Archived from groups: comp.dcom.voice-over-ip,comp.dcom.telecom.tech (More info?)

Let's say that I would like to have the best possible sound quality in
an Asterisk-based IVR system. What audio format (gsm, ulaw, etc.)
should I use? What sampling rate and sample size?

I always used 8000 Hz and 16 bit amplitude, but this was on non-VoIP
Dialogic boards. I assumed that 8 KHz was the maximum that the POTS
system could handle, but then I found in the Windows Sound Recorder a
mode called "Telephone Quality" which uses 11.025 KHz and 8 bits only.

Comments?

-Ramon

Archived from groups: comp.dcom.voice-over-ip,comp.dcom.telecom.tech (More info?)

In article <1120712168.177070.8400@g14g2000cwa.googlegroups.com>,
Ramon F Herrera <ramon@conexus.net> wrote:
>Let's say that I would like to have the best possible sound quality in
>an Asterisk-based IVR system. What audio format (gsm, ulaw, etc.)
>should I use? What sampling rate and sample size?
>
>I always used 8000 Hz and 16 bit amplitude, but this was on non-VoIP
>Dialogic boards. I assumed that 8 KHz was the maximum that the POTS
>system could handle, but then I found in the Windows Sound Recorder a
>mode called "Telephone Quality" which uses 11.025 KHz and 8 bits only.

real telco digital data uses 8000 8-bit samples/sec. (64kbit/sec.)

There's some (but not a whole lot) excuse for going higher sampling and/or
higher bit depth, so as to avoid 'off by one' errors, due to slight
timing and/or quantization differences between your gear, and the telco's.

For "best quality", use an uncompressed audio format. For voice, there
is _very_little_ "noticable" loss, using a 'lossy' compression format
at 32-kbit/sec.

Archived from groups: comp.dcom.voice-over-ip,comp.dcom.telecom.tech (More info?)

In article <11cqs7is2ukff7e@corp.supernews.com>,
Robert Bonomi <bonomi@host122.r-bonomi.com> wrote:
>In article <1120712168.177070.8400@g14g2000cwa.googlegroups.com>,
>Ramon F Herrera <ramon@conexus.net> wrote:
>>Let's say that I would like to have the best possible sound quality in
>>an Asterisk-based IVR system. What audio format (gsm, ulaw, etc.)
>>should I use? What sampling rate and sample size?
>>
>>I always used 8000 Hz and 16 bit amplitude, but this was on non-VoIP
>>Dialogic boards. I assumed that 8 KHz was the maximum that the POTS
>>system could handle, but then I found in the Windows Sound Recorder a
>>mode called "Telephone Quality" which uses 11.025 KHz and 8 bits only.
>
>real telco digital data uses 8000 8-bit samples/sec. (64kbit/sec.)

No. They sample at 8 KHz, but use a dynamic range compression called
A-law (or uLaw, if in rightopondia. Very similar, but not exactly like.).
The sampling needs to be at 13-14 bits, and it is advantagous to
oversample by one bit. The result transmitted over the wire is a
stream of 8-bit semilogarithmic values.

GSM samples at the same frequency and at 13 bits, and applies a
compression algorithm to the data, yielding a data stream of
packets; nominally at 12 kilobits. It sounds "mobile phone".

These encodings are reasonably good at handling small amounts of
data loss.

iLBC and G.729 (proprietary) use similar principles as GSM, but
are not as bandwidth-constraimed, at various points in the 20-40
kilobit range. iLBC uses a lot, G.729 a great deal of processing
power in both ends, but sound pretty good.

ADPCM and G.726 use adaptive PCM encodings, taking the idea of
dynamic compression in A-Law even further. They are not as
processing-intensive, but sound "blander" than G.729 and iLBC.

G.723.1 is the last one I'll cover; a pure vocoder, representing
coded phonemes instead of voice. Extremely good at getting sort
of intelligable speech through at 5.3 kilobits; but sounds VERY
much like D.Duck. Also proprietary.

>There's some (but not a whole lot) excuse for going higher sampling and/or
>higher bit depth, so as to avoid 'off by one' errors, due to slight
>timing and/or quantization differences between your gear, and the telco's.
>
>For "best quality", use an uncompressed audio format. For voice, there
>is _very_little_ "noticable" loss, using a 'lossy' compression format
>at 32-kbit/sec.

If you are using asterisk throughout, have a good network (but not
unlimited bandwidth) and have fast servers for asterisk I would go
for iLBC.

If you have a rotten network use GSM and large jitter buffers.

If you have unlimited bandwidth use G.711. (Alaw or uLaw depending
on continent).

Some phones and adapters may lead you to use G.726 or Speex.

-- mrr