[Asterisk-biz] Forklift a 2000 phone PBX - 5000 calls up - WAY up

[Zoa]

Strange thing, those gigabit nics for x86.
Strange thing those linux routers capable of handling a lot of packets.
Strange things those 2000 simultaneous calls on rtpproxy on x86.

Dont blame it on the hardware, blame it on the sysadmin if your dual
xeon can only handle 80mbit/s on its network card.

A xeon has 6,4 gb bandwidth, which would mean that if we followed your
logics, we could roughly do 5000 calls using g729 codec on a machine
with similar hardware. (as the g729 uses less bandwidth than the g711,
you are not encoding anyway).

Didnt you say last time you would publish the actual test setup 'in a
week' ? Still waiting here.

I cannot check your claims of 5000 calls as i dont have an SGI, but
everything i can check in your press report sounds like complete *...
Gee i wonder why im a non believer for those 5000 calls.

Give us the details of the test and make me happy by proving me wrong.

Zoa.

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
>>
>>
>
>A codex is a "a manuscript volume, especially of a classic work or of the
>Scriptures." A codec, on the other hand, is a compressor decompressor. So, is
>this a technical thread or a religious thread?  ;-)
>
>By our calculations, 5000 SIP channels = 10000 ulaw stream @ 80k  stream = .8
>gigabit. Our total available bus bandwidth is arount 56 Gbps, as compared to
>the pc with about 128 Mbps. We can add a bunch of NICs, so blowing past 1Gb
>isn't really a problem.
>
>We ran the benchmark with app_milliwatt which sends the audio on one side of
>the call. SIP by default will echo back anything it recieves on rtp ports which
>will completes the round trip.
>
>Anyway, whatever it is we are testing, we get over 5000 of them before the MOS
>starts to degrade. This is compared to 300 of them on a PC based machine with
>comparable hardware before call quality degrades.
>
>So yes, we are getting 5000 calls on one base unit, and we can cluster more
>base units on the same bus for a LOT of calls.
>
>
>
>-----Original Message-----
>From: asterisk-biz-bounces at lists.digium.com
>[mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of Paul Mahler
>Sent: Sunday, March 27, 2005 1:52 PM
>To: asterisk-biz at lists.digium.com
>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 Mahle
>pmahler at signate.com
>
>
>
>>On Thursday 24 March 2005 18:42, Michael Welter wrote:
>>
>>
>>
>>>I'm staring at an RFP--this company wants to replace a 2000 position
>>>PBX
>>>
>>>
>Sorry guy, we are getting upwards of 5000 SIP sessions on our starter 2U base
>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
>Asterisk-Biz at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-biz
>
>--
>No virus found in this incoming message.
>Checked by AVG Anti-Virus.
>Version: 7.0.308 / Virus Database: 266.8.4 - Release Date: 3/27/2005
>
>
>
>Paul Mahler
>www.signate.com
>
>
>--------------------------------------------------------------------------------
>
>
>Previous message: [Asterisk-biz] [ANN] DialExtension - Macintosh Desktop
>Application for Dialing Extensions
>Next message: [Asterisk-biz] Forklift a 2000 phone PBX - 5000 calls up - WAY up
>
>Messages sorted by: [ date ] [ thread ] [ subject ] [ author ]
>
>--------------------------------------------------------------------------------
>More information about the Asterisk-Biz mailing list
>
>
>Paul Mahler
>www.signate.com
>_______________________________________________
>Asterisk-Biz mailing list
>Asterisk-Biz at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-biz
>
>

-------------- next part --------------
A non-text attachment was scrubbed...
Name: signature.asc
Type: application/pgp-signature
Size: 254 bytes
Desc: OpenPGP digital signature
Url : http://lists.digium.com/pipermail/asterisk-biz/attachments/20050417/44d81760/signature.pgp