Wiley Siler
2005-Apr-20 19:15 UTC
[Asterisk-Users] Large Asterisk Setup (~500 Concurrent Calls +Scalability)
Interesting setup. Love to hear more about it as you get it done. Regarding your DSP/PCI Bus.... I think that I saw that Sangoma cards may not have the same problem. Something to check out at least. Also this card looks impressive if it ever materializes. Think it is still vapor though. Anyone got these yet? http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-38356249088.htm DSP onboard! Luck! Wiley ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Matt Roth Sent: Wed 4/20/2005 1:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 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 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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: application/ms-tnef Size: 7883 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050420/d4577360/attachment.bin
Iassen Hristov
2005-Apr-21 13:50 UTC
[Asterisk-Users] Re: Large Asterisk Setup (~500 Concurrent Calls +Scalability)
I suppose you know about the upcoming DS3000P <http://www.digium.com/index.php?menu=news/ds3000p> Supposedly it can handle 28 T1's, so I assume the DSP is done onboard. Maybe you should give Digium a call. P.S. Sorry for not threading, but I get the digest and have not figured out how to inject the original message-id header.> From: asterisk-users-bounces@lists.digium.com on behalf of Matt Roth > Sent: Wed 4/20/2005 1:10 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] 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.