WebRTC – Transport Issues.

Home » Asterisk Users » WebRTC – Transport Issues.
Asterisk Users 1 Comment

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?

Note: this concise hands-on guide shows you how to use the emerging Web Real-Time Communication (WebRtc) technology to build a browser-to-browser application, piece by piece.

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@;transport=wss

Any Insights would be appreciated

One thought on - WebRTC – Transport Issues.

  • 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.