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 <[EMAIL PROTECTED]> Espia - Emmegi Srl _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
