[asterisk-users] WebRTC - Transport Issues. - Solved

[Bryant Zimmerman]

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" <jcolp at digium.com>
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.
   

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