For estimating required bandwith, chek out:
asteriskguru.com/tools/bandw … ulator.php
Having the speech data (aka the Media) go through the asterisk, or straight between the UA’s is a choice.
Nine times out of ten you want to be in control of the call (including the disconnect of a call) so you would set your sip extensions with
This means that both the signalling as well as the media will go through the Asterisk.
Then it becomes a matter of how much codec transcoding takes place (which eats up the cpu).
This happens if extension “A” uses a different codec than extension “B” when they are talking to eachother. The Asterisk in the middle has to translate between these codecs.
You would have to experiment some. I’d say that a Xeon processor Asterisk should be able to handle something like 250 simultaneous calls (with media going through the Asterisk).