Re: [asterisk-users] WebRTC - Transport Issues. - Solved
Josh Thank you for the confirmation on this. The captures do confirm that I am using the wss. What was throwing me was I have only udp and wss in the transports and then the Primary once connected was showing the ws. At first I thought I was doing something wrong and the traffic was flowing unencrypted. You confirmed what I had hoped that the wss was just showing the underlying ws transport. A big thanks. We are excited to finally getting our webrtc test application out to some customers. Have a great week. Bryant From: "Joshua Colp" Sent: Sunday, March 12, 2017 7:35 PM On Sat, Mar 11, 2017, at 09:52 PM, Bryant Zimmerman wrote: > Hey all. I have webrtc up and running with asterisk 11. All is going well > with TLS now working. > At least I hope it is using TLS and wss. Based on what I am seeing I > have > UDP, WSS listed in the Allowed transports, but every time I connect the > Primary transport shows WS.. Why is this? Am I actually running ws in > wss > mode? You are using WSS (the Contact line has transport=wss which indicates it). Both WS and WSS will show "WS" for the Primary Transport. Another way to tell is to look at the SIP traffic and check the Via header for WSS. You can also check a packet capture. -- _ -- 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 - Transport Issues.
On Sat, Mar 11, 2017, at 09:52 PM, Bryant Zimmerman wrote: > Hey all. I have webrtc up and running with asterisk 11. All is going well > with TLS now working. > At least I hope it is using TLS and wss. Based on what I am seeing I > have > UDP, WSS listed in the Allowed transports, but every time I connect the > Primary transport shows WS.. Why is this? Am I actually running ws in > wss > mode? You are using WSS (the Contact line has transport=wss which indicates it). Both WS and WSS will show "WS" for the Primary Transport. Another way to tell is to look at the SIP traffic and check the Via header for WSS. You can also check a packet capture. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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 - Transport Issues.
Hey all. I have webrtc up and running with asterisk 11. All is going well with TLS now working. At least I hope it is using TLS and wss. Based on what I am seeing I have UDP, WSS listed in the Allowed transports, but every time I connect the Primary transport shows WS.. Why is this? Am I actually running ws in wss mode? Prim.Transp. : WS Allowed.Trsp : UDP,WSS Def. Username: 6167761066.2011 SIP Options : (none) Codecs : (ulaw) Codec Order : (ulaw:20) Auto-Framing : No Status : OK (71 ms) Useragent: SIP.js/0.7.7 Reg. Contact : sip:fed97qgu@192.0.2.35;transport=wss Any Insights would be appreciated. Thanks Bryant -- _ -- 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
