[asterisk-users] SIP RTP flow

[Mike Williams]

Hey,

This is probably a rather stilly question...

If I pick up my SIP phone that's registered to my asterisk server and dial a 
number that asterisk recognises as destined for a SIP trunk (could be a 
static route, or an ENUM lookup) or another SIP device registered on said 
asterisk server (internal extension to extension call), what route does the 
actual audio take?

The control connection (port 5060) obviously goes via the asterisk server as 
it has to work out where to send the control to, but I could quite easily 
imagine the audio going directly handset to remote server or handset to 
asterisk to remote, and handset to handset or handset to asterisk to handset.

Thanks

-- 
Mike Williams