Asterisk config question

I am a new user of asterisk. I need to configure asterisk if possible to run on a low resource machine. All the clients will be local to my network. Therefore I need to know if it is possible to configure asterisk to setup the call but have the clients pass audio and video directly to each other on the LAN without passing through and being processed in any way by the asterisk server?



Yes, as long as you use SIP and don’t use any conflicting features. Although there is a theoretical capability to set up calls so they never go through the core, I’m not sure if it working properly in the latest version, however direct media is possible, after a short period at the start of the call. The length of the delay will depend on things like whether any SIP packets get lost.

Conflicting features include conferencing, voice recording, access to call features via DTMF, music on hold, queue position announcements, putting on hold, voice mail, etc. Many of these only require RTP to the core when actually in use.

PS A more useful subject would have been: Underpowered machine - RTP direct from phone to phone?

Thanks for the reply.

A follow-up question:

If I needed to use voicemail, and knowing statistically that a very small percentage of clients were at any given time using voicemail, would the client to client call traffic still bypass the server leaving only the call handling traffic and the active voicemail connections?