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

Wed Apr 20 21:48:12 MST 2005

If you need a stable VoIP system I can recommend you just like LTenorio 
to use external VoIP gateways. We are using VoIP equipment (E1 gateways) 
from Audiocodes - the hardware is VERY stable and the prices are much 
better then Cisco.

If I was you I would make it on this way.

Lubo

---------
AppRadius Project: Full RADIUS AAA support for Asterisk PBX
http://appradius.minitelecom.org/
---------

Leandro Tenorio wrote:
> 	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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