[asterisk-users] WebRTC - Transport Issues. - Solved
BryantZ at zktech.com
Mon Mar 13 08:43:51 CDT 2017
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.
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
> with TLS now working.
> At least I hope it is using TLS and wss. Based on what I am seeing I
> 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
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.
-------------- next part --------------
An HTML attachment was scrubbed...
More information about the asterisk-users