[Asterisk-Users] Can this config sustain 30 users?

Tom Lynn tom at tomlynn.com
Tue Jun 13 09:24:35 MST 2006


Don't forget to be sure your power supplies are reliable, and if necessary
redundant.

On 6/13/06, Colin Anderson <ColinA at landmarkmasterbuilder.com> wrote:
>
> Maybe you are over thinking it. I have 200 users on a quad Xeon 700 with 2
> PRI's and we regularly have 40-60 channels up, no problem (believe me, if
> there was a problem I'd have 200 guys freaking on my head). I rarely see >
> 30% single-CPU usage, and that's only when Sendmail is invoked to send out
> a
> voicemail.
>
> But yes, transcoding and reasonable echocancel values is key. If you are
> connecting to the PSTN, ulaw all the way. If you are connecting to a
> provider, use the codec of your choice as long as your provider supports
> it,
> and make sure every phone and endpoint is set to use the same codec.
>
> I also have 30 IAX remote sites that support from 1 to 5 users, on P-II
> 233's. I use them because they are bulletproof and they are so cheap if
> something gets hosed we just throw it away and put in another one. Again,
> no
> problem
>
> Maybe try your cheapo machine and if it doesn't work try a better box. You
> already have the cheap machine, and the card will remain the same
> regardless
> of what box you use.
>
>
> -----Original Message-----
> From: Erick Perez [mailto:eaperezh at gmail.com]
> Sent: Tuesday, June 13, 2006 9:15 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Can this config sustain 30 users?
>
>
> Well thanks all for your responses. My original intention was to
> address the mistic know-how about machine calculations, and I still
> feel the shadows remain.
>
> Why? Because to achieve a 24 user PBX-only/One E1, I was going to
> install a Dual Xeon with 2 GB of RAM and a 3ware card runing RAID-1
> with two sata3 disks.
> Now This thread tells me that my dual core pentium d (a 700$ computer)
> will do the work. (the other equipment costs about 3500.00$). I do
> realize that i must minimize transcoding (ulaw all the way) but you're
> telling me it will work for 24 users (let's say 30 for round numbers)
> all with SIP phones in an IP network.
>
> Below are some comments that i found googling and doing some
> calculations myself. I do not enforce or deny any of them, please feel
> free to tell me if Im wrong.
>
> (not confirmed)a- A voice channel takes 30mhz (60mhz in duplex mode).
> So A 2.66 Ghz CPU can sustain about 43 calls (2600mhz/60mhz=43calls),
> not taking into account other factors that may increase/decrease the
> number of calls at the same time.
>
> b- 24 users talking ulaw (-+ 80kbps per channel) consume 1920kbps and
> in full duplex they consume 3840kbps (about 3.75 megabits/s).
>
> c- To Calculate the bandwidth DDR memory can achieve (example PC4200)
> ,to get the transfer rate, multiply the width of the module (8 Bytes)
> by the rated speed of the memory module (in MHz): (8 Bytes) x (533
> MHz/second) = 4,264 Mbytes/second or 4.2 Gbytes/second (34gigabits/s),
> hence the name PC4200
>
> So, will all of this in mind,
> CPU Dual Core 533FSB, 2.66 Ghz speed
> DDR533mhz, One gigabyte. (2x512)
> Two Sata disks (each sata pumps 1.5 gigabits/s)
> Motherboard Intel 945 at 533FSB
>
> Means that the cpu,the ram and the board can achieve (see point b)
> about 34 gigabits of data transfer, but 24 users only generate 3.75
> megabits. So this is more than covered.
> However if we take into account the lowest performing component on
> this system (the sata disks) we go down to 1.5gbits/s which still
> seems to be enough.
>
> Please please correct me if im wrong (or crazy)
>
> Thanks,
>
>
> References:
> http://en.wikipedia.org/wiki/Front_side_bus (bus bandwidth table)
> http://www.acme.com/build_a_pc/bandwidth.html
> http://www.lostcircuits.com/memory/ddrii/
> http://bvio.ngic.re.kr/Bvio/index.php?title=Front_side_bus
>
>
> On 6/13/06, Mike Fedyk <mfedyk at mikefedyk.com> wrote:
> > Erick Perez wrote:
> > > I just don't want to install it and then after a 5th user going to
> > > call someone the asterisk begin to crash due to lack of resuources.
> > Check the wiki for SIP load generation tools you can use to test your
> > setup on any number of calls you like.
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> >
>
>
> --
> ------------------------------------------------------------
> Erick Perez
> Panama Sistemas
> Integradores de Telefonia IP y Soluciones Para Centros de Datos
> Panama, Republica de Panama
> Cel Panama. +(507) 6694-4780
> ------------------------------------------------------------
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