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