Hello everyone,
I am trying to add an rtp participant to an audio bridge room (allow_rtp_participants is true). I want to have a different address to send the rtp to the janus than the address specified in the join request to receive the rtp from the janus. I saw in the code and from what I actually experienced (using wireshark and tcpdumps) that the sender and receiver addresses should be the same (i saw that the code uses the same file descriptor for the bind() and connect()).
is that the case? Is there a way to work around this?
When you use RTP participant, the RTP you establish is bidirectional. This means that you cannot use one address as the source of RTP packets, but receive packets on a different address. The address you use to send and receive packets must be the same, as it would be in a SIP call for instance.
Thanks for your answer!
I try to use instead in rtp forward(rtp participant to send audio and rtp forward to receive audio). But when I try to start rtp forward i see in Wireshark(recorded in janus) that the rtp packets not valid(I try to decode as rtp and gets random rtp events). This occurs even with one webrtc participant.
I use janus version 1.2.3.
Edit: just to be clear, the srtp is defined false so it’s not encrypted