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Hello, The demo server is working, RTSP can be played now.

Good day. Since build 5.2.1579 it is possible to capture RTSP stream published by camera as H265: Capturing H265 RTSP stream Also, H265 MPEG-TS stream may be published: H265 publishing Note that H265 requires transcoding to play it as WebRTC. Chrome does not play H265 via WebRTC, only from media...

Good day. Since build 5.2.1579 it is possible to capture RTSP stream published by camera as H265: Capturing H265 RTSP stream Also, H265 MPEG-TS stream may be published: H265 publishing

Good day. Since build 5.2.1579 it is possible to capture RTSP stream published by camera as H265: Capturing H265 RTSP stream

Добрый день. По поддержке RTSP камер с кодеком H265: в сборке 5.2.1579 добавлена возможность захвата RTSP потока, публикуемого камерой в кодеке H265. Подробнее здесь: Захват H265 RTSP потока

Good day. Since build 5.2.1579 it is possible to capture RTSP stream published by camera as H265: Capturing H265 RTSP stream

UDP TURN traffic may be blocked by provider, for example, by packet signature. We recommend to use TCP for relay.

Good day. Seems like server cannot bind UDP port when mixer media session is created: Please check if ports 30000-33000/udp are not used by any other application on the server.

Выглядит так, что от SIP стороны в INVITE SDP приходят кодеки speex 8000 и speex 16000, которые Chrome не может использовать одновременно Поэтому браузер не может применить SDP, которое предлагается сервером. В наших тестах (звонок с софтфона со следующими кодеками: Speex 8000, Speex 16000, GSM...

Пришлите пожалуйста также лог консоли в Chrome браузере. Developer Tools > Network > domain Среди этих сообщений должен быть SDP конфиг.

Good day. STREAM_STATUS.FAILED is received if some error occurs during the stream playback, usually it means the stream with the certain name is not published. You can get a reason using stream.getInfo() method in event handler: participant.getStreams()[0].play(display...

Good day. We cannot reproduce the problem in Media Devices example from WebSDK build 2.0.228: after clicking Test button, then Release button, the camera turns off the signal light. So please check your custom code, may be there is no video tag in localVideo div when unmount event occurs.

Выглядит так, что та сторона присылает кодек speex/8000. Такой кодек не поддерживается в браузере и бросает ошибку. 1. Попробовать выключить кодек speex на стороне сервера в конфиге flashphoner.properties codecs_exclude_sip=speex16 2. Снять дамп SIP трафика между WCS - сервером и внешним PBX...

It supports custom option like any other direct invoke hook (publishStream. playStream etc). Please see this example, the custom parameters passing should also work for call: 1. Client side code: var outCall = session.createCall({ callee: $("#callee").val(), visibleName...

You may publish a stream by one of the following ways: From a web camera in a browser via WebRTC Using RTMP encoder From an IP camera via RTSP Capturing VOD from a file RTP stream publishing via RTSP MPEG-TS RTP stream publishing HTTP incoming stream is not supported. But you can capture HTTP...

WebSDK (API) сейчас работает таким образом, что с сервером устанавливается 1 websocket - соединение, которое имеет 1 набор SIP credentials (login, password, proxy, port). Т.е. нельзя в одном соединении передать два и более наборов SIP credentials. Что можно попробовать сделать - это создать...

Let me know if I'm wrong. You are going to store your mp4 files on a web server and you are going to injest such files to Flashphoner WCS and finally get published stream. For example source file is, https://mywebserver.com/storage/Bunny.mp4. WCS receives this file as a stream and then it can...

sipHeaders are passed to PBX only, but not to other call party. custom is passed in REST hook /call to a backend server. So you should set up your own backend to receive REST hooks and then send a custom data to a client by your own way. Please see the details about REST hooks here.

No. WCS uses only one websocket port. You can also set up nginx-based reverse proxy to use another port.

Hello By default server process starts from user 'flashphoner', not from a root user. Try to launch server as root user. 1. Stop server. 2. Edit config /usr/local/FlashphonerWebCallServer/bin/setenv.sh WCS_NON_ROOT=false 3. Start server as sudo user Or sudo ./webcallserver set-root-mode...