I have try to make asterisk call direct client-to-client, but when I capture packets, rtp packet still go through Asterisk Server.
in sip.conf I have “directmedia=yes” and “directrtpsetup=yes”. any suggestion, how to make rtp packet direct client-to-client?
I read sip.conf from asterisk Developer Documentation, on that document wrote, “Asterisk by default tries to redirect the RTP media stream to go directly from the caller to the callee”. so, from this side, RTP packet will send direct caller-to-callee by default, is it right? but in my test, it is not.
should I modify other configuration?