[asterisk-users] When does Scalability requests Asterisk
Jay R. Ashworth
jra at baylink.com
Tue Sep 19 10:24:02 MST 2006
On Tue, Sep 19, 2006 at 07:01:41PM +0200, Roy Sigurd Karlsbakk wrote:
> 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.
Newbie question: that's if all the audio is passing over the server's
bus, right?
I'm looking at a pretty big system using either SIP or MGCP to tell a
bunch of FXS and T-1 media gateway boxes to talk to each other over a
dedicated GigE -- would that *be* a reinvite situation, generally, or
not? I'm assuming that since the server would only be doing MoH, VRX,
and the like, that I'm in much better shape loadwise, even at 40xSIP +
390xFXS.
Cheers,
-- jra
--
Jay R. Ashworth jra at baylink.com
Designer Baylink RFC 2100
Ashworth & Associates The Things I Think '87 e24
St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274
"That's women for you; you divorce them, and 10 years later,
they stop having sex with you." -- Jennifer Crusie; _Fast_Women_
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