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.

CLAUDE.md

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+# Project Notes
+
+## Architecture: Hub Model (Call as Centerpiece)
+
+All call logic lives in `ts/call/`. The Call is the central entity with N legs.
+
+### Key Files
+- `ts/call/call-manager.ts` — singleton registry, factory methods, SIP routing
+- `ts/call/call.ts` — the hub: owns legs, media forwarding
+- `ts/call/sip-leg.ts` — SIP device/provider connection (wraps SipDialog)
+- `ts/call/webrtc-leg.ts` — browser WebRTC connection (wraps werift PeerConnection)
+- `ts/call/rtp-port-pool.ts` — unified RTP port pool
+- `ts/sipproxy.ts` — thin bootstrap wiring everything together
+- `ts/webrtcbridge.ts` — browser device registration (signaling only)
+
+### WebRTC Browser Call Flow (Critical)
+
+The browser call flow has a specific signaling order that MUST be followed:
+
+1. `POST /api/call` with browser deviceId → CallManager creates Call, saves pending state, notifies browser via `webrtc-incoming`
+2. Browser sends `webrtc-offer` (with its own `sessionId`) → CallManager creates a **standalone** WebRtcLeg (NOT attached to any call yet)
+3. Browser sends `webrtc-accept` (with `callId` + `sessionId`) → CallManager links the standalone WebRtcLeg to the Call, then starts the SIP provider leg
+
+**The WebRtcLeg CANNOT be created at call creation time** because the browser's session ID is unknown until the `webrtc-offer` arrives.
+
+### WebRTC Audio Return Channel (Critical)
+
+The SIP→browser audio path works through the Call hub:
+
+1. Provider sends RTP to SipLeg's socket
+2. SipLeg's `onRtpReceived` fires → Call hub's `forwardRtp`
+3. Call hub calls `webrtcLeg.sendRtp(data)` → which calls `forwardToBrowser()`
+4. `forwardToBrowser` transcodes (G.722→Opus) and sends via `sender.sendRtp()` (WebRTC PeerConnection)
+
+**`WebRtcLeg.sendRtp()` MUST feed into `forwardToBrowser()`** (the WebRTC PeerConnection path), NOT send to a UDP address. This was a bug that caused one-way audio.
+
+The browser→SIP direction works independently: `ontrack.onReceiveRtp` → `forwardToSip()` → transcodes → sends directly to provider's media endpoint via UDP.
+
+### SIP Protocol Library
+
+`ts/sip/` is a zero-dependency SIP protocol library. Do not add transport or timer logic there — it's purely data-level (parse/build/mutate/serialize).