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

[Andreas Wehrmann]

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?

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?


All the best,
Andy