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Hello Please upload latest huge logs if you didn't clean all. /usr/local/FlashphonerWebCallServer/logs/server_logs/flashphoner.log This log is rotated hourly. We would check few hours where logs are too big. You can send download link via Report Form What was your previous version? Before...

No. You have to change just AWS rules. The AWS rules must match current settings. For example. If current settings are media_port_from =31001 media_port_to =32000 AWS rules should be: TCP 31001-32000 UDP 31001-32000

Regarding to TCP. It fails because TCP ports are closed on Amazon AWS security. 1. Check AWS instance security group settings. 2. Add TCP ports range: TCP 31000 - 33000 3. Make sure that media port range is matched to flashphoner.properties settings: media_port_from = 31000 media_port_to =33000

Regarding flashphoner.properties config All the settings and it's defaults https://docs.flashphoner.com/display/WCS52EN/Settings+file+flashphoner.properties Therefore if you don't see record_streams=false line this means it has default value

Hello Full Docs https://docs.flashphoner.com/display/WCS52EN Recording Docs https://docs.flashphoner.com/display/WCS52EN/Stream+recording Recording Examples https://flashphoner.com/embedding-webrtc-stream-recording-into-a-web-page-2/ https://docs.flashphoner.com/display/WEBSDK2EN/Stream+Recording

It seems you have a high packet lost rate. Try to switch to tcp: ice_tcp_transport = true https://docs.flashphoner.com/display/WCS52EN/Settings+file+flashphoner.properties

Good day. RoomApi is just a messaging signaling wrapper. You can play any stream published to server at participants page. For example: 1. A participant enters the room (without publishing a WebRTC stream) 2. A participant publishes RTMP stream with a predefined name like room1-user1 3. Other...

Обычно мы, наоборот, советуем включить TCP транспорт в случае потерь. Если использование UDP не приводит к явным проблемам, можно оставаться на этом транспорте, в этом случае ошибки в логах, свящзанные с получением битых пакетов, следует игнорировать.

SSL seems working corfrectly. But IP addresses (ip and ip_local settings) were incorrect. We fixed it using the command sudo /usr/local/FlashphonerWebCallServer/bin/webcallserver set-ip and then restarted the server. Now WebRTC publishing and playback works

Эта ошибка означает, что кто-то пытался зайти на HTTPS порт или WSS порт сервера, используя HTTP/WS. Можете ее игнорировать, если это не ваши клиенты Это может означать какие-то сетевые проблемы между WebRTC клиентом и сервером: контрольное сообщение было получено не полностью. Чтобы исключить...

Good day. We can't reproduce this on our test server with build 5.2.1847 (default settings) You can try to increase a maximum snapshot taking duration: Configuration snapshot_taking_interval_ms=10000 snapshot_taking_attempts=100 We already fixed this.

Please refer to the source code sample on GitHub: if(session && remoteVideo) { session.createStream({ name: streamName, display: remoteVideo }).on(STREAM_STATUS.PENDING, (stream) => { ... }).play(); } Usually, no additional parameters needed to...

Please refer to ACM docs or ask AWS support.

Add 30000-33000/TCP please. You can use AWS Certificate Manager. Please read this article or refer to AWS docs.

We successfully connected to your server, but you have not provided a valid SSL certificate for your domain, so nothing to import. Please buy a valid SSL certificate from some CA provider or register a domain with AWS Route53. Also seems like media ports (30000-33000) are blocked on server (both...

The pem file and IP address should be enough for SSH connection

Please read about the recording feature here: Stream recording

Please provide a valid RTSP URLs for both sources to compare and test using this form. We can't even detect a model by a brand name (Ganz, DigiEye etc) so stream access is needed. Or you can provide SSH access to the WCS instance and private RTSP URLs available from the server using this form.

The Media Devices example is too complex to convert it to one component. It should be used only as a bundle of code samples to implement them. Please see also this article where Media Devices functions are split to a code samples, one per function.