[Asterisk-Dev] IAX codec negociation (again...)

Jerris, Michael MI mjerris at ofllc.com
Mon Jun 20 07:12:17 MST 2005


> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Goldenear
> 
> I agree with you. we should have the choice between forcing 
> the use of the prefered codec (even if that sometimes means 
> transcoding) or forcing codec negociation and avoid transcoding.
> 
> Also, it would be very usefull if we could define both 
> incoming/outgoing
> codec(s) for each phone/enpoint in iax.conf and sip.conf. At 
> the moment, I far as I know, only the incoming codec(s) can 
> be specified separatly for each client... outgoing codecs are 
> common for all clients.
> 

This has been discussed many times before in great detail.  The fact is
it is not simple.  Asterisk is not a proxy that can just pass the
re-invite down the line to the other side, it can jump into the call and
do things, potentially affecting what codecs you would want to use.
Even the theory is quite complicated when you think of a call making
several hops from one end to the other.  What codec do you want each
voip hop to be?  How would we specify and communicate this down the call
path.  If you are really interested in making this happen, I would
suggest posting a sizable bounty for this, perhapse then somone will
take it on.  



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