Implement WebRTC real-time call analysis with security hardening (FRE-4497)
- signaling-server.ts: JWT auth, origin validation, JSON schema validation,
crypto.randomBytes peer IDs, message size limits, idle timeout, graceful shutdown
- alert-server.ts: JWT auth enabled by default, non-empty jwtSecret from env,
origin allowlist, per-subscriber callId filtering, bounded alert history with TTL,
alert cooldown, graceful shutdown with timeout
- call-analysis-engine.ts: Bounded eventBuffer/anomalyBuffer with FIFO eviction,
real quality metrics from signal properties, configurable buffer sizes
- audio-stream-capture.ts: Proper destroy() lifecycle with awaited stop(),
AudioWorklet support with ScriptProcessorNode fallback, bounded frame buffers
- Added ws dependency and server tsconfig
Co-Authored-By: Paperclip <noreply@paperclip.ing>
Michael Freno
ec4565f44c