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WCS-cервер установлен (последняя версия - 2641; настройки по умолчанию); активирована лицензия и импортированы сертификаты. Дополнительно на сервере ничего не устанавливалось и не менялось.

How streams are published and sent to Wowza? rest-api/push/terminate would be applicable if a stream published on WCS is pushed (rest-api/push/startup) to Wowza (i.e., there are two streams – one on WCS and one on Wowza): stream on Wowza will be terminated on the 'terminate' request.

Hello We plan to take the issue in progress. Please do not close RTSP stream for further testing.

The way stream name is composed can be changed in room-module.js.

Tried this with the latest version (0.5.25.2455 - 5.0.2640), Conference demo example in Chrome on Win: 'video' changed to false in conference.js, - no error occurs on publishing. Please add more information - OS or device, browser - access to microphone - what error

Начиная со сборки flashphoner-api-0.5.25.2455-4cc23064647cec5c915aba3bfecd00793a522d08: examples/demo/dependencies/mse/media-source-media-provider-no-babel-polyfill.js

For general product line it is currently out of our plans. Please contact sales@flashphoner.com, perhaps we will be able to arrange custom implementation for your case.

Hello, No updates yet. The issue is in our backlog. We will inform once we have any updates. It may take a time.

For publishing var constraints = { audio: {bitrate: 44100}, // 44100 bps video: true }; session.createStream({ name: streamName, display: localVideo, constraints: constraints }).publish();

As I see, you tested over UDP. Try to setup firewall to test TCP. Your server likely will show the same results.

Our server does not disable UDP traffic. You have to disable UDP locally on your firewall or router to check WebRTC/TCP streaming because if you do not disable UDP it will be connected over UDP in priority. I tested with following local firewall rules: 1. DNS over TCP allow 2. DNS over UDP...

You have to publish RTMP stream to the server using your RTMP client software and URL rtmp://wcs5-eu.flashphoner.com:1935/live Then play this stream in the Firewall Traversal Demo page

Hello We have reproduced this issue with your stream and server build 2631. I will inform you once we have any news. Stream looks good according ffprobe: Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709, progressive), 1280x720, 29.97 tbr, 90k...

Hello Can't reproduce. This is how we test: 1. Publish RTMP stream from Wirecast to WCS server as H.264+AAC 2. Play stream as WebRTC via Two Way Streaming example on iOS Safari 11. https://wcs5-eu.flashphoner.com/client2/examples/demo/streaming/two_way_streaming/two_way_streaming.html 3. Play...

Нашли утечку дескрипторов, связанную с коннектами с Flash Player Исправили в сборке 2621. Других утечек воспроизвести пока не удалось под нагрузкой. Обновитесь пожалуйста до сборки 2621 или выше чтобы проверить фикс.

Вмержили. Но до продакшена можем не успеть довести за оставшиеся 2 недели. Скорее всего релиз новой ветки сдвинется на январь 2018. Сейчас в новой ветке wcs5_monitoring, начиная со сборки 2617, добавлен функционал пуллинга стрима с origin-сервера по запросу с edge. Настройки Edge...

You can just try with our TURN server. 1. Open similar 'Firewall Traversal Demo' demo on your server. 2. Fill out TURN credentials. 3. Disallow UDP traffic on your firewall. 4. Try to play your stream over WebRTC/TLS.

It seems we have to implement adaptive server-side buffer for RTMP to accumulate gaps and provide smooth WebRTC playback. It may take a time. As an option you can try to play over our TURN server WebRTC via TLS 443 port...

Можно также попробовать оставить из голосовых кодеков только ulaw способом, описанным выше.

Попробуйте откатить изменения на клиенте и убрать кодек g722 на стороне сервера. Для этого на севрере в конфиге WCS_HOME/conf/flashphoner.properties нужно убрать кодек g722 Заменить codecs=opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv На...