[asterisk-dev] iLBC packet loss concealment

[Steve Underwood]

Mike Taht wrote:

>
>
> On 4/16/06, *Steve Underwood* <steveu at coppice.org 
> <mailto:steveu at coppice.org>> 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
>
>
> This is one reason why good quality, in particular, continuous, voice 
> recogition is nearly impossible over POTS. Even with limited domains, 
> at 8khz it's all billions of dollars of R&D can do just to 
> consistently mis-recognize "T-birds Pizza" in Los Gatos as "P-Birds 
> Pizza".
>
> I note that this article on the wiki debunks the ilibc assumption for 
> skype and goes into some depth:
>
> http://www.voip-info.org/wiki/view/Wideband+VoIP

I'd debunk the debunking. GIPS have changed the name, but their wideband 
codec is basically iLBC stretched to wideband.

>
> I went looking at mattf's asterisk wideband branch - seems out of 
> date, is it dead?
> http://svn.digium.com/view/asterisk/team/mattf/asterisk-wideband/
>
> and some interesting old threads...
> http://lists.digium.com/pipermail/asterisk-dev/2005-June/013314.html 
> <http://lists.digium.com/pipermail/asterisk-dev/2005-June/013314.html>

Regards,
Steve