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Good day. The main reasons for stream freezes and artifacts usually are: 1. Bad channel quality between publisher and server and/or between player and server. This may be prevented by switching to TCP transport on client side session.createStream({ name: streamName, display...

Good day. You can ignore this message, this means some exception in native C++ code which is handled internally. MCU conference uses stream mixer, so the more participants, the more CPU load. Please read the article How many mixers can a server have? about mixer load testing. Note that MCU...

Добрый день. В последних сборках, доступных на Docker Hub, начиная с 5.2.1045 сообщения /usr/local/FlashphonerWebCallServer/bin/webcallserver: line 315: sudo: command not found при запуске контейнера в интерактивном режиме выводиться не должны.

Yes, WCS can be RTMP server for third party players by default You can prevent RTMP access at all using this parameters: rtmp.address=127.0.0.1 rtmfp.address=127.0.0.1 In this case, WCS will listen RTMP port on local interface only, so RTMP server will not be available for external clients...

Предоставьте два SIP аккаунта на Вашем SIP-сервере для тестовых звонков, используя эту форму

В сборке 5.2.1049 добавлена возможность извлечь информацию о дорожках в файле multi-recorder при помощи offline_mixer_tool. Подробнее здесь.

Добрый день. К сожалению, Вы предоставили логи только за более позднее время, чем указано в комментариях к отчету. В дальнейшем просим указывать имя проблемного потока, чтобы можно было идентифицировать его в логах. Заикание с последующим рассинхроном обычно говорит о проблемах в канале...

Добрый день. На приведенном скриншоте видно, что не устанавливается WebRTC соединение, т.к. не проходит ICE. Проверьте доступность медиа портов сервера (порты из диапазона media_port_from-media_port_to, по умолчанию 31001-32000/udp) для входящих соединений, например, по этой методике. Также...

Please clarify how do you test: 1. How do you publish a stream? 2. How do you play the stream? 3. What exactly video becomes black: local video while publishing or remote video while playing? 4. What example do you use to test? Is the problem reproducing in Two Way Streaming example? If not...

Добрый день. Скорее всего, на канале пользователя есть потери. Переключение на TCP транспорт для audio потока должно помочь: const config = { ... transport: 'TCP', ... } Кроме того, битрейт для видеопотока у Вас выставлен некорректно: параметры maxBitrate, minBitrate...

We would not recommend letsencrypt in production. Just purchase an SSL certificate like this (namecheap.com/security/ssl-certificates/) for your domain and forget SSL issues for 1-3 years. Perhaps the iOS system you are testing does not know about certificate authority that issued the...

As we see, this page uses player.js default script to play. <script type="text/javascript" src="https://flash.polycall.net:8444/client2/examples/demo/streaming/player/player.js"></script> This script uses default constraints, and make a browser wait for both audio and video packets to be...

You should change this in add_wcs_to_conference.sh script too: curl -s -X POST -d "{ \"callId\": \"5620-$CONF_NUMBER\", \"dtmf\": \"1\", \"type\": \"INFO\" }" "$URL/send_dtmf" We've done it, and the error gone, Asterisk displays in logs [2021-09-29 23:13:10] VERBOSE[19955][C-00000301]...

Hello We have seen DTMF error WCS: 08:22:38,400 INFO RestApiRouter - HTTP-pool-3-thread-193 Use controller class com.flashphoner.rest.server.rest_v2.RestCallController with path /rest-api/call/send_dtmf 08:22:38,400 INFO RestCallController - HTTP-pool-3-thread-193 handleRequest...

We checked again with the following test: 1. Call from softphone1 to the extension number provided (7002). 2. Stop the call on WCS side by /rest-api/call/terminate. The hold music plays in softphone 3. Call from softphone2 with credentials defined in add_wcs_to_conference.sh script to the...

Good day. We fixed the issue in WCS build 5.2.1047. Web SDK should also be updated to 2.0.199.

Good day. We fixed the issue in WCS build 5.2.1047. Web SDK should also be updated to 2.0.199.

Good day. About the ticket WCS-3271. We fixed the issue in build 5.2.1045, RTSP stream from dump you've provided is playing successfully.

We checked the flow. Seems like Asterisk sends silence when WCS established call to the conference. If we establish SIP call via our test Asterisk/OpenSIPS PBX, all seems working. So this is may be Asterisk configuration issue. One issue is in SIP call parameters: sipDomain parameter ends with...

Good day. Yes, you can: 1. Publish WebRTC streams From a web camera in a browser via WebRTC 2. Add those streams to mixer automatically Automatic mixer creation configuration or using REST API REST queries 3. Republish mixer output stream as RTMP using REST API REST queries