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Hello,

Does anyone know if there are any means of doing high quality voip.
By high quality I mean sound quality that would be higher than traditional
phone between on net clients.

In theory it should be possible: if we use a voice stream compressed to
64kb/s using mp3 for example, we could have a very high quality end to end
communication for the same bandwidth as G711.

The question is: does any client support this kind of "high quality"
algorithm: softphone, Ip phones, PBX, pstn gw... ?

It would be a good way to promote voip: the main argument would not be only
price but sound quality.



--
Alexis de Goriainoff

Alexis de Goriainoff wrote:

> Hello,
>
> Does anyone know if there are any means of doing high quality voip.
> By high quality I mean sound quality that would be higher than
> traditional phone between on net clients.
>
> In theory it should be possible: if we use a voice stream compressed to
> 64kb/s using mp3 for example, we could have a very high quality end to end
> communication for the same bandwidth as G711.
>
> The question is: does any client support this kind of "high quality"
> algorithm: softphone, Ip phones, PBX, pstn gw... ?
>
> It would be a good way to promote voip: the main argument would not be
> only price but sound quality.
>
>
>
> --
> Alexis de Goriainoff

Is MP3 a high-quality sound codec?

--
root/administrator

> > --
> > Alexis de Goriainoff
>
> Is MP3 a high-quality sound codec?

mp3 is a codec that can produce sound with high bit rate.

>
> --
> root/administrator

Alexis de Goriainoff wrote:

>
>
>> > --
>> > Alexis de Goriainoff
>>
>> Is MP3 a high-quality sound codec?
>
> mp3 is a codec that can produce sound with high bit rate.
>

What will be the compression ratio of an MP3 file when set with a high-bit
rate to achieve a high-quality voice as mentioned in the OP' post? Will the
MP3 file become larger?

--
root/administrator

Alexis de Goriainoff wrote:
> Hello,
>
> Does anyone know if there are any means of doing high quality voip.
> By high quality I mean sound quality that would be higher than traditional
> phone between on net clients.
>
> In theory it should be possible: if we use a voice stream compressed to
> 64kb/s using mp3 for example, we could have a very high quality end to end
> communication for the same bandwidth as G711.
>

remember that traditional phone only use 300-3400 Hz of bandwidth
so does voip codecs
g711 codec (64kbit/s) causes no loss of quality
it's a no-compression codec


Kuba

On Tue, 18 Nov 2003 19:01:10 +0100, "Alexis de Goriainoff"
<> wrote:

>Hello,
>
>Does anyone know if there are any means of doing high quality voip.
>By high quality I mean sound quality that would be higher than traditional
>phone between on net clients.
>
>In theory it should be possible: if we use a voice stream compressed to
>64kb/s using mp3 for example, we could have a very high quality end to end
>communication for the same bandwidth as G711.
Look into G722, which has a 7 kHz input audio range and compresses
speech to 64Kb/s. There are proprietary codecs that have a 7kHz input
and compress speech to 16Kb/s.

Be aware that MP3 is an audio coding scheme, while speech codecs like
G.729 are speech coding schemes. The difference is that MP3 models
what the ear can hear, and speech codecs model what the human vocal
tract can produce. In general, speech codecs will sound good for
speech but not for music--we can hear a piano, but can't sound like
one.

>
>The question is: does any client support this kind of "high quality"
>algorithm: softphone, Ip phones, PBX, pstn gw... ?
Many videoconferencing devices support G.722.
>
>It would be a good way to promote voip: the main argument would not be only
>price but sound quality.

On Thu, 20 Nov 2003 15:51:08 +0100, kubel <>
wrote:

>Alexis de Goriainoff wrote:
>> Hello,
>>
>> Does anyone know if there are any means of doing high quality voip.
>> By high quality I mean sound quality that would be higher than traditional
>> phone between on net clients.
>>
>> In theory it should be possible: if we use a voice stream compressed to
>> 64kb/s using mp3 for example, we could have a very high quality end to end
>> communication for the same bandwidth as G711.
>>
>
>remember that traditional phone only use 300-3400 Hz of bandwidth
>so does voip codecs
>g711 codec (64kbit/s) causes no loss of quality
>it's a no-compression codec

IIRC, uLaw on TDM has a 4.4 MOS score. G.711 with 10 mSec packets
and no packet loss has a 4.2 MOS score. The best MOS score is 5.0, so
G.711 does have some loss of quality.

Also, the codec is only one factor in determining the quality of a
call. Delay and packet loss are two of the many other factors. See
www.voiptroubleshooter.com for more information on VoIP quality.

In article <bpdn0u$9i2$>,
Alexis de Goriainoff <> wrote:
>Hello,
>
>Does anyone know if there are any means of doing high quality voip.
>By high quality I mean sound quality that would be higher than traditional
>phone between on net clients.
>
>In theory it should be possible: if we use a voice stream compressed to
>64kb/s using mp3 for example, we could have a very high quality end to end
>communication for the same bandwidth as G711.
>
>The question is: does any client support this kind of "high quality"
>algorithm: softphone, Ip phones, PBX, pstn gw... ?


Check out Speex (which is free) or the codecs from Global IP Sound
(which are not). Some softphones have Speex in them. I believe
Skype uses the Global IP Sound codecs.
--
What's the future of TV and advertising after Tivo?
http://www.templetons.com/brad/tvfuture.html

"Hank Karl" <> wrote in message
news:...
> On Tue, 18 Nov 2003 19:01:10 +0100, "Alexis de Goriainoff"
> <> wrote:
>
> >Hello,
> >
> >Does anyone know if there are any means of doing high quality voip.
> >By high quality I mean sound quality that would be higher than
traditional
> >phone between on net clients.
> >
> >In theory it should be possible: if we use a voice stream compressed to
> >64kb/s using mp3 for example, we could have a very high quality end to
end
> >communication for the same bandwidth as G711.
> Look into G722, which has a 7 kHz input audio range and compresses
> speech to 64Kb/s. There are proprietary codecs that have a 7kHz input
> and compress speech to 16Kb/s.
>
> Be aware that MP3 is an audio coding scheme, while speech codecs like
> G.729 are speech coding schemes. The difference is that MP3 models
> what the ear can hear, and speech codecs model what the human vocal
> tract can produce. In general, speech codecs will sound good for
> speech but not for music--we can hear a piano, but can't sound like
> one.
>
> >
> >The question is: does any client support this kind of "high quality"
> >algorithm: softphone, Ip phones, PBX, pstn gw... ?
> Many videoconferencing devices support G.722.

I think that some broadcasters use G.722 or better for FM quality sound for
outside broadcast over ISDN - e.g. concerts, sports and so on.

Maybe you need to ask about something that traditionally has better sound
quality - try an audio group?

> >
> >It would be a good way to promote voip: the main argument would not be
only
> >price but sound quality.
--
Regards

Stephen Hope - remove xx from email to reply