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Which SIP providers support HD voice?

AVM's Fritz!box 7170 firmware 29.04.34-7269 supports this:
[url]http://www.avm.de/de/Service/Service-Portale/Service-Portal/Labor/labor.php[/url]

Thanks!

On Sat, 16 Jun 2007 16:39:44 +0200, "John Miller"
<john.miller@nospamplease.com> wrote:
[color=blue]
>Which SIP providers support HD voice?[/color]

What do you mean by HD voice?
[color=blue]
>AVM's Fritz!box 7170 firmware 29.04.34-7269 supports this:
>[url]http://www.avm.de/de/Service/Service-Portale/Service-Portal/Labor/labor.php[/url][/color]

Looks like they use a wideband codec. Is that what you mean?

Codec use is usually a matter for the endpoints, not the SIP provider,
unless they're doing something like transcoding all of the calls.

> 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.
[color=blue]
> Codec use is usually a matter for the endpoints, not the SIP provider,
> unless they're doing something like transcoding all of the calls.[/color]

Still, I'd like to know who supports it.

On Sat, 16 Jun 2007 22:05:57 +0200, "John Miller"
<john.miller@nospamplease.com> wrote:
[color=blue][color=green]
>> Looks like they use a wideband codec. Is that what you mean?[/color]
>
>Yes. 8 kHz of audio bandwith instead of the standard 3,3 kHz.
>[color=green]
>> Codec use is usually a matter for the endpoints, not the SIP provider,
>> unless they're doing something like transcoding all of the calls.[/color]
>
>Still, I'd like to know who supports it.[/color]

I know that BT use G.722 & G.722.2 (AMR-WB) on the Videophone 1000 &
PC Softclient.

But like I said, if you get two phones that support a wideband codec,
then there's no reason it wouldn't work with most SIP providers,
especially if both phones are on the same provider.

"John Miller" <john.miller@nospamplease.com> wrote in message
news:46744148$0$3765$c3e8da3@news.astraweb.com...[color=blue][color=green]
>> Looks like they use a wideband codec. Is that what you mean?[/color]
>
> Yes. 8 kHz of audio bandwith instead of the standard 3,3 kHz
> Still, I'd like to know who supports it.[/color]

What supports it i.s.o. who, as providers are not in the loop.

rfc3551:
L16 denotes uncompressed audio data samples, using 16-bit signed
representation with 65,535 equally divided steps between minimum and
maximum signal level, ranging from -32,768 to 32,767. The value is
represented in two's complement notation and transmitted in network
byte order (most significant byte first).

As far as i know, no other consumer SIP hardware supports L16.

Cheers,
Jack

"John Miller" <john.miller@nospamplease.com> wrote in message
news:46744148$0$3765$c3e8da3@news.astraweb.com...[color=blue][color=green]
>> Looks like they use a wideband codec. Is that what you mean?[/color]
>
> Yes. 8 kHz of audio bandwith instead of the standard 3,3 kHz.
>[color=green]
>> Codec use is usually a matter for the endpoints, not the SIP provider,
>> unless they're doing something like transcoding all of the calls.[/color]
>
> Still, I'd like to know who supports it.
>[/color]


Hi,


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!


Best wishes,



News Reader


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...?

News Reader wrote:
[color=blue]
>
> "John Miller" <john.miller@nospamplease.com> wrote in message
> news:46744148$0$3765$c3e8da3@news.astraweb.com...[color=green][color=darkred]
>>> Looks like they use a wideband codec. Is that what you mean?[/color]
>>
>> Yes. 8 kHz of audio bandwith instead of the standard 3,3 kHz.
>>[color=darkred]
>>> Codec use is usually a matter for the endpoints, not the SIP provider,
>>> unless they're doing something like transcoding all of the calls.[/color]
>>
>> Still, I'd like to know who supports it.[/color][/color]
[color=blue]
> 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![/color]

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:

[url]http://en.wikipedia.org/wiki/Nyquist?Shannon_sampling_theorem[/url]
[color=blue]
> 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...?[/color]

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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John Miller wrote:[color=blue]
> Which SIP providers support HD voice?[/color]

I don't know any. I don't think asterisk will support it.
[color=blue]
>
> AVM's Fritz!box 7170 firmware 29.04.34-7269 supports this:
> [url]http://www.avm.de/de/Service/Service-Portale/Service-Portal/Labor/labor.php[/url][/color]

As does the Snom370. Using G.722.

Tim