[Asterisk-Users] SIP (peer to peer?)
kpj at junghanns.net
Mon Dec 8 14:54:15 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.
or if the canreinvite= parsing in chan_sip.c is broken...
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