Hi!
I have a very strange problem (may be not with Asterisk) with incoming calls from SIP to WebRTC clients (based on jssip.net library).
Sometimes (2-5% of all incoming calls) I have no sound (on both sides) on incoming calls.
RTP is going fine in both sides (local network)
If I turn on mixMonitor on Asterisk, I can see only noise in call (looks like a problem with srtp keys, but not sure)
Asterisk 15.3.0, JsSIP 3.2.8, tested in Chrome, Chromium and Firefox
Are those 2-5% coming from a particular device type?
If you are using a Cisco RV325 Router get rid of it. We had this same problem with about the same failure rate and went thru a lot of work to trace it down to these Routers. We replaced the RV325 with the older RV016 and everything is now OK.
After some tests, problem was located in “webrtc encryption”.
–disable-webrtc-encryption (chrome/chromium) solves this problem.
I know, it’s not a good idea, but it works.