AK 4asterisk
2006-Sep-19 08:35 UTC
[asterisk-users] When does Scalability requests Asterisk
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? - AK -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060919/9688938a/attachment.htm
AK 4asterisk
2006-Sep-19 08:40 UTC
[asterisk-users] When does Scalability requests Asterisk
Never mind my previous question. I see that Ryan already asked that and you responded. It does sound like a pretty neat project. Would love to hear more once you finish the project that you are working on currently. - AK -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060919/30916d64/attachment.htm
Roy Sigurd Karlsbakk
2006-Sep-19 10:01 UTC
[asterisk-users] When does Scalability requests Asterisk
> 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 roy@karlsbakk.net