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

Olle E. Johansson oej at edvina.net
Mon Dec 8 14:28:05 MST 2003


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




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