[Asterisk-Users] Asterisk sound files, audio bandwidth, and sound
quality
Steve Underwood
steveu at coppice.org
Wed Sep 28 10:08:53 MST 2005
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
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