On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote: <snip>
> > If I'm doing exactly the same call originated with another extension, > there can't be seen these frequent changes. But the strange thing is, > that in both cases the part between extension and asterisk doesn't show > any codec changes ... . > > Deeper investigations show, that if the conference (callee) sends the > first rtp package (-> g711 - should be g722), things are going choppy, > if the extension (caller) sends the first package (g722), things are > running stable. > > > Any idea to convince asterisk always to use the first codec of ok sdp > or how to convince asterisk to put only one codec to ok sdp (the first). This is not currently an option in chan_pjsip but I'd suggest filing an issue[1] for this scenario with all available information. [1] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
