Introduce a TURN credentials protocol for requesting time-limited relay
server credentials from the relay node, improving NAT traversal for
WebRTC voice connections.
WebRTC layer improvements:
- Add ICE candidate buffering before remote description is set
- Implement automatic ICE restart with configurable max attempts
- Add disconnection timeout with delayed restart recovery
- Add per-peer connection state tracking and callbacks
- Include public STUN servers (Google, Cloudflare) by default
- Add detailed logging throughout the WebRTC lifecycle
Voice store improvements:
- Fetch TURN credentials from relay before establishing connections
- Track per-peer RTCPeerConnectionState with reactive signals
- Derive overall voice quality signal (good/connecting/degraded/failed)
- Evaluate and surface degraded connection state to the UI
- Clean up peer connection states on participant leave
- Fix video toggle resource leak (stop unused audio tracks)
Backend changes:
- Add TurnCredentialRequest/Response protocol types
- Wire turn_credentials behaviour into swarm and node event loop
- Add get_turn_credentials Tauri command
- Propagate errors from voice commands instead of silently dropping
- Add input validation for SDP types in send_voice_sdp
- Handle missing node handle as explicit error in all voice commands
- Add DirectoryResponse::Error variant handling
- Scope Linux WebKitGTK permission grants to UserMediaPermissionRequest
UI improvements:
- Show connection quality indicator in voice channel header
- Add per-peer connection state ring on participant tiles
- Display failed connection warning icon on participant tiles
- Stack muted/deafened indicators vertically to avoid overlap
- Show participant grid in degraded connection state
Also remove unused imports across several frontend components.
cloudwithax