SM > Sorry, should have been a little more specific. I've had
Asterisk running realtime SIP users/peers and
SM > realtime sql calls from the dialplan (all with MySQL), and
have had around 2.5k registered users and a
SM > peak (that I recall) of around 500 concurrent calls.
Wow that sounds pretty neat. Could you let us know what the HW
specs were?
The tests we've done shows that asterisk doing RTP bridging SIP/SIP
calls can handle up to approxmately 4-500 calls for a single Xeon 3.0
before locking up, spending approx 60-70% system/kernel time, _not_
usertime. We have not measured when audio quality starts to suffer,
but I would guess that happens around 300 or so. If you're allowed to
use reinvites (not having clients behind NAT and so on), the number
obviously climbes.
Note: NO you can NOT use reinvites for clients behind NAT in my
scenario: Several trunks/pstn-gateways talking SIP to a hub server
talking to clients. Clients register with hub server. pstngw gets a
call in, sends it to hub server, hub server sends reinvite to pstngw,
pstngw sends invite to client whose NAT gateway does not know the
pstngw's address and throws the packet away...
roy
---
"Humans mostly aren't particularly evil. They just get carried away
by new ideas, like dressing up in jackboots and shooting people, or
dressing up in white sheets and lynching people, or dressing up in
tie-dye jeans and playing guitars at people"
- Terry Pratchett
-------------------------------
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]
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