WebRTC ↔ SIP interop checklist (config-aware)
webrtc_sip_checklist[cost: free (pure CPU, no network) | read-only]
Return a curated checklist of WebRTC ↔ SIP requirements (WSS transport, ICE gathering, DTLS-SRTP fingerprint, rtcp-mux + BUNDLE, media relay / rtpengine, STUN/TURN, secure-context Origin allowlist, Opus codec, session-timer behavior across the bridge, STIR/SHAKEN signing). When configText is supplied, each item is marked as 'looks present' or 'check needed' based on simple regex signals.
Use when the user is building a WebRTC ↔ SIP bridge or troubleshooting one (no media, one-way audio, ICE failures).
Pair with: review_sip_config for the full structured outline; search_sip_docs(vendor=...) to ground each unchecked item in vendor docs; parse_sdp / compare_sdp_offer_answer when the bug is in SDP negotiation.
Input Schema
| Name | Required | Description | Default |
|---|---|---|---|
| vendor | No | Vendor slug. Omit and supply `configText` to auto-detect. | |
| configText | No | Optional config blob. When supplied, items with matching signals are marked as present; vendor is auto-detected if not supplied. | |
| filenameHint | No | Optional filename ("kamailio.cfg", etc.) to bias auto-detection. |