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The example above is a sample how to implement, not a working code. So please provide your code to check or access to webpage to test using this form. Please test in the Conference example. Are there any text messages loss? If not, seems like the problem is in your code.

Прежде всего, отметим, что трансляция экрана не работает в мобильных браузерах, поэтому необходимо использовать мобильный SDK в нативном приложении. Посмотрите, пожалуйста, описание примеров Android Screen sharing и iOS Screen Capturer Swift. Проверить задержку можно следующим образом: 1...

So, you publish WebRTC and play it as RTMP from server, right? In this case, you should catch /connect and /publishStream REST hooks on your backend (as we recommenede above), get direct server instance ip from sessionId field, form RTMP link like rtmp://direct_instance_ip:1935/live/streamName...

Здравствуйте. Должна быть задержка меньше секунды.

Hello All the streams are traced in SDR logs https://docs.flashphoner.com/display/WCS52EN/WCS+Core+logs#WCSCorelogs-SDRlogs 1. Reproduce your issue 2. Find affected stream in the latest sdr.log 3. Attach full sdr.log and logs/server/flashphoner.log Report form From your logs we can see that...

Проверили Ваш следующий отчет. Сам по себе отчет содержит ровно то же самое. Но по предоставленному коду можно предположить, что Вы используете в качестве источника картинки канвас. В этом случае фризы при уходе страницы в фон являются ожидаемым поведением. Использование setTimeout в этом случае...

Use server statistics page http://wcs:8081/?action=stat&groups=port_stats. You can also set up Prometheus+Grafana as described in this article 10 Important WebRTC Streaming Metrics and Configuring Prometheus + Grafana Monitoring, and get a graph for ports statistics.

Добрый день. К сожалению, в Вашем отчете не содержится ни кода Вашего приложения (предоставленная веб страница не содержит кода публикации), ни серверного лога, поэтому мы не можем даже определить используемый Вами в тестировании браузер. Дамп, который Вы собрали, также бесполезен без серверного...

By default, RTP bundle support is enabled on WCS, so 1 port is used to transfer video and audio packets. So every participant uses 1 port to publish its stream, and 3 ports to play other participants streams. Therefore, every participant uses 4 ports, 16 ports should be busy while all the...

By default, media port auditor interval is 5000 ms. So a port will be released in 15 seconds (3 tries, 5 seconds each). Please also clarify the following: 1. How do you test in details? 2. How much connections you use simultaneously (check statistics page)? 3. Please make sure you use the latest...

Домен должен быть привязан к entry point балансировщика, а не конкретного сервера.

You should not use receiveVideo or receiveAudio when playing stream. Use constraints to play audio only from audio+video stream: session.createStream({ name: streamName, display: remoteVideo, constraints: { audio: true, video: false...

If you're republishing WebRTC stream as RTMP to the same server, use RTMP URL like this rtmp://localhost:1935/live/streamName. In this case, RTMP stream will always be published to the same WCS. But if you want to connect to right WCS instance to play the stream published, you should use backend...

Good day. You can decrease media port checking interval: media_ports_auditor_interval=2000 In this case, a media port should be freed after 6 seconds (3 times 2 seconds each to check).

No, there is no any additional charge. Only the server instance itself is billed hourly by AWS. This depends on server capacity. Please read this article How many RTSP cams can I connect to the WebRTC server?

Please clarify: do you publishing RTMP to WCS and playing it as WebRTC, or vice versa? Anyway, in this case you should arrange a separate backend server and catch REST hooks /connect and /publishStream to detect on which server a stream is published. Then, you should send server IP address to...

Good day. We cannot reproduce the problem on our demo server in Media Devices example in Safari browser: 1. Publishing a stream with Send audio disabled (statistics shows audio only is publishing) 2. Playing a stream in Media Devices (all the swithes are by default). Audio is playing correctly...

WCS should work on all Centos 7 branches, we've also successfully tested it on Centos 8 and Centos Stream

You should add HTML5 audio element on the page with a desired audio file, and start to play it when CALL_STATUS.RING is received. then stop playing it when CALL_STATUS.ESTABLISHED is received (or other status different from CALL_STATUS.RING).

Good day. This issue was fixed in latest WebSDK build 2.0.198, please update and check.