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 [EMAIL PROTECTED] 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_ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
