News Reader wrote:
>
> "John Miller" <john.miller@nospamplease.com> wrote in message
> news:46744148$0$3765$c3e8da3@news.astraweb.com...
>>> Looks like they use a wideband codec. Is that what you mean?
>>
>> Yes. 8 kHz of audio bandwith instead of the standard 3,3 kHz.
>>
>>> Codec use is usually a matter for the endpoints, not the SIP provider,
>>> unless they're doing something like transcoding all of the calls.
>>
>> Still, I'd like to know who supports it.
> I thought G711 was a standard PCM full 8khz sampling codec...
>
> "G.711:
> - Sampling frequency 8 kHz
> - 64 kbit/s bitrate (8 kHz sampling frequency x 8 bits per sample) "
>
> G711 is a pretty universal codec!
As John Miller pointed out [hint: you've quoted it] standard PSTN audio
bandwidth is 3.3kHz, you're thinking about sampling frequency. One is half
the other, have a read of this if you want a more accurate explanation:
http://en.wikipedia.org/wiki/Nyquist...mpling_theorem
> P.s. Always seemed a little pointless to me going beyond 8khz when PSTN is
> max 8khz and mobile typically less? Admittedly, in the future as something
> better becomes the norm then such higher quality approaches may be more
> worthwhile...?
As others have alluded to in this thread, if you're making a SIP to VoIP
call, you can use whatever codec you want, including codecs that are higher
bandwidth than G.711. I personally find compressed audio more intelligible
in a noisy environment; for example compare listing to Radio 5 on MW to
listening to Radio 5 on Freeview.
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