Alessandro Ros

i was finally able to reproduce the issue, thank you very much!

this should be fixed by https://github.com/bluenviron/mediacommon/pull/269, however when streaming the file with FFmpeg, the server periodically resets the recording because it detects a drift between the stream relative duration and...

Hello Eric, I know that QUIC, WebTransports and also WebCodecs are game-changing technologies that are able to provide video streaming with better reliability, latency, simplicity and control than WebRTC and...

Also WebRTC DataChannels + WebCodecs is work mentioning since is a failback solution, with the same power as WebTransport + WebCodecs, but available on all browsers (while WebTransports is still...

Hello, message "RTP packet lost" has nothing to do with CSeq, but rather with the Sequence Number of RTP packets. In order to allow us to replicate the issue you...

You can run this in a separate binary that pulls data from the server's metrics interface, there's no need to change the server code.

Hello, summing up: Opus in WebRTC requires a precise timing that is often not available from the original source. There was a similar issue with LPCM in WebRTC, that we...

Hello, from the dump you provided it seems you are using a custom-built application to publish a H264 stream to the server. This stream is missing the H264 SPS and...

Hello, Regarding the RTMP issue, the problem is that FFmpeg truncates URLs (and JWTs) down to 1024 characters, causing the error. The constant is here: https://github.com/FFmpeg/FFmpeg/blob/1e76bd2f394a01c19073160c380adbcaa779f474/libavformat/rtmpproto.c#L55 We cannot do anything...

Hello, i checked and there are no differences between setting runOnDemand inside the configuration file or through the API. The misunderstanding here is that a runOnDemand command, as the name...