asterisk users - Sep 2005 - Asterisk sound files, audio bandwidth, and sound quality

Hello, everybody:

I'm developing an application using Asterisk and a TDM-400 card.

I understand the concept of the difference between GSM and WAV files
when using Asterisk, but I'm not happy with the sound quality with the
GSM compression. It's merely *acceptable* for a telephone call, but for
anything else, it leaves something to be desired.

Case in point -- if you compare the sound quality of the prompts with,
say, an Octel voicemail system, there's no contest -- and this is
through a standard telephone.

1. Can I *record* audio from a TDM-400 channel at a sample rate above
8000 kHz? Can I record at 16 bits? (This is most important.)

2. Can I use a prompt with a sample rate above 8000 kHz? Will the
TDM-400 card support it? How can I do this?

The voip-info wiki hints that Asterisk can support whatever codecs
listed after a "show codecs" command:
> *CLI> show codecs
> Disclaimer: this command is for informational purposes only.
>         It does not indicate anything about your configuration.
>         INT    BINARY        HEX   TYPE    NAME   DESC
>
--------------------------------------------------------------------------------
>           1 (1 <<  0)      (0x1)  audio    g723   (G.723.1)
>           2 (1 <<  1)      (0x2)  audio     gsm   (GSM)
>           4 (1 <<  2)      (0x4)  audio    ulaw   (G.711 u-law)
>           8 (1 <<  3)      (0x8)  audio    alaw   (G.711 A-law)
>          16 (1 <<  4)     (0x10)  audio    g726   (G.726)
>          32 (1 <<  5)     (0x20)  audio   adpcm   (ADPCM)
>          64 (1 <<  6)     (0x40)  audio    slin   (16 bit Signed
Linear PCM)
>         128 (1 <<  7)     (0x80)  audio   lpc10   (LPC10)
>         256 (1 <<  8)    (0x100)  audio    g729   (G.729A)
>         512 (1 <<  9)    (0x200)  audio   speex   (SpeeX)
>        1024 (1 << 10)    (0x400)  audio    ilbc   (iLBC)
>       65536 (1 << 16)  (0x10000)  image    jpeg   (JPEG image)
>      131072 (1 << 17)  (0x20000)  image     png   (PNG image)
>      262144 (1 << 18)  (0x40000)  video    h261   (H.261 Video)
>      524288 (1 << 19)  (0x80000)  video    h263   (H.263 Video)
>     1048576 (1 << 20) (0x100000)  video   h263p   (H.263+ Video)
What is the 16 bit signed Linear PCM format? How do I get Asterisk (1.2)
to use such a sound file instead of a *.gsm file?

Your feedback is very much appreciated.

Cheers,

Stephen Bosch

Stephen Bosch wrote:
>Hello, everybody:
>
>I'm developing an application using Asterisk and a TDM-400 card.
>
>I understand the concept of the difference between GSM and WAV files
>when using Asterisk, but I'm not happy with the sound quality with the
>GSM compression. It's merely *acceptable* for a telephone call, but for
>anything else, it leaves something to be desired.
>
>Case in point -- if you compare the sound quality of the prompts with,
>say, an Octel voicemail system, there's no contest -- and this is
>through a standard telephone.
>
>1. Can I *record* audio from a TDM-400 channel at a sample rate above
>8000 kHz? Can I record at 16 bits? (This is most important.)
>
>2. Can I use a prompt with a sample rate above 8000 kHz? Will the
>TDM-400 card support it? How can I do this?
>
>The voip-info wiki hints that Asterisk can support whatever codecs
>listed after a "show codecs" command:
>
>  
>
>>*CLI> show codecs
>>Disclaimer: this command is for informational purposes only.
>>        It does not indicate anything about your configuration.
>>        INT    BINARY        HEX   TYPE    NAME   DESC
>>--------------------------------------------------------------------------------
>>          1 (1 <<  0)      (0x1)  audio    g723   (G.723.1)
>>          2 (1 <<  1)      (0x2)  audio     gsm   (GSM)
>>          4 (1 <<  2)      (0x4)  audio    ulaw   (G.711 u-law)
>>          8 (1 <<  3)      (0x8)  audio    alaw   (G.711 A-law)
>>         16 (1 <<  4)     (0x10)  audio    g726   (G.726)
>>         32 (1 <<  5)     (0x20)  audio   adpcm   (ADPCM)
>>         64 (1 <<  6)     (0x40)  audio    slin   (16 bit Signed
Linear PCM)
>>        128 (1 <<  7)     (0x80)  audio   lpc10   (LPC10)
>>        256 (1 <<  8)    (0x100)  audio    g729   (G.729A)
>>        512 (1 <<  9)    (0x200)  audio   speex   (SpeeX)
>>       1024 (1 << 10)    (0x400)  audio    ilbc   (iLBC)
>>      65536 (1 << 16)  (0x10000)  image    jpeg   (JPEG image)
>>     131072 (1 << 17)  (0x20000)  image     png   (PNG image)
>>     262144 (1 << 18)  (0x40000)  video    h261   (H.261 Video)
>>     524288 (1 << 19)  (0x80000)  video    h263   (H.263 Video)
>>    1048576 (1 << 20) (0x100000)  video   h263p   (H.263+ Video)
>>    
>>
>
>What is the 16 bit signed Linear PCM format? How do I get Asterisk (1.2)
>to use such a sound file instead of a *.gsm file?
>  
>
For speech, signed 16 bit linear offers no significant benefit over 
a-law or u-law. The hardware on the TDM400 card doesn't really support 
sampling rates above 8000/second, and phones are limited to that 
bandwidth anyway. If you want higher quality for some purpose, normal 
analogue phone lines are the wrong place to be looking.

None of the codecs currently supported by * support sampling rates above 
8000/second.

Regards,
Steve