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/ :) Dan
-- _____________________________________________________________________ -- 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
