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

[Martin Joseph]

On Jan 28, 2006, at 7:15 PM, Vic wrote:

> Hi, Zoa,
>
> yes, these calls are from SIP to SIP. We will have more than 3000  
> (more like 5000)concurrent calls come into system and we will need to  
> handle them.
>
> We will also need an IVR function as well.
>
> I am not up to speed on Asterisk yet, so, I am a little bit confused  
> by all the different ways of doing it. Someone is talking about IAX:   
> I think it can only be used between Asterisk servers, right?
You can also use it as and end use agent (ie an ATA or a phone).  I am  
using an AG-168V which is a cheapo ATA that supports IAX directly.   
This is nice because it simplifies ports and firewall issues.
>
> In this particula rscenario we are getting calls as SIP directly from  
> carrier, so we will not need to do any conversion (I think). We just  
> route the calls to the destination, that's it.
>
> Any suggestions on how to proceed? Can Asterisk do it?
>
> I read somewhere that it takes about 30 MHz per one voice channel, so  
> if we want to have 5,000 calls, we will need 150,000 MHz? Thats like  
> 50 3 GHz machines... Not going to fly with our people. 
>
> Or do 30 MHz are only necessary for transcoding? In other words, if it  
> comes in as SIP and we keep it that way, can we make it a bt more  
> feasible number? 
Transcoding is a big consumer of CPU for sure.  This has nothing to do  
with SIP however and is related to the CODEC you are using at the end  
of the line and in between.  If all you calls are coming in and being  
delivered in the same format (ie g729), then you don't need to  
transcode anything, and the CPU load is much lighter. In fact you can  
setup asterisk to make a native bridge of these calls.

Perhaps you could try building a testbed?  That's what I would do.

Good Luck,
Marty

>
>  
>
>  Zoa <zoachien at securax.org> wrote:
>>  It can be done, are those 3000 calls sip to sip ? If so it could  
>> easily
>>  be done, if they are not sip to sip you will need a bunch of servers.
>>
>>  Zoa.
>>
>>  Vic wrote:
>>
>>  > Hi,
>>  >
>>  > we are currently considering different options for rolling out a  
>> large
>>  > scale IP PBX to handle around 3,000 + concurrent calls.
>>  >
>>  > Can this be done with Asterisk? Has it been done before?
>>  >
>>  > I really would like an input on this.
>>  >
>>  > Thanks!
>>  >
>>
>> >--------------------------------------------------------------------- 
>> ---
>>  >
>>  >_______________________________________________
>>  >--Bandwidth and Colocation provided by Easynews.com --
>>  >
>>  >Asterisk-Users mailing list
>>  >To UNSUBSCRIBE or update options visit:
>>  > http://lists.digium.com/mailman/listinfo/asterisk-users
>>  >
>>  >
>>
>>  _______________________________________________
>>  --Bandwidth and Colocation provided by Easynews.com --
>>
>>  Asterisk-Users mailing list
>>  To UNSUBSCRIBE or update options visit:
>>  http://lists.digium.com/mailman/listinfo/asterisk-users
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: text/enriched
Size: 3238 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060128/f7c94fa7/attachment.bin