We are trying to stream video from one server to another (one is a local LAN server, the other is public).
Unfortunately, a server may be behind a NAT which does port replacing. Therefore, we cannot predict the actual remote IP or PORT tuple for the incoming UDP RTP packages on the public server. I was wondering why this information is required in the connect call of the pipe transport?
Would it be a security issue if one would make the check in /worker/src/RTC/PipeTransport.cpp#L609 optional?
What other solution could help us to have the local LAN server (which serves local webrtc clients) also pipe the stream to a “cloud” server which serves public clients?
Yep, sorry. That’s right. You need to provide an exact remote IP and port in both sides and use port redirection if needed. Note that both sides need to know the remote exact IP:port because RTCP travels in both directions.
In mediasoup v4 we are gonna merge plain and pipe transports so it will be easier, but cannot provide an ETA.