[asterisk-users] Asterisk not using common codec between (SIP) endpoints

Joshua C. Colp jcolp at digium.com
Fri Oct 4 05:05:08 CDT 2019

On Fri, Oct 4, 2019, at 1:45 AM, Andreas Wehrmann wrote:
> On 03/10/2019 16:24, Joshua C. Colp wrote:
> > In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately codec negotiation is not written or implemented in the way you need. There are some hints provided internally for outgoing legs but the result is still ultimately independent. That is: Each leg is negotiated from Asterisk to the endpoint, not endpoint to endpoint via Asterisk. This works for the vast majority of users as they have media flowing through Asterisk (by choice or via use of features) and are fine with transcoding (generally using codecs which aren't that costly or low channel count).
> >
> > Asterisk 16 has some of the foundational work to improve this through the implementation of streams but noone has worked on extending the codec negotiation support.
> >
> > [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Function_PJSIP_MEDIA_OFFER
> >
> Hey Joshua,
> do you think it might be possible to achieve this by writing a 
> supplement for the PJ part?

PJSIP is only part of the equation, the information still has to transition across the core.

> What happens when the other side answers, but before the incoming call 
> is answered.
> Is there a place in the code where, at that point, I have information 
> about both channels
> and could theoretically influence the answer for the incoming call?

Nope. That information is not currently exchanged or available, which is part of the problem.

Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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