[Asterisk-biz] Large Asterisk Setup (~500 Concurrent Calls+Scalability)

[list]

I agree w/ you 100%. Let cisco or quintum do all the transcoding, it will 
save you a lot of headaches and countless hours of downtime.


----- Original Message ----- 
From: "Leandro Tenorio" <leandro_tenorio at ciudad.com.ar>
To: "'Commercial and Business-Oriented Asterisk Discussion'" 
<asterisk-biz at lists.digium.com>
Sent: Wednesday, April 20, 2005 4:52 PM
Subject: RE: [Asterisk-biz] Large Asterisk Setup (~500 Concurrent 
Calls+Scalability)


> I'm sure that most of the Asterisk users in the list will not agree
> we me, but in such large implementations, I prefeer to use a quite 
> different
> approach.
> In this kind of implementations I prefeer to use any TDM-VoIP
> gateway (Quintum / Cisco / etc, there are a lot of them, even used 
> equipment
> for less than $6k, new ones cost like $12k with 4T1s/E1s) and then use
> asterisk for extentions and all the services you want to provide.
> why, It's far more stable, easy to configure/maintain, and less pof
> in a specific designed equipment than a server.
> I also suggest you to put two asterisk servers for services, user
> registration, etc.
>
> LTenorio
>
>
>
> -----Original Message-----
> From: asterisk-biz-bounces at lists.digium.com
> [mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of Matt Roth
> Sent: Wednesday, April 20, 2005 5:14 PM
> To: Commercial and Business-Oriented Asterisk Discussion
> Subject: [Asterisk-biz] Large Asterisk Setup (~500 Concurrent Calls
> +Scalability)
>
> List Members,
>
> I am involved in the process of designing a large Asterisk setup for a 
> call
> center.  A graphical overview of our tentative design can be found
> here:
>
> http://home.comcast.net/~mroth01/LargeAsteriskSetup.gif
>
> Originally, we planned to implement this design by purchasing one
> multi-processor machine and putting multiple quad-span T1 cards (Wildcard
> TE4xxPs) into it.  Through research, it was determined that the PCI bus
> couldn't handle the digital signal processing (DSP) from more than one
> quad-span card.
>
> The goal of our new design is to offload the DSP to the Asterisk slave
> servers, then route the calls via IAX2 trunks to the Asterisk master 
> server.
> The Asterisk master server will provide us with a centralized point for
> queuing, digital recording, and music on hold, as well as configuration,
> monitoring, and reporting.  Configuration of the Asterisk slave servers
> would be limited to setting up extensions to terminate the incoming T1s 
> and
> setting up IAX2 trunks to the Asterisk master server.
> These configurations would be rare, so the slave servers would be 
> configured
> manually on the boxes themselves.
>
> Failover of the primary slave servers will be provided by backup slave
> servers configured to mirror one or more of the primary slave servers'
> extensions and IAX2 trunks.  The master server will be mirrored as well. 
> On
> failure, automatic T1 switching is an option, but we would initially be
> doing it manually.
>
> Scalability is provided by adding machines to the slave server pool, up to
> the point where the master server can no longer handle the call volume.
>
> An example of a typical incoming call's flow follows:
> - The call originates from the PSTN and reaches an inbound Asterisk slave
> server via an inbound T1.
> - The Asterisk slave server handles DSP and routes the call to the 
> Asterisk
> master server via an IAX2 trunk.
> - The Asterisk master server handles queuing the call and eventually 
> routes
> it to a SIP phone via a SIP channel.
>
> An example of a typical outgoing call's flow follows:
> - The call originates from a SIP phone and reaches the Asterisk master
> server via a SIP channel.
> - The Asterisk master server routes the call to an outbound Asterisk slave
> server via an IAX2 trunk.
> - The outbound Asterisk slave server handles DSP and passes the call off 
> to
> the PSTN via an outbound T1.
>
> Note that the master server must handle protocol bridging between IAX2 and
> SIP, but will not have to do any transcoding because we can control the
> codecs used on the servers and the SIP phones.  Digital recording as well 
> as
> monitoring, reporting, and configuration tasks will be offloaded to client
> machines via mounted drives and the Asterisk Management API in order to
> lessen the burden on the Asterisk master server.  The Asterisk master 
> server
> will also be responsible for opening a socket connection to an agent 
> station
> on each incoming call in order to pass the phone number that the call came
> in on.
>
> We have done a lot of research and were unable to find any documented 
> cases
> of a centralized design of this scale.  This is our preliminary design and
> is apt to have a few holes, mistakes, and possibly deal-breaking 
> oversights.
> Please provide any opinions that you have on the overall feasibility of 
> this
> design as well as any hardware recommendations for each of the components 
> or
> suggestions for improving the overall scheme.  If you see any bottlenecks 
> we
> have overlooked, please point them out and give any suggestions for
> circumventing them.
> Any ideas on how large this system could be scaled would also be
> appreciated.
>
> I also have a question regarding DSP: Does outbound DSP (digital to
> analog) require less processing than inbound DSP (analog to digital), and 
> if
> so by what ratio?
>
> There is a spot on the wiki
> (http://www.voip-info.org/wiki-Asterisk%20hardware%20recommendations)
> regarding this size Asterisk setup, but it has not been addressed yet.
> Hopefully, this will be the start of filling in that hole.
>
> Thank you for your time,
>
> Matthew Roth
> http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian
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