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

[Daniel Salama]

Nice setup. Please do feed the list with your findings.

I'm interested in a solution for digitally recording calls and I  
noticed you have a Digital Recording Client. Would you mind elaborating  
more on this? Is this another Asterisk box issuing the Monitor command?  
If it is, does it mean that even is Asterisk is "monitoring" a call on  
another Asterisk box, then the box with the call being monitored will  
not suffer any load overhead for Monitoring? I guess what I mean is  
that, will that offload the Asterisk box handling the actual call?

Thanks,
Daniel

On Apr 20, 2005, at 4:10 PM, Matt Roth wrote:

> 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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