[asterisk-dev] iLBC packet loss concealment (was: code-cleanup concerns)

[Anton]

Thanks for explanation Stieve, the next question arizes, 
sorry if it may be stupid :) , Does not asterisk use 64kbps 
internally (signed linear), so any codec would be converted 
to 64k first when audio is processed by Ast*?

On 16 April 2006 20:02, Steve Underwood wrote:
> These days, you can achieve far better quality than a
> normal phone call at rates much lower than 64kbps.
>
> The main quality issue with normal phone calls is they
> are limited to 4kHz bandwidth. This is insufficient for
> good quality speech. 8kHz bandwidth really improves
> things. It lets you distinguish things "f" from "s",
> which is almost impossible on a normal phone line. In the
> 1980s ISDN was introduced with the promise of 8kHz
> bandwidth (actually specified as 7.1kHz), using a codec
> called G.722. This uses 48, 56 or 64kbps, and is
> dramatically better than a normal phone call. Because
> fully digital end-to-end connections never became common,
> G.722 never became common either. These days, modern
> codecs do much better than G.722, but even clunky old
> G.722 at 48kbps is clearly better than u-law or A-law at
> 64kbps.
>
> So, forget these weird notions many people have about the
> magic of A-law and u-law phone calls:
>     - Normal phone calls have too limited bandwidth for
> good quality. One of the benefits of VoIP should be to
> break the 100 year old model of <4kHz bandwidth calls.
>     - They are not uncompressed - u-law and A-law are
> lossy compression schemes, which start at 96kbps, and
> compress this to 64kbps. They use a very simple, but very
> obsolete way of doing that. - Modern compression doesn't
> have to be about achieving indifferent quality at super
> low bit rates (e.g. G.729). It can be about achieving
> really good quality at medium bit rates in the 30-64kbps
> range.
>
> Regards,
> Steve
>
> Anton wrote:
> >I may be wrong, but all going about using loose codec
> > iLibc or Speex in skype and * in compariosion to PSTN
> > connection, and how that codecs may sound better than
> > PSTN? Or I'm missing something?
> >
> >On 16 April 2006 19:25, Steve Underwood wrote:
> >>Hi Anton,
> >>
> >>I'm sure we'd all love to hear an explanation for that.
> >>
> >>Regards,
> >>Steve
> >>
> >>Anton wrote:
> >>>IMHO: To sound better than PSTN you either must
> >>> transmit more than 64Kbps audio or have poor PSTN
> >>> connection to compare :)
> >>>
> >>>On 16 April 2006 18:33, Steve Underwood wrote:
> >>>>Andrew Kohlsmith wrote:
> >>>>>On Sunday 16 April 2006 04:05, Matt Ranney wrote:
> >>>>>>My users tell me that Skype sounds better than
> >>>>>>asterisk even when going to the PSTN, but I think
> >>>>>>this might be because the headsets they are using
> >>>>>>just sound better than their Cisco 7940 handsets.
> >>>>>
> >>>>>the iLBC codec used in Skype is the wideband
> >>>>> variety; it *is* better than PSTN.
> >>>>
> >>>>Skype is certainly better than the PSTN when going
> >>>>Skype to Skype, but why do people think it sounds
> >>>>better than the PSTN when going to the PSTN? Is it
> >>>>just psycological, or are they using an environment
> >>>>(e.g. headset) that just sounds better than their
> >>>>usual phone? Having they been using a Cisco 7940 with
> >>>>G.729, where a real PSTN to PSTN connection would
> >>>>sounds rather better? I find it an interesting
> >>>> comment that Matt made.
> >>>>
> >>>>Steve
>
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