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/index.ts

@@ -0,0 +1,12 @@
+export type { TCallState, TLegState, TLegType, TCallDirection, ICallStatus, ILegStatus, ICallHistoryEntry } from './types.ts';
+export type { ILeg } from './leg.ts';
+export { rtpClockIncrement, buildRtpHeader, codecDisplayName } from './leg.ts';
+export { RtpPortPool } from './rtp-port-pool.ts';
+export type { IRtpAllocation } from './rtp-port-pool.ts';
+export { SipLeg } from './sip-leg.ts';
+export type { ISipLegConfig } from './sip-leg.ts';
+export { WebRtcLeg } from './webrtc-leg.ts';
+export type { IWebRtcLegConfig } from './webrtc-leg.ts';
+export { Call } from './call.ts';
+export { CallManager } from './call-manager.ts';
+export type { ICallManagerConfig } from './call-manager.ts';