[asterisk-users] WebRTC - Transport Issues.
BryantZ at zktech.com
Sat Mar 11 18:52:35 CST 2017
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
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.
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