[Asterisk-Users] C7960 g729 question

[Rich Adamson]

Not sure I understood the comments... the existing situtation (as shown in
the original email) is a remote 7960 uses g729, and an internal 7960 that
uses g711 (both forced via sip.conf entries). The audio is oftentimes
choppy, "but" changing the g729 side to g711 (still passing through * with
canreinvite=no) clears up the audio. Same exact path in both cases without
a doubt. CPU is 2.2ghz, 100 meg full duplex, <2% utilization during a "single"
call in the * system.

Logic would suggest the g711-to-g711 (with canreinvite=no) should have been
a lower quality then the g729-to-g711 (due to any form of bandwidth contraints
that might impact either approach). But, its the other way around; the
g729-to-g711 is choppy. I'm searching for "where" to go look for the problem
and the only logical conclusion that I've come up with is that it appears
to be something associated with the transcoding.

Thoughts?

Rich

------------------------
> Have you checked the CPU load on your asterisk server? 
> 
> If you have no change of codec between the two sides, asterisk just
> pasts the audio straight through. If there is a change, conversion has
> to take place and conversion from g729 to anything else does require cpu
> power. 
> 
> Another thing you could do however is allow the C7960 to use g729 inside
> sip.conf just not as the first codec, that way it will use g711 but pass
> on calls direct if g729.
> 
> Hope that helps
> 
> Matthew Enger
> m.enger at xi.com.au
> 
> 
> On Sat, 2004-06-19 at 01:12, Rich Adamson wrote:
> > I have multiple voiceage g729 licenses installed on a RH9 box, and have
> > a remote C7960 configured to use it (low bandwidth). In calls like:
> > 
> >   Remote C7960 -> g729 -> asterisk -> g711 -> C7960
> > 
> > the audio is oftentimes rather choppy. Changing the remote 7960 to use
> > g711 seems to eliminate/reduce the choppyness. Any ideas on what might
> > be behind this?
> > 
> > 
> > 
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> -- 
> Matthew Enger <menger at xi.com.au>
> Xintegration
> 
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