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

[Zoa]

The PLC for iLBC used in asterisk is not the generic one, but the one 
inside iLBC iirc.

Zoa

John Todd wrote:

> At 2:45 PM +1000 4/16/06, Adrian Sietsma wrote:
>
>> Matt Ranney wrote:
>>  > Somehow the jitter buffering or packet loss concealment or whatever
>>
>>> magic that Skype uses makes it work better than asterisk/IAX over 
>>> the  same WAN link.
>>
>> Having done some testing of voip over WAN, I would say it is both. I 
>> have seen packet loss of up to 20-30% (using a CDMA wireless card). 
>> Skype's codecs, including iLBC, seem very tolerant of packet loss.
>>
>> Their client also has effective echo suppression, which makes using 
>> it without a headset fairly painless.
>>
>> Asterisk is also very prone to jitter problems, if I understand 
>> correctly. Since outgoing RTP packets are triggered by incoming, that 
>> would reflect any incoming jitter back to the sender, as well as 
>> passing it on. Issue 5374 should solve this.
>>
>> Adrian
>
>
>
> Time to break this out into a thread with a title that matches it's 
> content.
>
> I have not been 100% up-to-date on the implementation of iLBC's 
> Packet-Loss-Concealment code within Asterisk.  I know that there is 
> basic PLC in Asterisk, but is as good as it could possibly be with 
> iLBC's open-source implementation?  Are there any methods by which 
> this new asynchronous packet generation could improve the situation, 
> or is PLC by definition something that the receiver needs to worry about?
>
> Follow-up questions: is the existing PLC only available in IAX2, or is 
> it generally available in RTP?
>
> http://voip-info.org/tiki-index.php?page=Asterisk+new+jitterbuffer
>
> Skype has good quality because of several factors: they are primarily 
> soft-client based, so they can use better-than-telephony sound quality 
> on both sides of the connection to start with, and secondly because 
> they are using a patent-encumbered version of iLBC from Global IP 
> Sound (GIPS) which I assume has both better PLC as well as better 
> compression characteristics.  I'll put my $.02 towards Asterisk 
> supporting better codecs at higher bitrates (Speex, notably, has the 
> ability to do much better than "MOS 4", or "toll-quality" sound - 
> stereo, higher bitrates, etc.)  since this is next-generation 
> telephony system implementation, not last-generation telephony system 
> design.
>
> I digress... anyone (SteveK?) have comments on generalized iLBC PLC 
> for SIP?
>
> JT
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