Hello Kevin, On 29.01.20 at 20:22 Kevin Harwell wrote: > Greetings! > > 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 [1] 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. > > [1] 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: incoming_sdp_receive_prefs=local outgoing_sdp_send_prefs=remote outgoing_sdp_receive_prefs=local incoming_sdp_send_prefs=local transcode=avoid 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? Thanks Michael -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev