initial commit — SIP B2BUA + WebRTC bridge with Rust codec engine

Full-featured SIP router with multi-provider trunking, browser softphone
via WebRTC, real-time Opus/G.722/PCM transcoding in Rust, RNNoise ML
noise suppression, Kokoro neural TTS announcements, and a Lit-based
web dashboard with live call monitoring and REST API.

ts/codec.ts

@@ -0,0 +1,40 @@
+/**
+ * Audio codec translation layer for bridging between WebRTC and SIP.
+ *
+ * All actual codec work (Opus, G.722, PCMU, PCMA) is done in Rust via
+ * the smartrust bridge. This module provides the RTP-level transcoding
+ * interface used by the webrtcbridge.
+ */
+
+import { Buffer } from 'node:buffer';
+import { transcode, isCodecReady } from './opusbridge.ts';
+
+/** Opus dynamic payload type (standard WebRTC assignment). */
+export const OPUS_PT = 111;
+
+export interface IRtpTranscoder {
+  /** Transcode an RTP payload. Always async (Rust IPC). */
+  payload(data: Buffer): Promise<Buffer>;
+  fromPT: number;
+  toPT: number;
+}
+
+/**
+ * Create a transcoder that converts RTP payloads between two codecs.
+ * Returns null if the codecs are the same or the Rust bridge isn't ready.
+ *
+ * @param sessionId - optional Rust codec session for isolated state per call
+ */
+export function createTranscoder(fromPT: number, toPT: number, sessionId?: string, direction?: string): IRtpTranscoder | null {
+  if (fromPT === toPT) return null;
+  if (!isCodecReady()) return null;
+
+  return {
+    fromPT,
+    toPT,
+    async payload(data: Buffer): Promise<Buffer> {
+      const result = await transcode(data, fromPT, toPT, sessionId, direction);
+      return result || Buffer.alloc(0); // return empty on failure — never pass raw codec bytes
+    },
+  };
+}