[asterisk-users] dynamic codec preferences

[Michael Graves]

On Tue, 31 Mar 2009 16:12:45 -0400, Mike wrote:

>Maybe something like that could be done by using "set groups" and counting
>the number of calls, and at a specified threshold (i.e. 6 simultaneous
>calls) you`d specify g729 for new calls.  
>
>Shifting an ongoing call might be impossible though.
>
>Mike

I'd not expect to shift an ongoing call. But if you know that your IP
connection is constrained then when you have x calls in progress you
might benefit from forcing any additional new calls into G.729. At
least until your simultaneous call volume drops a bit.

I image that most people who use G.729 simply use it all the time. But
then, why take the quality hit if your call momentary volume is low?

Michael

>
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
>> bounces at lists.digium.com] On Behalf Of Michael Graves
>> Sent: Tuesday, March 31, 2009 11:08
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] dynamic codec preferences
>> 
>> Has anyone here ever had the occasion to setup a system that would
>> dynamically alter it's codec preferences based on trafffic? That is,
>> presuming that the system is on a limited bandwidth connection is would
>> start to prefer a compressed codec as the call volume increased?
>> Perhaps shifting from G.711 to G.729?
>> 
>> Michael
>> --
>> Michael Graves
>> mgraves<at>mstvp.com
>> http://blog.mgraves.org
>> o713-861-4005
>> c713-201-1262
>> sip:mgraves at mstvp.onsip.com
>> skype mjgraves
>> fwd 54245
>> 
>> 
>> 
>> 
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--
Michael Graves
mgraves<at>mstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mgraves at mstvp.onsip.com
skype mjgraves
fwd 54245