Summary

Add a WebRTC signaling server to JSS. The server handles matchmaking between peers — the actual media/data flows peer-to-peer via the browser's built-in WebRTC API.

Phased approach

1. Signaling server — WebSocket route that relays SDP offers/answers and ICE candidates between authenticated users
2. TURN relay — fallback for firewalled users who can't connect directly
3. Multi-party / rooms — group calls
4. Call history — store session metadata in /db/
5. Data channels — peer-to-peer file transfer
6. WebTorrent — seed pod files via WebRTC-based torrents

Phase 1 scope

• New WebSocket route (/.webrtc or similar)
• Relay SDP offers/answers between two WebID-authenticated users
• Forward ICE candidates
• Simple call model: caller targets a callee's WebID
• Similar in scope to the existing nostr relay plugin (~100-150 lines)

Why JSS is a good fit

• WebSocket infrastructure already exists (notifications plugin)
• WebID authentication gives identity for free
• WAC can control who can call whom
• /db/ route available for call history/metadata
• Pod profiles (foaf:knows) provide natural contact lists