On Sat, Apr 8, 2017 at 7:23 AM, Dan Jenkins <dan.jenkin...@gmail.com> wrote:
> > On Fri, Apr 7, 2017 at 9:44 PM, Teijo <g.aloi...@gmail.com> wrote: > >> Hello, >> >> I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only >> problem until now which remained was that if dtls_rekey was set to the >> value other than 0, call hanged up when using chrome after the time where >> dtls_rekey was set. >> >> I suppose that "bad media description" shown in Chrome's window which >> causes call to fail, has appeared with Chromes newer versions (currently 58 >> beta installed) or with Asterisk 13.15.0. Audio codec I'm using is Opus. >> >> Has somebody else encountered this problem, or more better resolved it? >> >> Best regards, >> >> Teijo >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > Hi Teijo > > Take a read of https://nimblea.pe/monkey-business/2017/01/19/webrtc- > asterisk-and-chrome-57/ :) > 13.15.0 should address rtcp-mux issues. If there are still issues outstanding, it might be worth reporting a bug on issues.asterisk.org. Best wishes :-) -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
-- _____________________________________________________________________ -- 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