Eliminate delay SIP-SIP
Posted on 2012-08-23
Excuse my ignorance of SIP, but my understanding is that once a call is setup with SIP, the RTP can take a direct route from UA to UA, avoiding the SIP registrar. With my asterisk box I'm getting real delay with SIP-SIP calls (but not SIP-PSTN), whilst I'm recording the SIP-PSTN calls so they need to be routed thru the asterisk box, the SIP-SIP (i.e. internal) calls don't need to go via asterisk, is there a way with the dialplan to somehow cut asterisk out of the route, but still remain it as SIP registrar?
(Im just thinking its an extra 4 hops as the asterisk is at at remote/cloud location)