Can anyone tell me how to set up Asterisk (or Trixbox) to allow the RTP stream to be set up between User Agents instead of being directed through the server. I realize that features like recording won’t work, but I have a bottleneck issue on a project and want to avoid running the RTP streams through the server.

this will happen by default if you do not set canreinvite=no. you can further encourage it to happen with canreinvite=yes set on a per-peer/friend basis which will override the general setting.

hope that helps!


you can confirm the flow of RTP with

‘rtp debug’

on the CLI