> 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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