[asterisk-dev] Voice parameters with RTP Init

[Hari kris]

Thanks for the answers. 
 
In current version, Should RTP Packets needed for any other processing like voice quality measurements etc., Or is there a simple way to disable such modules? 
 
> Date: Tue, 2 Sep 2008 09:08:30 -0500> From: jcolp at digium.com> To: asterisk-dev at lists.digium.com> Subject: Re: [asterisk-dev] Voice parameters with RTP Init> > ----- "Hari kris" <hari_vhk at hotmail.com> wrote:> > > If I use an external DSP which not only does voice codecs but also> > performs RTP/UDP packets out to the destination endpoint directly.> > Then what should be the design approach:> > a) How do we disable Host asterisk chan_sip to disable on-host RTP?> > b) Do other modules require RTP packets?> > c) How do we disable certain voice monitoring modules which run on RTP> > packets?> > > > Can I get any more documentation or some pointers to the discussion on> > integration of an external DSP processor?> > > > The capability to do this (a pluggable RTP architecture essentially) is being ironed out and will go into trunk. It is not possible in any current version of Asterisk to do it though currently.> > -- > Joshua Colp> Software Developer> Digium, Inc.> > _______________________________________________> --Bandwidth and Colocation Provided by http://www.api-digital.com--> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona> Register Now: http://www.astricon.net> > asterisk-dev mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-dev
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