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

[Matt Roth]

David,

> What someone is missing, is that TDM and VoIP aren't "converted." TDM 
> PRI's include signaling in the same bitstream. VoIP uses separate data 
> paths for signaling and voice. The voice data can be the same, or 
> different.

It is my understanding that TDM is circuit switched and VoIP is packet 
switched.  It would seem to me that at some point in a TDM-VoIP gateway, 
a change from circuit switching to packet switching is happening, and 
vice versa depending on the direction of the signals.  I was just 
wondering if anyone could detail that process and tell me if it is 
resource-intensive.  If I'm completely off-base, please point me in the 
right direction.

> Agreed. But not to him. T1 refers to the line coding on 2 physical 
> pairs of wire to encode and carry a 1.544 Mbps datastream. T2 is four 
> of those signals multiplexed onto 2 pairs of wire. T3 is 28 DS1's (7 
> DS2's) multiplexed onto two coaxial cables. DS1 is a logical concept 
> that defines what to do with that signal, what the bits mean; each DS1 
> is made up of 24 DS0 time slots. An ISDN PRI is the definition that 
> one uses 23 of the time slots for 23 voice channels, plus one DS0 
> dedicated to signaling. A DS3 is 28 DS1's (and is usually carried on a 
> T3 physical layer, does it begin to make sense?)

Yes, thank you.

> It's too bad. A lot of people without any telephone background try to 
> make up stuff using pieces of the old terminology and wonder why they 
> stay confused. They could look it up, but they don't. For instance 
> DID's. DID has a specific meaning and inward service from the PSTN 
> handed off on VOIP isn't it.

Do you have a good, reliable source that I could take a look at?

>>> - 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)?
>>>   
>>
>> I *believe* that it will write the data in G.711 format. Don't rely 
>> on this though.
>>  
>>
> No. It writes data to whatever format the sound card supports, usually 
> 16 bit linear (raw) which becomes .wav if you add file headers to it.

Is there a way to specify the format?  What if there is no sound card on 
the Asterisk server?

Thank you,

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