[Asterisk-Users] 5,000 concurrent calls system rollout question

[Damon Estep]

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of C F
> Sent: Tuesday, January 31, 2006 9:50 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout
> question
> 
> On 1/31/06, Damon Estep <damon at suburbanbroadband.net> wrote:
> >
> >
> > > -----Original Message-----
> > > From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-
> > > bounces at lists.digium.com] On Behalf Of C F
> > > Sent: Tuesday, January 31, 2006 4:03 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users] 5,000 concurrent calls system
rollout
> > > question
> > >
> > > I don't know how much 1+1 by you is, but lets recalculate this for
a
> > > moment:
> > > First the bandwidth per channel:
> > > http://www.airewaves.com/aire/support/bandwidth_explain.php
> > > 1.5mbps (mega *BITS* not BYTES per second) to a full T1, which
equals
> > > 1536 Kbits, each channel then takes 64kbps.
> > > 64*5,000=320,000kbps.
> > > 32,000/1,024=312.5 Mbps (round off to Mbps), no where close to a
Gb.
> > > Every single PC made in the last 4 years I came across, can handle
> > > this type of bandwidth.
> > > BTW, this all amounts to just over 39 MBYTES per second.
> > 312.5/8=39.0625
> > >
> >
> > Not that I disagree with your point, the bandwidth is not huge, but
the
> > math is a little fuzzy;
> 
> No it's not, the math was meant for the T1 calculations, reread the
> post I replied to.

I guess I just assumed that that the connection to asterisk would have
to be IP since it is absolutely impossible to connect ~208 T1s directly
to a single asterisk server. You would have to use an external media
gateway. I am not aware of any 200x T1 or 8x T3 cards for asterisk :)


> 
> >
> > First of all, a g.711u stream over UDP is closer 80k than 64k, the
> > payload is 64k + udp overhead + IP overhead.
> >
> > Now consider that the call is originated as SIP (llok back a few
days in
> > the thread), and lets assume the call goes to an external hard or
> > softphone, and lets also assume that there is a reason to keep the
RTP
> > stream running through asterisk (monitoring, recording,
transferring,
> > dtmf, ability to re-enter IVR, etc).
> >
> > I make all the assumptions safely since the thread was started by
> > someone looking to set up a large call center and I have followed
thread
> > out of curiosity.
> >
> > So a 80k full duplex RTP stream originates on media gateway
somewhere,
> > hits the asterisk box, is internally bridged, and is sent back out
to a
> > phone somewhere. My math says this puts a 160kbps full duplex load
in
> > the NIC.
> >
> > Ok, now lets go for 5000 of them. 160kbps*5000=800000kbps or 800mbps
-
> > full duplex.
> >
> > Have you ever seen a NIC or switch that can run GigE full duplex at
80%
> > utilization and not at least start to fall apart?
> >
> > To get to a comfortable load you would need 2x GigE NICs (for ~40%
> > utilization), of course now we are adding additional overhead for
the
> > bonded NIC trunking protocol.
> >
> > Is still contend this is not practical without multiple very high
end
> > servers and round robin call origination from the upstream provider
> > delivered over something like GigE or OCx.
> >
> > Maybe someone will step up and post some real-world application
limits
> > based on experience...
> >
> >
> >
> >
> >
> >