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Hello This is working scheme. Make sure AWS MediaLive supports RTMP ingestion. Webcam > [WebRTC] > WCS > [RTMP] > AWS MediaLive See also similar forum thread https://forum.flashphoner.com/threads/from-web-browser-to-amazon-ivs-server.13438/#post-26542 Website guide: Testing...

Good day. By default, open files limit is set to 10000. You should increase this value by settin LimitNOFILE[/ICODE parameter in service description file [ICODE]/etc/systemd/system/webcallserver.service, for example, to 1000000 [Service] User=flashphoner Group=flashphoner LimitNOFILE=100000 ...

Good day. Since Web SDK build 2.0.202 we've added Two Way Streaming application example in Vue.js. Please read this doc and review GitHub sources. In our tests, audio only streams are successfully playing in iOS Safari, and audio+video too.

WCS requires ip and ip_local to be IPv4 addresses, IPv6 is only additional address. In our test above, we used private IPv4 network 172.16.0.0/12 (like Docker for example). So you should use such network. We raised the ticket WCS-3350 to implement local IPv6 setting to bind ports if this possible.

Good day. ip and ip_local should always be IPv4 addresses You should also set WCS internal TURN server to listen other port if you are using port 3478 to connect to coturn on the same host turn_port=3479 Or bind coturn to port 443 for example. Also, you should use transport=tcp. ip_v6...

Добрый день. Пожалуйста, соберите отчет, как описано здесь, и пришлите, используя эту форму. Уточните также, как играет выходной поток микшера в момент, когда проблема воспроизводится: есть ли в потоке фризы? Какова загрузка CPU в этот момент?

Microsoft dropped support for Legacy Edge. We recommend to use Chromium Edge for WebRTC publish/playback, it works just like Chrome (on which is based).

Если Вы переключили публикацию аудио на TCP транспорт, как мы рекомендовали выше, это максимум, что можно сделать: TCP обеспечивает гарантированную доставку. Это может означать, что битрейт публикации выше, чем пропускная способность канала. В этом случае поможет только полная остановка...

Good day. Since Web SDK build 2.0.201 we've added some React integration examples: Two Way Streaming React Conference React

Добрый день. В сборке Web SDK 2.0.201 исправлена проблема, приводившая к ошибке. Кроме того, добавлены примеры интеграции Web SDK в React приложение: Two Way Streaming React Conference React

The parameter stream_record_policy_template is applied to any multiple stream recording file, including room recording. So you can use it, but with soem known limits. Please read details here. We raised the ticket WCS-3347 to add this setting. As workaround, you can modify...

Pleas note: if you are using custom backend replacing default RoomApp your custom application must forward all the queries to the standard http://wcs:8081/apps/RoomApp and must return all the responses from it to client. For example: $api_method = array_pop(explode("/"...

Server is configured. You are testing in iframe. 1. Try to test directly using link: https://demo.flashphoner.com/client2/examples/demo/streaming/video-chat/video-chat.html 2. Check web console messages F12 Some ad-blockers can affect scripts inside of iframes.

Hello How did you test video chat sample? This is how we test video chat and how it works step by step: 0. Open page https://demo.flashphoner.com/client2/examples/demo/streaming/video-chat/video-chat.html 1. Enter unique nickname "123" 2. Click "Join" 3. Copy invite link 4. Open your...

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.