[Asterisk-Users] super high bandwidth codec

[Andrew C. Brown]

Brian Capouch wrote:
> Geoff Manning wrote:
> 
>>>>> Skype uses wideband-ilbc.
>>>>>
>>>>
>>
>> I don't think thats right. I think it just uses iLBC over it's own
>> proprietary Voip protocol.
>> http://www.skype.com/help/faq/technical.html
>> How much bandwidth does Skype use while I'm in a call?
>>     Skype automatically selects the best codec depending on the
>> connection
>> between yourself and the person you are calling. On average, Skype uses
>> between 3-16 kilobytes/sec depending on bandwidth available for other
>> party,
>> network conditions in between, callers CPU performance, etc.
> 
> 
> I don't think that's correct.
> 
> I don't have the link to the Columbia paper where they tried (with only
> mixed success) to figure out what all nefarious stuff Skype does
> (hijacking port 80 being the most pernicious) but I'm pretty sure they
> have figured out that if possible, it will use the (proprietary)
> wideband version of iLBC.
> 

FYI: One can find the columbia paper link by going to the VoIP wiki's
Skype page.

According to GIPS datasheets, GIPS offers two proprietary wideband
codecs. iPCM-wb and iSAC. Both have 16kHZ sample rate* which is double
the 8kHz of PSTN, iLBC and most of the other codecs, hence the
relatively wonderous sound quality which I, among others, covet for
Asterisk.

The channel bit rate is respectively (it varies dynamically)
iLBC (free)	13-15kbps
iSAC ($)	10-30kbps
iPCM-wb ($)	80kbps

iPCM-wb doesn't seem to offer any outright fidelity advantage over iSAC
since they are the same sample rate. I presume all those extra bits are
redundancy to make the quality more robust.

* Ref: http://www.globalipsound.com/solutions/solutions_Codecs.php