[Asterisk-Users] Re: Re: T1/DS1/ISDN PRI

Matt Roth mroth at imminc.com
Fri Apr 29 10:27:59 MST 2005


David Josephson,

> Not off-base, but you haven't made it all the way home yet. This is 
> another layer of the puzzle, and again we are not talking about apples 
> and apples here. "Circuit switched" means that there is a (real or 
> virtual) circuit that takes data on an input port and delivers it to 
> an output port somewhere. "Packet switched" means that each packet of 
> data is examined by each port it passes, to see where it should be 
> sent. Normally this layer of VoIP traffic is handled not in Asterisk, 
> but in a router. You could run the router on the same Linux box that's 
> running Asterisk (and send packets to different Ethernet ports 
> depending on their destination address) but normally this task is 
> handled by a separate router. There is a small computational overhead 
> associated with adding and decoding Ethernet packets but the main 
> routing work is done outside Asterisk, and isn't too intensive. You 
> could read up on TCP/IP routing and understand how this works in more 
> detail.

We plan on using a Gb switch with 100 Mbps ports to handle the routing.

> It's not something you can "take a look at" in my experience. Some of 
> the Bell System training material that comes up on eBay is good. You 
> need to follow the progress from circuit-switched voice telephony 
> circa 1930 through modern TDM, and then look at the development of 
> TCP/IP switching separately.

75 years of telephony and network technology to cover, eh?  Looks like 
it's going to be a long weekend.  ; )

> No sound card, no monitor. Recording to the various file formats is 
> possible, as Herman mentioned.

This seems like an odd limitation to me.  Any idea why it's designed so 
that you must have a sound card to digitally record calls?  They could 
always be moved to another box in order to listen to them.

> Your reference picture is fine ... but note that Asterisk can be the 
> TDM/VoIP gateway, particularly when Digium releases their DS3 card 
> (644 voice channels!) working, a lot more cheaply than a standalone 
> box from some hardware vendor.

I'm not sure that the DS3000P is in our timeframe.  I am interested in 
knowing how it will perform, considering more than two Digium quad-span 
cards currently overload the CPU with interrupts.  It seems that Monitor 
cannot handle digitally recording more than ~50 concurrent calls, 
either.  Maybe these limitations are being addressed as we speak.

Thank you for sharing your knowledge with me,

Matthew Roth
http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian



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