[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_


More information about the asterisk-users mailing list