[asterisk-users] WebRTC - Transport Issues.

Joshua Colp
Sun, 12 Mar 2017 23:34:30 -0000

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