What are the steps to set up Asterisk in a way that ensures the SDP remains unaltered?

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.

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