[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 Patrick
> Sent: Sunday, January 29, 2006 7:40 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] 5,000 concurrent calls system rollout
> question
> 
> On Sun, 2006-01-29 at 17:24 +1000, Rob Thomas wrote:
> [snip]
> > If you do, honestly, need to handle 5k calls, you'd probably have to
> > have a bank of Cisco 5850s doing the termination
> 
> Or have a look at the Lucent APX8100 box for some added carrier class
> humpf. Supports more than 8000 DS0's (channels) and does transcoding
in
> hardware DSP's so well suited to handle your 5000 concurrent calls and
> you don't need a stack of them like with the Cisco 5850.
> 
> Weblink:
>
http://www.lucent.com/products/subcategory/0,,CTID+2013-STID+10443-LOCL
> +1,00.html
> Datasheet:
> http://www.lucent.com/livelink/090094038000eb91_Brochure_datasheet.pdf
> 
> Like Rob I'd love to sell this to you but I doubt Lucent would even
pick
> up the phone to answer my "how to become a VAR" enquiry. Best contact
> them directly :)
> 
> Regards,
> Patrick (no affiliation with Lucent)
> 
The original poster of this message stated in an earlier message that
the calls would be handed off to him SIP, so the media conversion is
being done buy an upstream carrier, presumably on a Lucent or Sonus.

With the growing availability of SIP origination and termination, high
density channels banks like the APX8000 are becoming items only needed
by wholesale carriers.

Of course this varies by geographical region, but to use a APX 8000 you
need at least PRI service over DS1/E1, and ideally PRI service over
DS3/E3.

The challenge I see with a 5000 INBOUND call setup originated SIP is
that the calls will need to be load balanced across many * boxes, no 1
asterisk box is going to take 5000 CONCURRENT calls (500 would impress
me).

I would suggest; 

Check to see if the SIP origination provider can give you a "round
robin" delivery of calls over 10 or so * boxes (IP addresses), or find
an external method of doing it yourself (like a smart session border
controller).

IF the calls are terminated to hardphones or softphones (as opposed to
purely IVR), make sure you can do RTP re-invites so, when appropriate,
the RTP stream is offloaded from * (but consider the impact of doing
so).

Calculate bandwidth needs carefully - 5000 * 70-75kbps  (a/ulaw plus
packet overhead) requires a GIG-E IP link from you SIP provider and some
very robust networking in between.

Terminating 5000 calls on * is relatively uncharted ground, there MAY be
some others doing it, but good luck getting them to reveal the company
jewels.

At the very least, this type of implementation would require a team of
the VERY BEST asterisk consultants - might want to call Mark himself if
you are serious.

Damon