Is there a way to divert the RTP, and remove the server from the call “loop” once a call is established to minimize server load?
Assuming I have a “A” SIP user, “B” FreePBX server and a trunk to a “C” termination provider. In a call initiated by user A, the signaling goes through the sever, to the C termination provider. The way I understand the call process, once a call is initiate, the server can/should step-out and the RTP should run directly between the A SIP user and C provider (until hangup).
The situation we see today is that the server is involved with RTP sessions, during the whole conversation (using Hummer TCP dump).
- What trunk settings are needed to remove the RTP streaming through the server?
- Any other server global settings should be modified?
- The point is to minimize server load, would it work?
thanks ahead for all input,