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Good day. Please update WCS to the latest build 5.2.1040. Then, change the following parameter in /usr/local/FlashphonerWebCallServer/bin/setenv.sh from WCS_NON_ROOT=true to WCS_NON_ROOT=false Then test. If this does not help, please provide SSH access to the instance using this form.

Данная настройка должна быть установлена на том сервере, куда публикуются входящие WebRTC потоки, т.е. на Origin сервере. На остальных серверах ее выставлять не нужно. По конфигурации видим несколько проблем: 1. Уточните, пожалуйста, следующее: у вас настройки HLS активны на всех трех видах...

Good day. Please clarify in what log this message appears? In this is RTMP ingest point log, this can be channel issue between WCS instance and RTMP ingest point. In this case, migrate WCS instance to datacenter closer to RTMP ingest point or vice versa, use ingest point closer to WCS instance...

Добрый день. По тикету WCS-3320. В сборке 5.2.1038 актуальное состояние потока на момент остановки публикации (hasAudio, hasVideo, record) приходит в событии /unPublishStream.

1. Install JDK 14 2. Download installation archive wget https://flashphoner.com/downloads/builds/WCS/5.2/FlashphonerWebCallServer-5.2.926.tar.gz 3. Unpack archive tar -xzf FlashphonerWebCallServer-5.2.926.tar.gz 4. Install WCS cd FlashphonerWebCallServer-5.2.926 sudo ./install.sh Then...

Please read this doc about HLS playback. Thre are 3 HLS samples: - VideoJS based - HLS.js based - Native HLS player (MacOS and iOS Safari)

Good day. Please read this article What kind of server do you need to run a thousand WebRTC streams? All the tweaks are described in chapter Preparing for testing.

We reproduced the problem playing this dump. Will let you know when fix it.

Please clarify what do you trying to do? To manage stream mixing, use mixer REST API To find streams in RoomApi room use /stream/find as shown above. There is no any other REST API to manage rooms.

You should stop a pulled RTMP stream with /pull/rtmp/terminate REST query POST /rest-api/pull/rtmp/terminate HTTP/1.1 Host: locathost:8081 Content-Type: application/json { "uri":"rtmp://abc.xyz:1935/live/ch_1" } Then you can start a new stream with another source by /pull/rtmp/pull...

Добрый день. Видео может начать проигрываться только после получения ключевого кадра (I-frame или key frame). Поэтому чаще всего долгое проключение связано с редким получением ключевых кадров. Если Вы публикуете поток по WebRTC или захватываете с RTSP камеры, в этом случае поможет настройка...

We checked you server. It seems working: But there's not enough memory for server to work on the instance: The minimum required is 2 Gb RAM (1 Gb for Java heap). Otherwise, server can stop due to out of memory. Please add more RAM to the instance.

Good day. Please collect a report as described here and send using this form.

If you're using AWS Marketplace AMI, you have not pay additional charges, just hourly payment for instance. If you're deploy Amazon Linux for example and install WCS manually, you should buy a license.

Good day. Please provide SSH access to the instance using this form

Good day. You can achieve this using REST API /pull/rtmp/pull: POST /rest-api/pull/rtmp/pull HTTP/1.1 Host: localhost:8081 Content-Type: application/json { "uri":"rtmp://abc.xyz:1935/live/ch_1", "localStreamName":"ch1_for_viewers" } In this case you should use localStreamName...

Good day. Seems m5.2xlarge should be enough. Please also make a tuning steps described in this article, chapter Preparing for testing, points 1-5

Добрый день. WCS работает только на Linux x86_64. Поэтому нет, на Android устройстве WCS запустить нельзя, нужен отдельный сервер. Минимальные требования приведены здесь.

This depends on streams published count. You will have as many subscribers on this server as streams published. Please read this article about one-to-one streaming testing. The server with 24 CPU cores, 80 Gb RAM is enough to handle 200 720p 2500 kbps streams in this case.

Это ограничение библиотеки WebRTC, код внутри которой мы не меняем. iOS SDK обеспечивает только websocket сигналинг и вызовы для установки WebRTC соединения.