[asterisk-dev] rtp data in an app question

[Russell Bryant]

Gallmeier, Jonathan wrote:
> Since I'm using a SIP interface, will setting the format via the
> ast_set_read_format() cause the SIP stack to re-negotiate the codec with
> the SIP phone? Or, will it simply insert a codec and translate the audio
> format by decoding and re-encoding? I'd rather re-negotiate the audio
> codec with the phone if possible.

It will set up a translation path, not renegotiate with the other end.

-- 
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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