[Asterisk-Users] SIP (peer to peer?)

[Nicolas Bougues]

On Mon, Dec 08, 2003 at 10:20:43PM +0100, Brancaleoni Matteo wrote:
> SIP control messages goes always through the server
> (port 5060) , only RTP media streams is p2p .
> 
> you can see RTP passing not p2p but by * server if:
> * the phone doesn't supports reinvites
> or
> * set in sip.conf canreinvite=no in the user definition
> 

Or of course, if Asterisk thinks that it needs to process the stream :
for instance, if you want Asterisk to be able to transfer your call
(t/T options for Dial).

-- 
Nicolas Bougues
Axialys Interactive