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This issue not occured while we tested you server (see above). Usually, audio may be out of sync video if there are packet losses on publisher or player channel. Please check the channel quality using Media Devices example as described here. You can also try to switch WebRTC transport to TCP for...

Good day. We cannot reproduce the issue. Both JMC and JConsole (Oracle JDK 8) successfully connect to WCS server with the following settings in wcs-core-properties -Dcom.sun.management.jmxremote=true -Dcom.sun.management.jmxremote.local.only=false -Dcom.sun.management.jmxremote.ssl=false...

Please do the following: 1. Check if stream playback on iPhone works in Two Way Streaming example shipped with WCS 2. Carefully modify the TwoWay Streaming code to reproduce the issue 3. Send us the code modified using this private form

We answered you in this topic

Good day. We tested this as follows: 1. Modify the file /usr/local/FlashphonerWebCallServer/client2/examples/demo/streaming/webrtc-as-rtmp-republishing/webrtc-as-rtmp-republishing.js 2. Open WebRTC as RTMP example...

Good day. Yes, it is mandatory to set local renderer to capture video in Android SDK. Before picture is sent to server, it must be drawn on surface. We tested Screen Sharing example application with the latest Android SDK 1.1 build 1.1.0.17, it seems to be working.

We working on it in ticket WCS-3032 as mentioned above. But we do not provide any ETA on forum tickets. So we suppose temporary solutions to use until the problem will be investigated and fixed. If temporary solutions are not acceptable, you'll have to wait for fix. Please also keep the stream...

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.