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

[Olle E. Johansson]

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 if the both ends have incompatible codec settings and Asterisk is able to translate.

/O