Asterisk Users / Asterisk Biz List Members,
About a week ago I cross-posted a message titled "Large Asterisk Setup (~500 Concurrent Calls + Scalability)" to Asterisk-Users and Asterisk-Biz. For reference, the threads generated by that message are archived at the following locations:
http://lists.digium.com/pipermail/asterisk-users/2005-April/102823.html http://lists.digium.com/pipermail/asterisk-biz/2005-April/004590.html
First, I would like to thank you all for your excellent suggestions and contributions. My misunderstanding of the Digium quad-span card's scaling limitations was corrected (PCI bus traffic is not the problem, it is the number of interrupts generated by the Zaptel drivers) and I was directed to replace the Asterisk Slave Servers in this diagram (http://home.comcast.net/~mroth01/LargeAsteriskSetup.gif) with a VoIP gateway.
Following up on that suggestion, I began researching T-1 time division multiplexing in order to understand where the DSP load on an Asterisk server originates and the best options for purchasing a VoIP gateway to offload that processing onto. The results of that research can be found at the following links:
T-1 Multiplexing - PSTN Side (http://home.comcast.net/~mroth01/T1-PSTN.gif)
T-1 Multiplexing - CPE Side (http://home.comcast.net/~mroth01/T1-CPE.gif)
Basic T-1 Time Division Multiplexer (http://home.comcast.net/~mroth01/T1-TDM.gif)
Telephony Glossary (http://home.comcast.net/~mroth01/telephony-glossary.html)
Sources (http://home.comcast.net/~mroth01/sources.html)
My understanding of the T-1 TDM and the PSTN side is pretty solid, as it is mainly based off of Intel Corporation's T1/E1 Technology Primer (see Sources), but the CPE side is largely deduced from what I knew about the PSTN side. There may be holes or mistakes, so I would appreciate any corrections or additions that you can offer. Specifically, I would like a detail of the TDM - VoIP conversion process, similar to the basic T-1 TDM one I provided.
The differences between a T-1, DS-1, and ISDN are subtle and not universally agreed upon. For a discussion of these issues see the following links:
What's the diff between a T1 and a DS1 (http://pbxtech.info/showthread.php?t=1100)
PRI setup (http://pbxtech.info/showthread.php?t=1250)
In closing, I have a few questions:
- Is my understanding of using the same codecs and signaling protocols on both sides of the Asterisk server in order to circumvent transcoding and conversions on the server correct?
- Are there any other host-intensive processes that I should consider offloading to the gateway, such as echo cancellation?
- What does the PCM µ-law codec used in T-1 multiplexing map to in terms of Asterisk codecs (G.711 µ-law, perhaps)?
- What codec does the Monitor application use when digitally recording calls (if possible, I would like to avoid transcoding the streams when recording and let sox handle the conversions on a different box)?
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
