We have an Asterisk v15 based Call Center. Call comes to Asterisk-based IVR server and transfers to the asterisk servers . End points are web based webrtc using sip.js
All calls connected to IVR ,IVR prompts successfully. When call connects to an agent, some calls (around %20) Callling party getting just big loud noise and agent can not hear anything. We checked settings about codec etc. All seems fine. it is seems like related to TLS and not able to decode rtp time to time or we have some other issue like stun, ice config etc. We could not able to figure out.
All WebRTC Clients and Asterisk Servers on the same network.
We will be very happy for your idea and help.