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>

server/package.json

@@ -0,0 +1,6 @@
+{
+  "dependencies": {
+    "@types/ws": "^8.18.1",
+    "ws": "^8.20.0"
+  }
+}