Prepare distinct server services
Asterisk's documented WebRTC arrangement uses its HTTP/WebSocket service together with a PJSIP WebSocket transport and endpoint, authentication and AOR objects. The browser endpoint also needs compatible ICE and DTLS-SRTP settings. The official tutorial contains older example versions; use its architecture with the actual Asterisk 22 configuration reference and installed modules. Use a certificate the browser can validate. A self-signed example is not a reason to ask ordinary users to ignore certificate errors. Keep HTTP management interfaces restricted when exposing a WebSocket calling endpoint.
Validate in layers
First confirm the HTTPS listener and WebSocket URI are available to the intended client. Next establish WSS, then authenticate the endpoint and verify its contact. Only after those work should you debug an INVITE and its negotiated media. - Record the endpoint's actual allowed codecs and WebRTC settings. - Inspect ICE state and the chosen address pair on an external network. - Confirm DTLS completes and two-way media counters increase. - Test hold, DTMF, transfer and hangup through the intended dialplan. Do not paste standalone Asterisk configuration into FreePBX-generated files. A FreePBX deployment needs the supported controls and extension points for that installation.
Keep a reproducible integration record
Record Asterisk version, loaded modules, browser version, web client release, proxy configuration and certificate identity. Save redacted configuration fragments and observed results separately from assumptions. If Opus transcoding is required, verify the actual codec module and capacity rather than equating pass-through negotiation with transcoding support. For a failure, identify the first unsuccessful stage and preserve both browser diagnostics and Asterisk signaling evidence. A successful registration screenshot is insufficient evidence of a working browser calling service.
Sources & applicability
Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.
Examples require adaptation to your topology. No live PBX or hardware testing is claimed.