Hi,
I've been looking at a transport protocol (SRT) preserving RTP headers. RTP isn't a supported scheme for the sample app I'm using (srt-live-transmit) but I can define the chunk size and decode a transport stream after conversion to and back form SRT.
My setup is a server running the app taking in an RTP stream (defined as udp with a chunk size of 1328 to get the app to work) that's flipped into an SRT stream, send over a LAN to a receiving server running the app to flip it back to udp://127.0.0.1:1234. I can then use a transport stream analyser on the recieving server looking at rtp://127.0.0.1:1234 to decode the stream.
My input stream has 3 sequence errors in it which appear in the output stream but when comparing the packets I see that while the SSRC is the same, the sequence numbers and timestamps have changed.
Reading the RFC I have interpreted that the app is working as a 'translator' preserving the SSRC and the timestamps are different as they are derived from the sampling clock at the sender, which would be different when sending from srt-live-transmit to the transport stream analyser application.
I've not done much work with these protocols to this level of detail before. Could anyone help confirm or deny this please?
Thanks,
alx
