On 05/12/2017 at 08:49 PM, Joshua Colp wrote: > 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
https://issues.asterisk.org/jira/browse/ASTERISK-26996 Thanks, regards, Michael -- _____________________________________________________________________ -- 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
