[asterisk-users] When does Scalability requests Asterisk to U
se SER ?
rushowr at phreaker.net
Wed Sep 20 11:10:16 MST 2006
Kristian Kielhofner wrote:
> Rushowr wrote:
>>> S McGowan,
>>> I don't know if you missed my question (from the slew of questions
>>> received and answered), but I was wondering about transcoding and PSTN
>>> channels. What kind of codecs were used and was there any transcoding
>>> happening? Was this box only responsible for VoIP-to-VoIP calls or was
>>> there also PSTN trunks as well? Again, I'm amazed by this example
>>> since it
>>> seems to be way over what anyone else normally reports as usable.
>>> Thanks again,
>> Ryan, I answered, but for some reason this pop account tends to be
>> strange... Anyway, we were not doing any transcoding and our PSTN
>> connectivity was handled via a Tier 1 ISP that does SIP only PSTN
>> connectivity solutions with G.711u. So, basically as far as Asterisk was
>> concerned, there was SIP and RDP, that's all.
> So there was 2500 SIP registrations with qualify, 500 active calls
> with SIP and RTP, realtime, and CDR logging via MySQL (all on the same
> What source changes did you make? What OS tweaks?
> Kristian Kielhofner
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None, literally. CentOS 4.3, Asterisk Trunk that was updated practically
weekly, at least on the dev box. The production server wouldn't get
recompiled unless a fix was in from trunk.
Incidentally, I just doublechecked my numbers with my former co-worker.
He confirms we had roughly the following numbers/setup:
*2,500 registered SIP users, 95% being qualified by Asterisk
*Max of 300 concurrent calls, with about half that on average (my
mistake earlier with the 500 estimation)
*Realtime SIP Users/Peers, Voicemail, and dialplan calls
*Static extensions with MySQL queries for data retrieval/manipulation
*NO Reinvites allowed due to the fact that most clients were residential
CPU: Dual 3Ghz XEON
RAM: 2GB RAM
rushowr at phreaker.net
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