[asterisk-dev] [Code Review] SIP Preferred codec only

[Joshua Colp]

----- "Klaus Darilion" <klaus.mailinglists at pernau.at> wrote:

> David Vossel wrote:
> > ----------------------------------------------------------- This is
> > an automatically generated e-mail. To reply, visit: 
> > http://reviewboard.digium.com/r/206/ 
> > -----------------------------------------------------------
> > 
> > (Updated 2009-03-24 14:02:50.318794)
> > 
> > 
> > Review request for Asterisk Developers.
> > 
> > 
> > Summary (updated) -------
> > 
> > Added an option to respond to a SIP invite with only the single
> most
> > preferred joint codec.  This limits the options of what codecs the
> > other side can use.
> Hi!
> 
> What is the use case of this option? Doesn't Asterisk support
> switching 
> codecs on-the-fly (without reINVITE)?
> 

We support changing the format of incoming and outgoing audio when an RTP packet is received
that is different then the previous format. We have no method of changing the outgoing format
internally.

This option can be used to force the entire dialog to one format from the beginning of the call.

I can imagine it being used in a scenario where a SIP proxy is in use with multiple clients with different
configurations behind it. While a client may request both ulaw and g729 we actually only want to force them to
g729 to conserve our own resources. It gives a bit tighter control over the negotiation without failing the call.

-- 
Joshua Colp
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org