[asterisk-dev] Asterisk 18 Planning: Codec Negotiation
m1278468 at mailbox.org
Wed Jan 29 15:11:55 CST 2020
On 29.01.20 at 20:22 Kevin Harwell wrote:
> Over the years there have been numerous requests to improve the codec
> negotiation process in Asterisk. Specifically, regarding what codecs are
> offered, in what order, how Asterisk chooses which codec(s) to use, and of
> course how transcoding is affected by that.
I'm really happy to hear that you are going to improve the codec
handling! Thanks for that!
> Well hopefully that wait will soon be over. Currently we have plans to work
> on this for Asterisk 18.
Will Asterisk 18 be a LTS version?
> The bulk of that work will be around the addition
> of new chan_pjsip options that will allow a user to better control codec
> offerings, and order.
> I've added a page to the wiki  beneath the Asterisk 18 roadmap page that
> explains what those options are (along with a couple current codec related
> ones), and how they will work. Please, if you have any interest in this
> topic read through that page and let us know what you think, or how things
> can be improved.
>  https://wiki.asterisk.org/wiki/display/AST/Codec+Negotiation
>From my point of view, it should always be possible to prevent
transcoding as long as there is one codec which can be used on both
sides. If there is more than one codec equal on both sides, it's good to
have the possibility by your planned options if the local or the remote
most preferred codec should be used.
Default configuration for me would be like that:
>From my understanding, this should avoid any unnecessary transcoding as
long as there's just one common codec on both sides and should always
prefer the codecs desired by the caller.
Did I got this correctly?
More information about the asterisk-dev