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

[Brancaleoni Matteo]

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

Matteo.

Il lun, 2003-12-08 alle 22:17, Wim Venneman ha scritto:
>  Hi all,
> 
> Has anyone have an idea why, if you capture the files on a Asterisk
> network (ex with Ethereal) you always see the communication between
> the two sip phones( hard or soft) passing through the asterisk server
> (on UDP layer) 
> 
> Isn't SIP a protocol that (after that it has established the call) ,
> he connects the two users with each other?
> 
>  
> 
> Maybe a stupid question, but I'm not a SIP expert.
> 
>  
> 
> Thank you for your help.
> 
>  
> 
> Wim
-- 
Brancaleoni Matteo <mbrancaleoni at espia.it>
Espia - Emmegi Srl