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Eliminate delay SIP-SIP

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)
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Silas2
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Silas2
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José MéndezCommented:
You may want to read about the canreinvite peer option:

http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite

Besides, take into account that some Dial() options such as r,R,t,T will force audio to pass by the Asterisk box.
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Silas2Author Commented:
Thanks for that, is has definately 'cut out the middleman' (i set the reinvite=yes as well, i don't know if that's related,i.e. forces a reinvite where pos) but didn't change the dial option. It has cut the delay.
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José MéndezCommented:
So is it working as you wanted now?
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Silas2Author Commented:
Yes, thanks.
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José MéndezCommented:
Welcome
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