I have a few questions and would appreciate some assistance:
We have a T1 between a router in the Philippine and US.
Using G723 Codec, how many concurrent calls can be supported over this T1, not looking for theoretical answer, more like practical real life answer.
Since this is a "direct T1" (The Telco told us so!) from there to here, any "compression" "queuing" "QOS" that I need to do or would like to do to improve call quality? I'm thinking maybe doing cRTP on the routers to improve bandwidth.
Now I coulda sworn you said they said the t1 line was:
Which means it definitely is NOT a direct line. It goes over ATM. If it goes over ATM it ain't a direct line. The two are mutually exclusive.
Now as far as tweaking goes, what is best depends heavily on
1) what type of VOIP are you running? (ie: h323, sip. etc) and
2) what type of routers are on both ends of the T1? What type of QOS do they support? Diffserv, 802.1q priority? Basically you want to set up the routers to look for characteristics in the packets that will identify them as VOIP traffic (ie: upd packets to or from specific ports)and make them higher priority, and put them into a faster queue. How this is done varies greatly depending on the type of VOIP and type of router.
BUT, if you are still getting > 250ms ping times over the T1, your call quality is going to be pretty crappy. Some of it may be helped if you crank up the jitter biffers on the VOIP system to compensate. But with a delay of a quarter second, users WILL notice a lag in the converstation.
QUESTION: when the providor tells you the T1 is a direct line, do they mean direct line to the end site at the US? Or direct to the internet or some other endpoint? what you want is direct to the end site, no ATM in between.
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