Greetings, I’m currently engaged in a project where it’s imperative that the SDP remains unaltered. My objective is to establish a direct RTP stream between peers, bypassing the need for routing through Asterisk. Both the client and the Asterisk server are situated on the same network, and both clients utilize a PJSIP extension.
There are no steps as it is not supported. Each call leg is independent and Asterisk acts as a B2BUA. If you don’t want the audio side, then a SIP proxy is generally a better fit. There is the direct media support which will re-INVITE after call setup if conditions are right to allow direct media flow.
This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.