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/tsconfig.json

@@ -0,0 +1,17 @@
+{
+  "compilerOptions": {
+    "target": "ES2022",
+    "module": "NodeNext",
+    "moduleResolution": "NodeNext",
+    "lib": ["ES2022"],
+    "outDir": "./dist",
+    "rootDir": ".",
+    "strict": true,
+    "esModuleInterop": true,
+    "skipLibCheck": true,
+    "forceConsistentCasingInFileNames": true,
+    "declaration": true
+  },
+  "include": ["**/*.ts"],
+  "exclude": ["node_modules", "dist"]
+}