[Asterisk-Users] SIP (peer to peer?)
mbrancaleoni at espia.it
Mon Dec 8 14:20:43 MST 2003
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
* set in sip.conf canreinvite=no in the user definition
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.
Brancaleoni Matteo <mbrancaleoni at espia.it>
Espia - Emmegi Srl
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