chat example works with 5.2.1131 on demo server.

Yes, any stream published to WCS server is available to play via WebRTC, including mobile SDKs Yes, if a platform provides an RTMP URL to play Yes. Please look at the following examples: Video Conference Flash Video Chat Android Video Conference iOS Video Conference

Hello >Is there a new WCS release also required? No. This fix is Web SDK only. >will the video chat demo sample need any change? You can track all changes on our github with comment #WCS-3445 https://github.com/flashphoner/flashphoner_client/commits/wcs_api-2.0 Therefore you can see all...

You can try to use Adobe AIR on mobile devices. Flash example applications sources are still available with WebSDK, so you can build them and try with Adobe AIR. If this works, you can publish RTMP streams from mobile devices and play them from desktop as WebRTC using Javascript API.

In WebSDK build 2.0.216, it is possible to pass participant stream playback options. Use this to disable automatic stream audio unmuting in Android Edge browser and to prevent a participant stream freeze. Then, user must click a button to unmute audio var options = { unmutePlayOnStart...

Good day. Unfortunately, Flash is dead and not supported in all the modern browsers and operating systems. But WCS still supports RTMP publishing and playback. You can even build WebSDK from sources including Flash support if you find a browser which still supports Flash. So we recommend to use...

Здраствуйте. Из коробки, такой функции нет. Есть возможность перехвата и обработки сигнала, к которой вы можете подключить свои алгоритмы (фильтры) изменения голоса. https://docs.flashphoner.com/display/WCS52EN/Server+audio+processing На входе сигнал PCM16

It will be released tomorrow

Good day. Please provide an access to the server, SIP PBX credentials and two SIP accounts to make a test call between them using this form.

Good day. Please check the following logs: 1. Server log around the time when server stopped /usr/local/FlashphonerWebCallServer/logs/server_logs 2. System messages log /var/log/messages 3. Check if there is a log file /usr/local/FlashphonerWebCallServer/logs/errorPID.log, where PID is process...

Добрый день. В сборке WebSDK 2.0.215 добавлена возможность определять параметры логирования не только для приложения в целом, но и для сессии, потока или звонка. При этом на каждом из уровней используется свой экземпляр объекта логирования. Пример приведен здесь: Определение параметров...

Добрый день. Возможная причина в Вашей последовательности операций (flow). По логам (в том числе и предоставленным ранее) она выглядит так: 1. Ведущий публикует поток 2. По REST API /mixer/startup создается микшер 3. По REST API /mixer/add поток ведущего добавляется в микшер 4. К потоку микшера...

Добрый день. Если речь идет о захвате потока с камеры на WCS, то поддерживается передача имени и пароля в RTSP URL, при этом служебные символы должны быть экранированы: Аутентификация при захвате потока Если речь идет о проигрывании захваченного потока по WebRTC, то есть возможность стартовать...

Мы проверили Ваш сервер. Если публиковать поток в микшер (например, fptest#fpmixertest), на подключая зрителей к микшеру, то запись успешно создается, и после остановки микшера по таймауту отрабатывает скрипт on_record_hook.sh, в котором запись копируется в AWS S3 хранилище, после чего удаляется...