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Good day. Please make sure you've done the perfomance tuning described here (espesially p 11, 12, 13), and stream distribution optimization is enabled. Also, you coonfuguration seems not enough for 1700 subscribers. In our tests in the article What kind of server do you need to run a thousand...

Good day. All the special characters in RTSP URL must be escaped (URI encoded). Also, username should be divided from password by :. For example, the correct URL is: rtsp://username:oaass%40123@someit:544/someaddress

Checked the server - the admin panel is working.

Please provide SSH access to the instance using this form, we will check

Good day. Seems like the server is not running. Please restart it sudo systemctl restart webcallserver If the problem still persists, please provide SSH access to the instance using this form

This behaviour concerns AWS EC2 instances only. In AWS EC2, we recommend to clean the logs keeping folders structure: find /usr/local/FlashphonerWebCallServer/logs/ -type f -delete rm -rf /usr/local/FlashphonerWebCallServer/logs/client_logs/*

Good day. This is client logs pushing feature. It is intended to push client browser logs to server logs and is working for WebSDK only. All the stream states are logged in server log on the INFO level: 07:30:06,819 INFO RestClient - API-ASYNC-pool-13-thread-8 SEND REST OBJECT ==>...

Good day. All the SIP metrics are described in this table (see call_stats and sip_stats sections): sip_calls Number of SIP calls sip_calls_established Number of active SIP calls sip_calls_in Number of incoming SIP calls sip_calls_out Number of outgoing SIP calls...

Pulling HLS (using for playback HLS URL for a stream published not on the WCS) is not supported. External stream can be pulled as RTMP or RTSP. Stream published on WCS can be played as HLS.

This option enables WebRTC over TCP feature. In this case, RTP traffic goes over TCP instead of UDP. It requires the same media ports to be opened for incoming connections, for example 31001-32000/tcp. Those ports are used subsequentally. TURN feature is intended to bypass firewalls because it...

Good day. There is no known TURN server implementation working with WebRTC over TCP, because TURN actually wraps media UDP traffic to TCP. So you can either use TURN or use WebRTC over TCP, but not both.

WCS-3389 - исправлено в сборке 5.2.1092.

Добрый день. По этой проблеме есть тикет WCS-3389. Сообщим здесь, когда будет исправление.

Добрый день. В сборке 5.2.1091 добавлена возможность описать вариант размещения картинок с демонстрацией экрана для определенного количества участников. Подробности читайте здесь.

Good day. You should not buy any additional license because recording features are already included. You can record all the REST streams captured by adding the following parameter to flashphoner.properties file: record_rtsp_streams=true Please also read the details about stream recording here.

We checked your server. About PHP warning: the reason is here: $api_method = array_pop(explode("/", $_SERVER['REQUEST_URI'])); Just replace it to $request_uri = $_SERVER['REQUEST_URI']; $api_method = array_pop(explode("/", $request_uri)); and warning should not appear again. About...

Please check line 2 of your index.php script, may be you're passing array to a function. In the log we see that stream is not publishing due to NPE 01:57:08,866 ERROR MediaHandler - API-ASYNC-pool-7-thread-119 publishStream exception java.lang.NullPointerException ... 01:57:08,866...

If there is no transcoding on WCS while republishing a WebRTC stream as RTMP, outgoing RTMP stream bitrate is always the same as published strem bitrate. Our tests with the latest Firefox build confirm that. So please check WebRTC incoming bitrate on WCS using REST API queries /stream/find or...

You should buy a valid SSL certificate, then import it via web intefrace as described here.

Good day. We fixed the connection to an existing session issue in WCS build 5.2.1087, Phone Min example corresponding fix is in WebSDK build 2.0.206. Please read this docs for details: Connection to an existing session We also raised the tickets to implement this feature support in Android SDK...