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As another workaround, you can also restream local RTSP camera to WCS using ffmpeg, for example ffmpeg.exe -rtsp_transport tcp -rtsp_flags prefer_tcp -i rtsp://local_rtsp_camera_url -preset ultrafast -acodec aac -ar 48000 -vcodec h264 -strict -2 -g 48 -f flv "rtmp://wcs:1935/live/rtsp_stream"...

We reproduced the issue with your RTMP stream and raised the ticket WCS-3032. Will let you know about results in this topic. As workaround, please try to play RTSP stream instead if this is available for WCS server.

We tested publishing and playback on your server. Web admin interface port 8444 seems to be blocked, so we connected using example pages on other host. The stream is succesfully published vis WebRTC and played via WebRTC and HLS So please refer to the docs about publishing WebRTC and playing...

The playFirstVideo() cratch is intended only for Safari browser (both for MacOS and iOS). This function is not necessary on Windows, Linux and Android browsers (but it does not break something if called).

Good day. Please provide SSH access to your WCS and RTMP stream URL to capture using this private form (please insert RTMP link to RTSP URL field). We will check.

Добрый день. В сборке 5.2.872 добавлена возможность передать параметры, аналогичные настройкам микшера в файле flashphoner.properties, при создании микшера по запросу /mixer/startup. В этом случае параметры будут применены только к этому экземпляру микшера. Например, запрос { "uri"...

Добрый день. В сборке 5.2.872 добавлена возможность передать параметры, аналогичные настройкам микшера в файле flashphoner.properties, при создании микшера по запросу /mixer/startup. В этом случае параметры будут применены только к этому экземпляру микшера. Например, запрос { "uri"...

Please check WebRTC as RTMP example source code. By default, webserver URL is set to url field function init_page() { $("#url").val(setURL()); loadPlayer(); ... } You can set any URL to this field, for example function init_page() { $("#url").val("wss://wcs_address:8443")...

You should call Session.createStream() and Stream.play() only when promise returned by playFirstVideo() is resolved, and in context of the promise. Please see Two Way Streaming example, playing section function playBtnClick() { if (validateForm("playerForm")) { ... if...

Usually, we do not provide any ETA on forum. Our working hours is 10:00-20:00 GMT+3 at workdays. We checked your server. Seems like WCS cannot listen default websocket ports because these ports are busy by Apache In /etc/apache2/conf/httpd.conf you wrote and If you need to proxy websocket...

RTSP push is not supported. You can just pull RTSP stream from endpoint.

Unfortunately we do not offer support sessions like zoom, skype and other messenger. It is out of forum-level support. We need direct SSH access (IP address, password or public key authentication). You can submit SSH access directly using private form suggested above. For better security you...

Please use the form by this link.

Please clarify what you mean under server input field.

To reproduce the issue, please send us the application code using this link.

For some reason the request get the redirect response. Please send us SSH access to your server using this form - we'll check the setup.

We cannot reproduce this issue in sample applications Conference and Video Chat with the latest Android SDK 1.1 build 1.1.0.17 (devices: Samsung S10+ and Xiaomi Redmi 9, both Android 10). Please check if the issue is reproducing with sample applications. If not, please modify the code minimally...

Good day. You should place your webserver and WCS instance on server available from China. See AWS docs for example. Another option is to use VPN to access your webserver and WCS instance from China.

Yes, RTMP streams from other servers can be pulled to WCS (and in that case RTMP urls can be used in the embedded player). In this case, if the stream is published to demo.flashphoner.com, then it should be available for playback by its RTMP url in VLC and in the embedded player on your server...

If you're publishing RTMP stream to WCS and playing it via WebRTC, you should play it by its name, not by full URL. If stream is published by URL rtmp://demo.flashphoner.com:1935/live/stream1, it must be played as stream1 1. Publish (file name only differs) 2. Play