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

[Matthew Fredrickson]

On Apr 16, 2006, at 1:06 PM, Mike Taht wrote:

>
>
> On 4/16/06, Steve Underwood <steveu at coppice.org > wrote:
>> 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 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/
>

Not entirely out of date.  I have a lot of work to do still and haven't 
had much time to do it, but I think in a week or two I should be able 
to get things moving again with it.  I just put in an (untested) g.722 
codec implementation and I'm working on a new resampling codec at the 
moment.

Matthew Fredrickson