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

Steve Underwood steveu at coppice.org
Sun Apr 16 08:02:38 MST 2006


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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