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/call/leg.ts

@@ -0,0 +1,104 @@
+/**
+ * ILeg interface — abstract connection from a Call hub to an endpoint.
+ *
+ * Concrete implementations: SipLeg (SIP devices + providers) and WebRtcLeg (browsers).
+ * Shared RTP utilities (header building, clock rates) are also defined here.
+ */
+
+import { Buffer } from 'node:buffer';
+import type dgram from 'node:dgram';
+import type { IEndpoint } from '../sip/index.ts';
+import type { TLegState, TLegType, ILegStatus } from './types.ts';
+import type { IRtpTranscoder } from '../codec.ts';
+import type { SipDialog } from '../sip/index.ts';
+import type { SipMessage } from '../sip/index.ts';
+
+// ---------------------------------------------------------------------------
+// ILeg interface
+// ---------------------------------------------------------------------------
+
+export interface ILeg {
+  readonly id: string;
+  readonly type: TLegType;
+  state: TLegState;
+
+  /** The SIP Call-ID used by this leg (for CallManager routing). */
+  readonly sipCallId: string;
+
+  /** Where this leg sends/receives RTP. */
+  readonly rtpPort: number | null;
+  readonly rtpSock: dgram.Socket | null;
+  remoteMedia: IEndpoint | null;
+
+  /** Negotiated codec payload type (e.g. 9 = G.722, 111 = Opus). */
+  codec: number | null;
+
+  /** Transcoder for converting to this leg's codec (set by Call when codecs differ). */
+  transcoder: IRtpTranscoder | null;
+
+  /** Packet counters. */
+  pktSent: number;
+  pktReceived: number;
+
+  /** SIP dialog (SipLegs only, null for WebRtcLegs). */
+  readonly dialog: SipDialog | null;
+
+  /**
+   * Send an RTP packet toward this leg's remote endpoint.
+   * If a transcoder is set, the Call should transcode before calling this.
+   */
+  sendRtp(data: Buffer): void;
+
+  /**
+   * Callback set by the owning Call — invoked when this leg receives an RTP packet.
+   * The Call uses this to forward to other legs.
+   */
+  onRtpReceived: ((data: Buffer) => void) | null;
+
+  /**
+   * Handle an incoming SIP message routed to this leg (SipLegs only).
+   * Returns a SipMessage response if one needs to be sent, or null.
+   */
+  handleSipMessage(msg: SipMessage, rinfo: IEndpoint): void;
+
+  /** Release all resources (sockets, peer connections, etc.). */
+  teardown(): void;
+
+  /** Status snapshot for the dashboard. */
+  getStatus(): ILegStatus;
+}
+
+// ---------------------------------------------------------------------------
+// Shared RTP utilities
+// ---------------------------------------------------------------------------
+
+/** RTP clock increment per 20ms frame for each codec. */
+export function rtpClockIncrement(pt: number): number {
+  if (pt === 111) return 960; // Opus: 48000 Hz x 0.02s
+  if (pt === 9) return 160;   // G.722: 8000 Hz x 0.02s (SDP clock rate quirk)
+  return 160;                  // PCMU/PCMA: 8000 Hz x 0.02s
+}
+
+/** Build a fresh RTP header with correct PT, timestamp, seq, SSRC. */
+export function buildRtpHeader(pt: number, seq: number, ts: number, ssrc: number, marker: boolean): Buffer {
+  const hdr = Buffer.alloc(12);
+  hdr[0] = 0x80; // V=2
+  hdr[1] = (marker ? 0x80 : 0) | (pt & 0x7f);
+  hdr.writeUInt16BE(seq & 0xffff, 2);
+  hdr.writeUInt32BE(ts >>> 0, 4);
+  hdr.writeUInt32BE(ssrc >>> 0, 8);
+  return hdr;
+}
+
+/** Codec name for status display. */
+export function codecDisplayName(pt: number | null): string | null {
+  if (pt === null) return null;
+  switch (pt) {
+    case 0: return 'PCMU';
+    case 8: return 'PCMA';
+    case 9: return 'G.722';
+    case 111: return 'Opus';
+    case 101: return 'telephone-event';
+    default: return `PT${pt}`;
+  }
+}