Re: [asterisk-users] Webrtc and iOS devices
Hello, I upgraded to 16.9.0 and then 16.10.0. I feel that there is something in iOS webrtc implementation which cause issues. If womebody has succeeded with combination iOS, browser (like Safari) and webrtc (conference call), it would be nice to hear. Best regards, Teijo Dan Jenkins kirjoitti 28.4.2020 klo 13.41: I honestly couldn't tell you if it would resolve it but there aren't many people going to be willing to help problem solve anything if you're running 13 - you'll get more support on 17 for example. Very easy to bring up a new instance or VM in the grand scheme of things to test the theory and get it working on most recent version of Asterisk On Tue, Apr 28, 2020 at 11:37 AM Teijo wrote: Hello, Currently audio conference. Should upgrading Asterisk from 13 to newer version resolve webrtc/iOS problem? Best regards, Teijo Dan Jenkins kirjoitti 28.4.2020 klo 12.18: First things first, upgrade from 13 - WebRTC has moved a long a lot since then. If you can't upgrade everything to 13 then run another asterisk specifically for WebRTC and bridge to your other Asterisk Is this just an audio conference? On Sun, Apr 26, 2020 at 10:21 PM Teijo wrote: Hello, Has somebody get combination Asterisk (I'm using version 13.32.0, webrtc and iOS (version 13.4.1) with Safari or any other browser working properly in confbridge conference calls? I hope my Asterisk webrtc related settings are not totally wrong, because several other browsers from Windows seem to work. 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 -- _ -- 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 -- _ -- 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] Webrtc and iOS devices
Hello, Currently audio conference. Should upgrading Asterisk from 13 to newer version resolve webrtc/iOS problem? Best regards, Teijo Dan Jenkins kirjoitti 28.4.2020 klo 12.18: First things first, upgrade from 13 - WebRTC has moved a long a lot since then. If you can't upgrade everything to 13 then run another asterisk specifically for WebRTC and bridge to your other Asterisk Is this just an audio conference? On Sun, Apr 26, 2020 at 10:21 PM Teijo wrote: Hello, Has somebody get combination Asterisk (I'm using version 13.32.0, webrtc and iOS (version 13.4.1) with Safari or any other browser working properly in confbridge conference calls? I hope my Asterisk webrtc related settings are not totally wrong, because several other browsers from Windows seem to work. 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 -- _ -- 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] Webrtc and iOS devices
Hello, Has somebody get combination Asterisk (I'm using version 13.32.0, webrtc and iOS (version 13.4.1) with Safari or any other browser working properly in confbridge conference calls? I hope my Asterisk webrtc related settings are not totally wrong, because several other browsers from Windows seem to work. 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
[asterisk-users] Webrtc and iOS 11
Hello, I've been using webrtc (Jscommunicator) with Asterisk occasionally. I have understood (not sure from where) that iOS 11 (Safari) would support webrtc. When testing I failed, but as said I'm somewhat unsure about iOS 11 webrtc support and how comprehensive it is. Has anybody more exact information? 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
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 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 :-) 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"
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, 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
[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
[asterisk-users] dtls_rekey
Hello, I'm using Asterisk 13.10.0 (built from source) in Debian Jessie. If I set: dtls_rekey=60 call seems to hang up in Chrome 53.0.2785.80 64-bit beta when rekeying should happen. Log entries in messages.log are like this: [Aug 27 21:54:12] ERROR[20210][C-0028] res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0x7fd5880172b8' due to reason 'sslv3 alert unexpected message', terminating [Aug 27 21:54:12] WARNING[20210][C-0028] res_rtp_asterisk.c: RTP Read error: Unspecified. Hanging up. It may be worth of mentioning that I have: dtls_cipher=HIGH:!SSLv3 If I do not set dtls_rekey which afaik means no rekeying, the problem does not occur. I have also tested with Firefox 50 A2 32-bit, and setting dtls_rekey=60 is not a problem with it. Both browsers were Windows versions. Has somebody encountered this problem? Best, Teijo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference 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 13.10.0 Now Available
21.7.2016, 20:38, Asterisk Development Team kirjoitti: Bugs fixed in this release: --- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by Alexander Traud) Now it's possible to use dtls_cipher settings such like: dtls_cipher=ALL:!SSLv3 or dtls_cipher=HIGH:!SSLv3 Thank you! Best, Teijo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtls_cipher
Hello, After updating OpenSSL packages from version 1.0.1f to 1.0.2h I can get audio also with Chrome when setting dtls_cipher=ALL. Why dtls_cipher=ALL:!SSLv3 does not work? Best, Teijo 9.7.2016, 18:37, Teijo kirjoitti: Hello, I'm using Asterisk 13.9.1 (compiled from source) in Ubuntu 14.04.4. I'm testing webrtc with jscommunicator (www.jscommunicator.org), and I'm using wss. My problem is as follows: If I set dtls_cipher=ALL I can get audio with Firefox 47.0.1, but with Chrome 52.0.2743.60 beta, I just get log entry in messages.log telling about SSLv3 handshake failure; I'm not sure, but I suppose that I might get similar results with other Chrome versions. If I set dtls_cipher=ALL:!SSLv3 connection is established with both browsers, but there is no audio. At the moment I have no idea what to try next. Best, Teijo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dtls_cipher
Hello, I'm using Asterisk 13.9.1 (compiled from source) in Ubuntu 14.04.4. I'm testing webrtc with jscommunicator (www.jscommunicator.org), and I'm using wss. My problem is as follows: If I set dtls_cipher=ALL I can get audio with Firefox 47.0.1, but with Chrome 52.0.2743.60 beta, I just get log entry in messages.log telling about SSLv3 handshake failure; I'm not sure, but I suppose that I might get similar results with other Chrome versions. If I set dtls_cipher=ALL:!SSLv3 connection is established with both browsers, but there is no audio. At the moment I have no idea what to try next. Best, Teijo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
