[asterisk-dev] Voice parameters with RTP Init

[Tzafrir Cohen]

On Tue, Sep 02, 2008 at 04:18:55AM +0000, Hari kris 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? 

Write an alternative codec_* module . See, e.g. codec_zap / codec_dahdi .

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               Tzafrir Cohen
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