Re: [asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"
Hello, Yes. When I today understood to set rtcp_mux=yes, at least Chrome (60.0 beta) worked (quickly tested) as expected. I'm sure that some day dtls_rekey can be set to the other value than 0 as well with Chrome. Best regards, Teijo 10.4.2017, 16.57, Matt Fredrickson kirjoitti: On Sat, Apr 8, 2017 at 7:23 AM, Dan Jenkinswrote: On Fri, Apr 7, 2017 at 9:44 PM, Teijo 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 :-) Tämä osa viestin runkoa ladataan pyydettäessä. -- _ -- 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
Re: [asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"
On Sat, Apr 8, 2017 at 7:23 AM, Dan Jenkinswrote: > > On Fri, Apr 7, 2017 at 9:44 PM, Teijo 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
Re: [asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"
Thank you Dan for this information. Best regards, Teijo 8.4.2017, 15:23, Dan Jenkins kirjoitti: On Fri, Apr 7, 2017 at 9:44 PM, Teijowrote: 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
Re: [asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"
On Fri, Apr 7, 2017 at 9:44 PM, Teijowrote: > 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
[asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"
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