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

[John Todd]

[continuing top-threading madness ;-]

I agree with everything said thus far in this thread, and I 
especially agree with Steve's assertions that 4khz is an artificial 
and sub-standard limit to set for ourselves when we have so much more 
to work with.  Let me throw a few actual questions to the group, so 
we can stop being theoretical and try to improve our "product":

Statements:
  1) I have heard rumors (not from anyone at Skype) that the iLBC 
codec used by Skype actually _does_ sound better than normal iLBC 
even at higher bitrates, due to some clever manipulation of dynamic 
range.  Can anyone confirm or deny this?  This might suggest how PSTN 
calls sound "better" even though it's still only G.711 on the PSTN 
side.

  2) I've often had problems with Skype's voice quality, so I won't 
give them the perfect scores that everyone else is attributing to 
them.  However: Kudos for them for throwing out the old rules.


Questions:

  1) Does the iLBC codec code in Asterisk perform Packet Loss 
Concealment (PLC)?  If so, is it being done in the most optimal way? 
Is it just in IAX2 or in SIP as well?  (there was an answer on this 
by Zoa, but it still isn't clear.)

  2) Are the higher bitrate patches I've seen scurrying around the SVN 
branch mailing list currently in TRUNK or still in "experimental test 
phase"?  (see oej's "test-this-branch", and bug #5084, which really 
just seems to support G.722 passthrough, and is not a codec 
implementation)

  3) Is this all a moot point until more end devices support iLBC and 
better PLC? (Hint: I don't think so, since I'd love to terminate all 
my LD minutes to an Asterisk server with high-bitrate, 
loss-concealing iLBC instead of just using G.711 like I do today. 
Bitrates are not my problem; sound quality is my problem, but if I 
get better than 64kbps for better sound quality then that's a bonus.)

JT


>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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