We never did get to the bottom of the passing of audio. If you pass 5k calls at ulaw that would be about 3.2 gigabit of bandwidth. If I recall I heard something about using app_echo in your testing. If you recall app_echo doesn't send any audio unless it gets audio. Thus you can signal 5k calls but don't expect to pass audio. And does sipps support passing of audio? I don't recall it doing so. I know we spoke at von and we would like to know the low down dirty details of how your tests are done. Maybe a step by step "how-we-did-it.txt" to really show it off so we can get some objective input. I'll hold back my opinion till we see some doc and info to back it up.
Thanks, /b
On Mar 27, 2005, at 9:32 PM, Paul Mahler wrote:
5000 Users at once on Asterisk? There is no way in hell that will happen
even if everyone is using the same codex.
Steve
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Sunday, March 27, 2005 1:52 PM To: [email protected] Subject: Re: [Asterisk-biz] Forklift a 2000 phone PBX - 5000 calls up
We sell an asterisk based solution that starts at 5000 simultaneous SIP sessions on the first 2u server and goes up from there. Way up.
Paul Mahler [EMAIL PROTECTED]
Sorry guy, we are getting upwards of 5000 SIP sessions on our starter 2U baseOn Thursday 24 March 2005 18:42, Michael Welter wrote:
I'm staring at an RFP--this company wants to replace a 2000 position PBX
unit. Our current problem is that we are using SIPP the open source benchmark,
but it keeps breaking when we go over 5000 simultaneous calls. I'm not sure how
many calls we can really get, but we'll keep trying until we find out. But, if
you need more calls than that, we can keep adding base units to the bus. The
bus is about 50 Gbits. Plenty of room to keep growing. :-)
Paul Mahler www.signate.com _______________________________________________ Asterisk-Biz mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-biz
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