[Asterisk-Users] super high bandwidth codec

[Andrew C. Brown]

Tzafrir Cohen wrote:
> On Tue, Jul 26, 2005 at 10:42:35PM -0700, Andrew C. Brown wrote:
> 
>>>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.
> 
> 
> And also when you use MeetMe, right?
> 

I'm researching that. All I've been able to find so far is
http://lists.digium.com/pipermail/asterisk-users/2005-May/107214.html
which says that basically, no, Asterisk can't yet handle anything but
8KHz sample rates (though I suppose that doesn't necessarily preclude
reinvited peer to peer VoIP calls where Asterisk removes itself from the
audio path).

If you find any more references on that issue, please post them. This
question of high quality voice is going to keep coming up so I'd like
there to be Wiki page to bring people up to date on all this we're
discussing. And frankly I'd like to help build some momentum towards
increased spectrum voice telephony. Right now, few people even think to
ask and VoIP to them is just about saving money rather than improving
the product.