[Asterisk-Users] Re: T1/DS1/ISDN PRI
David Josephson
dlj at altaphon.com
Thu Apr 28 12:40:28 MST 2005
>
>
>>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.
>>
>>
>You are severely confused, using wrong terminology, so it is very hard for
>us to understand what exactly you are trying to say and what you are
>asking.
>
>
Now, be gentle.
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.
>>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:
>>
>>
>They are not subtle and they are very clear.
>
>
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?)
>>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)
>>
>>
>Don't try to gain knowledge from web forums, you'll only become dumber, it
>is like learning about hosting by reading WHT. I feel dumber already after
>reading those posts.
>
>
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.
>There's no difference. DS1 is a standard signaling with 1.54Mbps raw
>capacity. T1 is a product name for DS1 by your carriers. PRI is "primary
>rate ISDN" which is DS1 partitioned into 23 "Bearer" channels for calls,
>and one "Data" channel for ISUP call signaling.
>
>
Not quite. See above. T1 is the physical interface, DS1 is what you
carry on it.
>>- 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?
>>
>>
>Yes
>
>
>
>>- Are there any other host-intensive processes that I should consider
>>offloading to the gateway, such as echo cancellation?
>>
>>
>Yes, 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)?
>>
>>
>Yes, PCM u-law codec is exactly the same as G.711 ulaw.
>
>
And outside of North America and Japan, a-law is used.
>
>
>>- 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.
--
David Josephson
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