[asterisk-users] WebRTC - Transport Issues.

[Bryant Zimmerman]

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 at 192.0.2.35;transport=wss
  Any Insights would be appreciated.
  
 Thanks
 Bryant

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