[Asterisk-Users] 5,000 concurrent calls system rollout question
Wai Wu
wwu at Calltrol.com
Mon Jan 30 07:01:46 MST 2006
64kb/s channel will require a bandwidth of about 90kb/s using ONE voice packet per UDP packet. The overhead is a bit low if you put TWO voice packets in ONE UDP packet.
check this out.
http://web1.egvrn.net/tokata/VoIP%20Bandwidth%20Consumption.pdf
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Kristian
Larsson
Sent: Monday, January 30, 2006 10:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout
question
On Mon, Jan 30, 2006 at 12:49:15PM +0100, janvb at caselaboratories.com wrote:
>
> >Using G711A (ie, "worst case" bandwidth wise):
> >it's 64kbit/s not 64Kbyte/s
> >so it's 320Megabits per seconds
> >
> >
> That will only do if you talk a lot with your mother in law! ;-)
>
> For the rest of the conversation (those with both speaking):
>
> 5000 * 64k * 2 = 640M
Indeed you are correct, I'll defend myself with
stating that I presumed we were talkin "full duplex" ;)
>
> It should in theory work with a 1Gbits Ethernet, but you would be
> counting on ca 65% utilization. I would normally plan with < 30-40 %
> utilization and you need 2 for redundancy anyway.
Though now you're wrong ;)
65% isn't correct. If you're counting both in and
out traffic you'll have to assume that the Gigg
card is capable of 1Gbps in each direction thus
2Gbps in total and 640M of 2000G is about 30% or
just as much as 320M is of 1G.
I don't know the average packet size of a voice
RTP packet but I guess it's quite small. Being a
network guy I've dealt quite a lot with software
routers and a normal Linux machine can forward
about 500kpps, and this is mere forwarding if you
run this via Asterisk you should probably split
that by ten.
--
Kristian Larsson, Net At Once AB
Email: kristian at netatonce.se
Phone: +46 470 592717
Cell: +46 704 910401
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