[Asterisk-Users] super high bandwidth codec

[Andrew C. Brown]

Michael Graves wrote:
> On Tue, 26 Jul 2005 18:50:11 -0700, Andrew C. Brown wrote:
> 
> 
>>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
>>
> 
> 
> A recent blog entry indicated that GIPS was issuing licenses for its
> technology from a mere $50k for "unlimited licenses" with respect to an
> agreement with Microsoft. I don't have a huge concern about bandwidth
> limits. If I could get better quality than G.711 in the same bandwidhth
> that would be great.
> 
> However, since I'm using IAX2 based DIDs and termination would it
> really matter? That is, if the ITSPs are connection to the PSTN via TDM
> interconnects wouldn't any calls be limited to G.711 quality anyway?

IAX2 is a protocol, not a codec, so has little impact on sampling
quality. But the second assumption is correct. If you are going to PSTN
at any point in the chain, you are back to 8kHz sample rate and that
extra spectrum you put over iSAC or whatever is tossed out the window.