How RTP packets are handled in case of RTSP to RTSP proxy

[gizak]

Question

Hi I recently checked this awesome project. It is everything I need for my work (multiple protocols and stream routing, etc.). But currently I encountered a problem: I need to setup a RTSP server, with one end pushing the stream to it and read out from another end. The RTSP worked fine but it seems that all RTP packets were "reinterpreted" or translated and re-packed.

Here is how packets sent (H264 mode0, no STAP, no FU)

Here is how packets read from mediamtx server:

It seems that all sending packets were decoded and re-packed. But in my case I need H264 mode0 (packetization-mode=0, no aggregation, no STAP, no FU) to be compatible for some old receiver. How the RTP packets are handled from RTSP to RTSP? Is there a workaround to allow RTP passthru or similar?

[aler9]

Hello, the situation you are describing is caused by two separate issues: first, we are currently remuxing all packets that have a length greater than the UDP MTU. This allows us to produce streams that are compatible with almost any client. The second issue is that i looked around and i found out that we are currently not supporting H264 packetization-mode=0, therefore the remux is always in packetization-mode=1, but this is wrong.

I think the first step into solving this situation is to support packetization-mode=0. I've created a feature request here: bluenviron/gortsplib#585

Then we'll try to see what to do with the automatic remuxing feature. Since FU-A packets are not supported with packetization-mode=0, we could disable that, but i need to think about this.