I am building an application with gstreamer-webrtc in which timestramp in RTCP and RTP is curcial . while using a mediaserver SFU (mediasoup) the media server consumes and consumes/terminates and produces its own RTCP packets. https://mediasoup.discourse.group/t/forwarding-ntp-timestamp-from-source/3427/4?u=usama. I need the original RTCP packets. is there any other SFU that does not consumes/terminates the RTCP and just relay it
I have built something similar with Pion, see broadcast. One WebRTC Peer uploads RTP, and then it is fanned out to all the other peers unmodified.
You can do the same with RTCP see rtcp-processing. Accept the incoming RTCP packets and then write to all connected receivers.
If you want to put code up on GitHub happy to help with any specific API questions!