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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...

Для использования в продакшн среде, да. Встроенный веб-сервер WCS предназначен для тестирования, то есть Вы можете модифицировать код примеров WebSDK, расположенных в каталоге /usr/local/FlashphonerWebCallServer/client2, и добавлять туда свои примеры, тестировать их, но на продакшн нагрузку это...

Is the issue reproducing in Two Way Streaming example? If not, please modify the example code to reproduce the issue and send using this form

What application do you mean? Please provide the link, the application code, the RTSP URL to test using this form You can play all the recorded streams either in browser directly (if the files are in /usr/local/FlashphonerWebCallServer/records folder) by URL...

You can play recording files in browser directly by URL https://wcs:8444/client/records/file.mp4, but not as HLS or live RTSP.

Good day. Since build 5.2.1088, multiple streams recording folder can be set using the following parameter multi_record_dir=/opt/media Note that this folder should be writable, for example: sudo chmod o+w /opt/media

Then, clarify how exactly do you test: what stream name do you set to Stream field in Player example? Note that you cannot play a file this way, as we mentioned above. You can play only WebRTC stream, RTSP stream or RTMP stream.

Use full path to run keytool: sudo /usr/java/jdk-14.0.1/bin/keytool ...