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You can stream RTMP with Gstreamer to WCS then pull it from WCS with WebRTC, RTSP or RTMP.

We will check and let you know in this topic

You can use two Gstreamer pipelines simultaneously as supposed here. If it does not work for the same source device, it looks like Gstreamer limit.

Good day. We have tested RTMP translation from Gstreamer to WCS. The stream is published successfully with the following command gst-launch-1.0 -e videotestsrc ! queue ! videoconvert ! x264enc ! flvmux streamable=true ! queue ! rtmpsink location='rtmp://WCS_IP:1935/live/myStream'

Добрый день. Мы протестировали трансляцию RTMP потока при помощи Gstreamer на WCS. Поток успешно публикуется. Пример командной строки: gst-launch-1.0 -e videotestsrc ! queue ! videoconvert ! x264enc ! flvmux streamable=true ! queue ! rtmpsink location='rtmp://WCS_IP:1935/live/myStream'

Добрый день. Проблема исправлена в сборке WebSDK 0.5.28.2753.85, пожалуйста, обновите WebSDK и проверьте на своем сервере. На демо сервере demo.flashphoner.com WebSDK также обновлен.

Good day. We fixed this issue in WebSDK build 0.5.28.2753.85, please update WebSDK and check.

Good day. Flashphoner WebCallServer can play mp4 files as VOD, please see details here. So if you IP camera can only save recording to a file, you can convert this file with ffmpeg to H264+AAC in mp4 container, than play it as VOD in browser. But if your IP camera can stream RTSP, with...

Hello. Please clarify: you start a voice call like Phone example, then publish a video stream like Streamer example? In this case, it's probably a good solution to get access to media devices before making a call, look at Phone Video example source code.

Добрый день. Если Вы не используете REST hook для взаимодействия с бэкендом, можно отключить REST клиента полностью настройкой в файле flashphoner.properties disable_rest_requests=true Если REST hook нужны, можно ограничить логирование REST клиента, для этого в файл...

Good day. We work on RTMPS support and let you know on results.

Created internal issue WCS-1957, will inform when there is an update. Bitrate and FPS are changing in real-time. The functionality for monitoring is under development. Currently, WebRTC statistics via Web SDK can be used as described here.

Set global_bandwidth_check_enabled to true to enable bandwidth stats: in WCS_HOME/conf/flashphoner.properties global_bandwidth_check_enabled=true

Hello, Yes, using CLI: ssh -p 2001 admin@localhost User 'demo' can be disabled update user -a false demo or, the password can be changed passwd demo

In case cdn_ip=<hostname>, the other nodes will be able to connect if the hostname resolves to the server public IP. That can be check using ping - ping <its own hostname> - on the server. If the hostname resolves to 127.0.0.1, edit /etc/hosts: remove lines resolving the hostname not to the...

Добрый день. Мы работаем над этой проблемой, о результатах сообщим здесь.

Crash logs show that something is happened when recording is stopped, so please collect logs as described here and send to support@flashphoner.com If problem still persists you can switch recording off as quick workaround.

Good day. The similar issue was fixed in build 5.2.86, CDN node that uses transcoding can crash under circumstances. After this build, there was another fixes, so we recommend to update to build 5.2.96 and check if problem still persists

Проверили нарезку HLS при транскодинге в CDN с входящим RTMP-потоком - проблема не воспроизводится. Если подобное повторится, сохраните и вышлите, пожалуйста, hls-файлы (директорию WCS_HOME/hls/<имя потока>). Есть старые доступы (не действуют); вышлите, пожалуйста, новые.

Добрый день На видео заметна ошибка в консоли Overconstrained error с указанием на недопустимую ширину картинки. В связи с этим пожалуйста, уточните, какую камеру используете. Кроме того, проверьте, работает ли стриминг в примере Media Devices, там можно переключать разрешения (по умолчанию...