[asterisk-users] When does Scalability requests Asterisk

[Roy Sigurd Karlsbakk]

> 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
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Roy Sigurd Karlsbakk
roy at karlsbakk.net