Divert RTP load

Asterisk Asterisk General
1

Hi,

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,

2

directmedia=yes in combination with peers that support it and a dialplan that doesn’t require Asterisk to respond to dialed digits or record the call.

Most external service providers do not support it and it gets complicated if you mix routable and non-routable networks.