[asterisk-dev] Configuration for recording quality?

[Steve Underwood]

Alexander Lopez wrote:

>snip
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>>On 8/4/06, Jan du Toit <jan.du.toit at decisionworx.com> wrote:
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>>>Hi.
>>>
>>>I have recently posted a mail on the users mailing list, asking
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>around
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>>>how to change the quality setting of files that asterisk record for
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>you.
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>>>For instance change the 8kHz for meetme recordings to 32kHz.
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>>>
>>I don't think there would be any point (that is to say, any quality
>>improvement) in recording at anything other than 8KHz at present,
>>since all audio passing through Asterisk is sampled at 8KHz anyway. If
>>you need 32KHz files for compatibility with another application, it
>>should be fairly trivial to upsample the recorded files using a
>>utility such as sox.
>>
>>-Rusty
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>>
>
>There was a discussion about having the audio passed through * be at a
>higher rate than 8k, thinking behind it was that the 8K limit is imposed
>by the PSTN and VoIP does not exibit that inherent limitation. I do not
>know of any endpoints that support a higher rate (>8k), it would be nice
>to be able to use a High-Quality MeetMe room for a podcast-type live
>discussion.
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Want a list? Start with:
    Many ISDN phones (G.722)
    Many recent GSM phones (AMR-WB)
    Any 3G phone (AMR-WB)
    A wide range of SIP and H.323 phones (typicall G.722, but some are 
getting other codecs now).
    Most Skype phones (a codec from GIPS, which is kind of a wideband 
version of iLBC).

The world is waiting for Asterisk to catch up. :-)

>Will the incorporation of Video into a MeetMe room change this at all??
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Probably not. Most video conferencing still uses narrow band voice.

Steve