I have a WebRTC soft-phone built with JsSIP that needs to register to an Asterisk 18 server over WSS. SIP credentials are confirmed correct, yet the browser console shows an authentication failure. The signalling path is protected with TLS certificates, so the problem is somewhere in the certificate handling or the way Asterisk presents the challenge. Your job is to trace and eliminate the registration failure, then hand back a clean configuration and proof that the client can successfully register and place a test call. Environment details you will touch: – Asterisk 18 (pjsip stack enabled) – JsSIP running in a standard browser (wss://) – TLS with server and client certificates already issued Acceptance criteria: • JsSIP completes REGISTER without 401/403 responses • Call set-up (INVITE/200 OK/ACK) proceeds and audio flows both ways • Updated pjsip.conf/http.conf, any certificate tweaks, and a short README explaining each change If you need temporary access, SSH can be provided through a jump box, or you can guide me step-by-step over a screen-share. Let me know your preferred approach and an estimated turnaround time so we can get this resolved quickly.