v1.25.0

.dockerignore

@@ -0,0 +1,16 @@
+node_modules/
+.nogit/
+nogit/
+.git/
+.playwright-mcp/
+.vscode/
+test/
+dist_rust/
+dist_ts_web/
+rust/target/
+sip_trace.log
+sip_trace_*.log
+proxy.out
+proxy_v2.out
+*.pid
+.server.pid

.gitea/workflows/docker_tags.yaml

@@ -0,0 +1,32 @@
+name: Docker (tags)
+
+on:
+  push:
+    tags:
+      - '*'
+
+env:
+  IMAGE: code.foss.global/host.today/ht-docker-node:dbase_dind
+  NPMCI_LOGIN_DOCKER_GITEA: ${{ github.server_url }}|${{ gitea.repository_owner }}|${{ secrets.GITEA_TOKEN }}
+  NPMCI_LOGIN_DOCKER_DOCKERREGISTRY: ${{ secrets.NPMCI_LOGIN_DOCKER_DOCKERREGISTRY }}
+
+jobs:
+  release:
+    if: github.event_name == 'push' && startsWith(github.ref, 'refs/tags/')
+    runs-on: ubuntu-latest
+    container:
+      image: ${{ env.IMAGE }}
+
+    steps:
+      - uses: actions/checkout@v3
+
+      - name: Prepare
+        run: |
+          pnpm install -g pnpm
+          pnpm install -g @git.zone/tsdocker
+
+      - name: Release
+        run: |
+          tsdocker login
+          tsdocker build
+          tsdocker push

.smartconfig.json

@@ -8,5 +8,16 @@
         "production": true
       }
     ]
+  },
+  "@git.zone/tsrust": {
+    "targets": ["linux_amd64", "linux_arm64"]
+  },
+  "@git.zone/tsdocker": {
+    "registries": ["code.foss.global"],
+    "registryRepoMap": {
+      "code.foss.global": "serve.zone/siprouter",
+      "dockerregistry.lossless.digital": "serve.zone/siprouter"
+    },
+    "platforms": ["linux/amd64", "linux/arm64"]
   }
 }

CLAUDE.md

@@ -1,41 +1,103 @@
 # Project Notes
 
-## Architecture: Hub Model (Call as Centerpiece)
+## Architecture: Hub Model in Rust (Call as Centerpiece)
 
-All call logic lives in `ts/call/`. The Call is the central entity with N legs.
+The call hub lives in the Rust proxy-engine (`rust/crates/proxy-engine/`). TypeScript is the **control plane only** — it configures the engine, sends high-level commands (`hangup`, `make_call`, `webrtc_offer`, etc.), and receives events (`incoming_call`, `call_answered`, `device_registered`, `webrtc_audio_rx`, …). No raw SIP/RTP ever touches TypeScript.
 
-### 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)
+The `Call` is still the central entity: it owns N legs and a central mixer task that provides mix-minus audio to all participants. Legs can be `SipProvider`, `SipDevice`, `WebRtc`, or `Tool` (recording/transcription observer).
 
-### WebRTC Browser Call Flow (Critical)
+### Key Rust files (`rust/crates/proxy-engine/src/`)
 
-The browser call flow has a specific signaling order that MUST be followed:
+- `call_manager.rs` — singleton registry, call factory methods, SIP routing (inbound/outbound/passthrough), B2BUA state machine, inbound route resolution
+- `call.rs` — the `Call` hub + `LegInfo` struct, owns legs and the mixer task
+- `sip_leg.rs` — full SIP dialog management for B2BUA legs (INVITE, 407 auth retry, BYE, CANCEL, early media)
+- `rtp.rs` — RTP port pool (uses `Weak<UdpSocket>` so calls auto-release ports on drop) + RTP header helpers
+- `mixer.rs` — 20 ms-tick mix-minus engine (48 kHz f32 internal, per-leg transcoding via `codec-lib`, per-leg denoising)
+- `jitter_buffer.rs` — per-leg reordering/packet-loss compensation
+- `leg_io.rs` — spawns inbound/outbound RTP I/O tasks per SIP leg
+- `webrtc_engine.rs` — browser WebRTC sessions (werift-rs based), ICE/DTLS/SRTP
+- `provider.rs` — SIP trunk registrations, public-IP detection via Via `received=`
+- `registrar.rs` — accepts REGISTER from SIP phones, tracks contacts (push-based device status)
+- `config.rs` — `AppConfig` deserialized from TS, route resolvers (`resolve_outbound_route`, `resolve_inbound_route`)
+- `main.rs` — IPC command dispatcher (`handle_command`), event emitter, top-level SIP packet router
+- `sip_transport.rs` — owning wrapper around the main SIP UDP socket
+- `voicemail.rs` / `recorder.rs` / `audio_player.rs` / `tts.rs` — media subsystems
+- `tool_leg.rs` — per-source observer audio for recording/transcription tools
+- `ipc.rs` — event-emission helper used throughout
 
-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
+### Key TS files (control plane)
 
-**The WebRtcLeg CANNOT be created at call creation time** because the browser's session ID is unknown until the `webrtc-offer` arrives.
+- `ts/sipproxy.ts` — entrypoint, wires the proxy engine bridge + web UI + WebRTC signaling
+- `ts/proxybridge.ts` — `@push.rocks/smartrust` bridge to the Rust binary, typed `TProxyCommands` map
+- `ts/config.ts` — JSON config loader (`IAppConfig`, `IProviderConfig`, etc.), sent to Rust via `configure`
+- `ts/voicebox.ts` — voicemail metadata persistence (WAV files live in `.nogit/voicemail/{boxId}/`)
+- `ts/webrtcbridge.ts` — browser WebSocket signaling, browser device registry (`deviceIdToWs`)
+- `ts/call/prompt-cache.ts` — the only remaining file under `ts/call/` (IVR prompt caching)
 
-### WebRTC Audio Return Channel (Critical)
+### Rust SIP protocol library
 
-The SIP→browser audio path works through the Call hub:
+`rust/crates/sip-proto/` is a zero-dependency SIP data library (parse/build/mutate/serialize messages, dialog management, SDP helpers, digest auth). Do not add transport or timer logic there — it's purely data-level.
 
-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)
+## Event-push architecture for device status
 
-**`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.
+Device status flows **via push events**, not pull-based IPC queries:
 
-The browser→SIP direction works independently: `ontrack.onReceiveRtp` → `forwardToSip()` → transcodes → sends directly to provider's media endpoint via UDP.
+1. Rust emits `device_registered` when a phone REGISTERs
+2. TS `sipproxy.ts` maintains a `deviceStatuses` Map, updated from the event
+3. Map snapshot goes into the WebSocket `status` broadcast
+4. Web UI (`ts_web/elements/sipproxy-devices.ts`) reads it from the push stream
 
-### SIP Protocol Library
+There used to be a `get_status` pull IPC for this, but it was never called from TS and has been removed. If a new dashboard ever needs a pull-based snapshot, the push Map is the right source to read from.
 
-`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).
+## Inbound routing (wired in Commit 4 of the cleanup PR)
+
+Inbound route resolution goes through `config.resolve_inbound_route(provider_id, called_number, caller_number)` inside `create_inbound_call` (call_manager.rs). The result carries a `ring_browsers` flag that propagates to the `incoming_call` event; `ts/sipproxy.ts` gates the `webrtc-incoming` browser fan-out behind that flag.
+
+**Known limitations / TODOs** (documented in code at `create_inbound_call`):
+- Multi-target inbound fork is not yet implemented — only the first registered device from `route.device_ids` is rung.
+- `ring_browsers` is **informational only**: browsers see a toast but do not race the SIP device to answer. True first-to-answer-wins requires a multi-leg fork + per-leg CANCEL, which is not built yet.
+- `voicemail_box`, `ivr_menu_id`, `no_answer_timeout` are resolved but not yet honored downstream.
+
+## WebRTC Browser Call Flow (Critical)
+
+The browser call signaling order is strict:
+
+1. Browser initiates outbound via a TS API (e.g. `POST /api/call`) — TS creates a pending call in the Rust engine via `make_call` and notifies the browser with a `webrtc-incoming` push.
+2. Browser sends `webrtc-offer` (with its own `sessionId`) → Rust `handle_webrtc_offer` creates a **standalone** WebRTC session (NOT attached to any call yet).
+3. Browser sends `webrtc_link` (with `callId` + `sessionId`) → Rust links the standalone session to the Call and wires the WebRTC leg through the mixer.
+
+**The WebRTC leg cannot be fully attached 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 goes through the mixer, not a direct RTP relay:
+
+1. Provider sends RTP → received on the provider leg's UDP socket (`leg_io::spawn_sip_inbound`)
+2. Packet flows through `jitter_buffer` → mixer's inbound mpsc channel
+3. Mixer decodes/resamples/denoises, computes mix-minus per leg
+4. WebRTC leg receives its mix-minus frame, encodes to Opus, and pushes via the WebRTC engine's peer connection sender
+
+Browser→SIP works symmetrically: `ontrack.onReceiveRtp` → WebRTC leg's outbound mpsc → mixer → other legs' inbound channels.
+
+## SDP/Record-Route NAT (fixed in Commit 3 of the cleanup PR)
+
+The proxy tracks a `public_ip: Option<String>` on every `LegInfo` (populated from provider-leg construction sites). When `route_passthrough_message` rewrites SDP (`c=` line) or emits a `Record-Route`, it picks `advertise_ip` based on the destination leg's kind:
+
+- `SipProvider` → `other.public_ip.unwrap_or(lan_ip)` (provider reaches us via public IP)
+- `SipDevice` / `WebRtc` / `Tool` / `Media` → `lan_ip` (everything else is LAN or proxy-internal)
+
+This fixed a real NAT-traversal bug where the proxy advertised its RFC1918 LAN IP to the provider in SDP, causing one-way or no audio for device-originated inbound traffic behind NAT.
+
+## Build & development
+
+- **Build:** `pnpm run buildRust` (never `cargo build` directly — tsrust cross-compiles for both `x86_64-unknown-linux-gnu` and `aarch64-unknown-linux-gnu`)
+- **Cross-compile setup:** the aarch64 target requires `gcc-aarch64-linux-gnu` + `libstdc++6-arm64-cross` (Debian/Ubuntu). See `rust/.cargo/config.toml` for the linker wiring. A committed symlink at `rust/.cargo/crosslibs/aarch64/libstdc++.so` → `/usr/aarch64-linux-gnu/lib/libstdc++.so.6` avoids needing the `libstdc++-13-dev-arm64-cross` package.
+- **Bundle web UI:** `pnpm run bundle` (esbuild, output: `dist_ts_web/bundle.js`)
+- **Full build:** `pnpm run build` (= `buildRust && bundle`)
+- **Start server:** `pnpm run start` (runs `tsx ts/sipproxy.ts`)
+
+## Persistent files
+
+- `.nogit/config.json` — app config (providers, devices, routes, voiceboxes, IVR menus)
+- `.nogit/voicemail/{boxId}/` — voicemail WAV files + `messages.json` index
+- `.nogit/prompts/` — cached TTS prompts for IVR menus

Dockerfile

@@ -0,0 +1,74 @@
+# gitzone dockerfile_service
+## STAGE 1 // BUILD
+FROM code.foss.global/host.today/ht-docker-node:lts AS build
+
+# System build tools that the Rust dep tree needs beyond the base image:
+#   - cmake      : used by the `cmake` crate (transitive via ort_sys / a webrtc
+#                  sub-crate) to build a C/C++ library from source when a
+#                  prebuilt-binary download path doesn't apply.
+#   - pkg-config : used by audiopus_sys and other *-sys crates to locate libs
+#                  on the native target (safe no-op if they vendor their own).
+# These are normally pre-installed on dev machines but not in ht-docker-node:lts.
+RUN apt-get update && apt-get install -y --no-install-recommends \
+      cmake \
+      pkg-config \
+ && rm -rf /var/lib/apt/lists/*
+
+# buildx sets TARGETARCH automatically for each platform it's building:
+#   linux/amd64  -> TARGETARCH=amd64
+#   linux/arm64  -> TARGETARCH=arm64
+# We use it to tell tsrust to build ONLY the current container's arch. This
+# overrides the `@git.zone/tsrust.targets` list in .smartconfig.json, which is
+# right for local dev / CI (where you want both binaries) but wrong for per-
+# platform Docker stages (each stage would then also try to cross-compile to
+# the OTHER arch — which fails in the arm64 stage because no reverse cross-
+# toolchain is installed).
+#
+# With --target set, tsrust builds a single target natively within whichever
+# platform this stage is running under (native on amd64, QEMU-emulated on arm64).
+ARG TARGETARCH
+
+COPY ./ /app
+WORKDIR /app
+RUN pnpm config set store-dir .pnpm-store
+RUN rm -rf node_modules && pnpm install
+
+# tsrust --target takes precedence over .smartconfig.json's targets array.
+# Writes dist_rust/proxy-engine_linux_amd64 or dist_rust/proxy-engine_linux_arm64.
+# The TS layer (ts/proxybridge.ts buildLocalPaths) picks the right one at runtime
+# via process.arch.
+RUN pnpm exec tsrust --target linux_${TARGETARCH}
+
+# Web bundle (esbuild — pure JS, uses the platform's native esbuild binary
+# installed by pnpm above, so no cross-bundling concerns).
+RUN pnpm run bundle
+
+# Drop pnpm store to keep the image smaller. node_modules stays because the
+# runtime entrypoint is tsx and siprouter has no separate dist_ts/ to run from.
+RUN rm -rf .pnpm-store
+
+## STAGE 2 // PRODUCTION
+FROM code.foss.global/host.today/ht-docker-node:alpine-node AS production
+
+# gcompat + libstdc++ let the glibc-linked proxy-engine binary run on Alpine.
+RUN apk add --no-cache gcompat libstdc++
+
+WORKDIR /app
+COPY --from=build /app /app
+
+ENV SIPROUTER_MODE=OCI_CONTAINER
+ENV NODE_ENV=production
+
+LABEL org.opencontainers.image.title="siprouter" \
+      org.opencontainers.image.description="SIP proxy with Rust data plane and WebRTC bridge" \
+      org.opencontainers.image.source="https://code.foss.global/serve.zone/siprouter"
+
+# 5070  SIP signaling (UDP+TCP)
+# 5061  SIP-TLS (optional, UDP+TCP)
+# 3060  Web UI / WebSocket (HTTP or HTTPS, auto-detected from .nogit/cert.pem)
+# 20000-20200/udp  RTP media range (must match config.proxy.rtpPortRange)
+EXPOSE 5070/udp 5070/tcp 5061/udp 5061/tcp 3060/tcp 20000-20200/udp
+
+# exec replaces sh as PID 1 with tsx, so SIGINT/SIGTERM reach Node and
+# ts/sipproxy.ts' shutdown handler (which calls shutdownProxyEngine) runs cleanly.
+CMD ["sh", "-c", "exec ./node_modules/.bin/tsx ts/sipproxy.ts"]

changelog.md

@@ -1,5 +1,90 @@
 # Changelog
 
+## 2026-04-14 - 1.25.0 - feat(proxy-engine)
+add live TTS streaming interactions and incoming number range support
+
+- add a new start_tts_interaction command and bridge API to begin IVR or leg interactions before full TTS rendering completes
+- stream synthesized TTS chunks into the mixer with cancellation handling so prompts can stop cleanly on digit match, leg removal, or shutdown
+- extract PCM-to-mixer frame conversion for reusable live prompt processing
+- extend routing pattern matching to support numeric number ranges like start..end, including + prefixed values
+- add incomingNumbers config typing and frontend config update support for single, range, and regex number modes
+
+## 2026-04-14 - 1.24.0 - feat(routing)
+require explicit inbound DID routes and normalize SIP identities for provider-based number matching
+
+- Inbound route resolution now returns no match unless a configured inbound route explicitly matches the provider and called number.
+- Normalized routing identities were added for SIP/TEL URIs so inbound DIDs and outbound dialed numbers match consistently across provider-specific formats.
+- Call handling and incoming call events now use normalized numbers, improving routing accuracy for shared trunk providers.
+- Route configuration docs and the web route editor were updated to support explicit inbound DID ownership, voicemail fallback, and IVR selection.
+- Mixer RTP handling was enhanced to better support variable packet durations, timestamp-based gap fill, and non-blocking output drop reporting.
+
+## 2026-04-14 - 1.23.0 - feat(runtime)
+refactor runtime state and proxy event handling for typed WebRTC linking and shared status models
+
+- extract proxy event handling into dedicated runtime modules for status tracking and WebRTC session-to-call linking
+- introduce shared typed proxy event and status interfaces used by both backend and web UI
+- update web UI server initialization to use structured options and await async config save hooks
+- simplify browser signaling by routing WebRTC offer/ICE handling through frontend-to-Rust integration
+- align device status rendering with the new address/port fields in dashboard views
+
+## 2026-04-12 - 1.22.0 - feat(proxy-engine)
+add on-demand TTS caching for voicemail and IVR prompts
+
+- Route inbound calls directly to configured IVR menus and track them with a dedicated IVR call state
+- Generate voicemail greetings and IVR menu prompts inside the Rust proxy engine on demand instead of precomputing prompts in TypeScript
+- Add cacheable TTS output with sidecar metadata and enable Kokoro CMUdict support for improved prompt generation
+- Extend proxy configuration to include voiceboxes and IVR menus, and update documentation to reflect Kokoro-only prompt generation
+
+## 2026-04-11 - 1.21.0 - feat(providers)
+replace provider creation modal with a guided multi-step setup flow
+
+- Adds a stepper-based provider creation flow with provider type selection, connection, credentials, advanced settings, and review steps.
+- Applies built-in templates for Sipgate and O2/Alice from the selected provider type instead of separate add actions.
+- Adds a final review step with generated provider ID preview and duplicate ID collision handling before saving.
+
+## 2026-04-11 - 1.20.5 - fix(readme)
+improve architecture and call flow documentation with Mermaid diagrams
+
+- Replace ASCII architecture and audio pipeline diagrams with Mermaid diagrams for better readability
+- Document the WebRTC browser call setup sequence, including offer handling and session-to-call linking
+
+## 2026-04-11 - 1.20.4 - fix(deps)
+bump @design.estate/dees-catalog to ^3.71.1
+
+- Updates the @design.estate/dees-catalog dependency from ^3.70.0 to ^3.71.1 in package.json.
+
+## 2026-04-11 - 1.20.3 - fix(ts-config,proxybridge,voicebox)
+align voicebox config types and add missing proxy bridge command definitions
+
+- Reuses the canonical IVoiceboxConfig type from voicebox.ts in config.ts to eliminate duplicated type definitions and optionality mismatches.
+- Makes voicemail timing and limits optional in voicebox config so defaults can be applied consistently during initialization.
+- Adds VoiceboxManager.addMessage and updates recording handling to use it directly for persisted voicemail metadata.
+- Extends proxy bridge command typings with add_leg, remove_leg, and WebRTC signaling commands, and tightens sendCommand typing.
+
+## 2026-04-11 - 1.20.2 - fix(proxy-engine)
+fix inbound route browser ringing and provider-facing SDP advertisement while preventing RTP port exhaustion
+
+- Honor inbound routing `ringBrowsers` when emitting incoming call events so browser toast notifications can be suppressed per route.
+- Rewrite SDP and Record-Route using the destination leg's routable address, using `public_ip` for provider legs and LAN IP for device and internal legs.
+- Store provider leg public IP metadata on legs to support correct per-destination SIP message rewriting.
+- Change the RTP port pool to track sockets with `Weak<UdpSocket>` so ports are reclaimed automatically after calls end, avoiding leaked allocations and eventual 503 failures on new calls.
+- Remove unused dashboard/status, DTMF, relay, and transport helper code paths as part of engine cleanup. 
+
+## 2026-04-11 - 1.20.1 - fix(docker)
+install required native build tools for Rust dependencies in the build image
+
+- Add cmake and pkg-config to the Docker build stage so Rust native dependencies can compile successfully in the container
+- Document why these tools are needed for transitive Rust crates that build or detect native libraries
+
+## 2026-04-11 - 1.20.0 - feat(docker)
+add multi-arch Docker build and tagged release pipeline
+
+- Add a production Dockerfile for building and running the SIP router with the Rust proxy engine and web bundle
+- Configure tsdocker and tsrust for linux/amd64 and linux/arm64 image builds and registry mapping
+- Add a tag-triggered Gitea workflow to build and push Docker images
+- Update runtime binary resolution to load architecture-specific Rust artifacts in Docker and CI environments
+- Add Docker-related package scripts, dependency updates, and ignore rules for container builds
+
 ## 2026-04-11 - 1.19.2 - fix(web-ui)
 normalize lucide icon names across SIP proxy views
 

package.json

@@ -1,23 +1,28 @@
 {
   "name": "siprouter",
-  "version": "1.19.2",
+  "version": "1.25.0",
   "private": true,
   "type": "module",
   "scripts": {
     "bundle": "node node_modules/.pnpm/esbuild@0.27.7/node_modules/esbuild/bin/esbuild ts_web/index.ts --bundle --format=esm --outfile=dist_ts_web/bundle.js --platform=browser --target=es2022 --minify",
     "buildRust": "tsrust",
+    "build": "pnpm run buildRust && pnpm run bundle",
+    "build:docker": "tsdocker build --verbose",
+    "release:docker": "tsdocker push --verbose",
     "start": "tsx ts/sipproxy.ts",
     "restartBackground": "pnpm run buildRust && pnpm run bundle; test -f .server.pid && kill $(cat .server.pid) 2>/dev/null; sleep 1; rm -f sip_trace.log proxy.out && nohup tsx ts/sipproxy.ts > proxy.out 2>&1 & echo $! > .server.pid; sleep 2; cat proxy.out"
   },
   "dependencies": {
-    "@design.estate/dees-catalog": "^3.70.0",
+    "@design.estate/dees-catalog": "^3.77.0",
     "@design.estate/dees-element": "^2.2.4",
     "@push.rocks/smartrust": "^1.3.2",
     "@push.rocks/smartstate": "^2.3.0",
+    "tsx": "^4.21.0",
     "ws": "^8.20.0"
   },
   "devDependencies": {
     "@git.zone/tsbundle": "^2.10.0",
+    "@git.zone/tsdocker": "^2.2.4",
     "@git.zone/tsrust": "^1.3.2",
     "@git.zone/tswatch": "^3.3.2",
     "@types/ws": "^8.18.1"

readme.md

@@ -20,7 +20,7 @@ siprouter sits between your SIP trunk providers and your endpoints — hardware
 - 🎯 **Adaptive Jitter Buffer** — Per-leg jitter buffering with sequence-based reordering, adaptive depth (60–120ms), Opus PLC for lost packets, and hold/resume detection
 - 📧 **Voicemail** — Configurable voicemail boxes with TTS greetings, recording, and web playback
 - 🔢 **IVR Menus** — DTMF-navigable interactive voice response with nested menus, routing actions, and custom prompts
-- 🗣️ **Neural TTS** — Kokoro-powered announcements and greetings with 25+ voice presets, backed by espeak-ng fallback
+- 🗣️ **Neural TTS** — Kokoro-powered greetings and IVR prompts with 25+ voice presets
 - 🎙️ **Call Recording** — Per-source separated WAV recording at 48kHz via tool legs
 - 🖥️ **Web Dashboard** — Real-time SPA with 9 views: live calls, browser phone, routing, voicemail, IVR, contacts, providers, and streaming logs
 
@@ -28,39 +28,26 @@ siprouter sits between your SIP trunk providers and your endpoints — hardware
 
 ## 🏗️ Architecture
 
-```
-┌─────────────────────────────────────┐
-│           Browser Softphone          │
-│     (WebRTC via WebSocket signaling) │
-└──────────────┬──────────────────────┘
-               │ Opus/WebRTC
-               ▼
-┌──────────────────────────────────────┐
-│            siprouter                  │
-│                                      │
-│  TypeScript Control Plane            │
-│  ┌────────────────────────────────┐  │
-│  │ Config · WebRTC Signaling      │  │
-│  │ REST API · Web Dashboard       │  │
-│  │ Voicebox Manager · TTS Cache   │  │
-│  └────────────┬───────────────────┘  │
-│          JSON-over-stdio IPC         │
-│  ┌────────────┴───────────────────┐  │
-│  │ Rust proxy-engine (data plane) │  │
-│  │                                │  │
-│  │ SIP Stack · Dialog SM · Auth   │  │
-│  │ Call Manager · N-Leg Mixer     │  │
-│  │ 48kHz f32 Bus · Jitter Buffer  │  │
-│  │ Codec Engine · RTP Port Pool   │  │
-│  │ WebRTC Engine · Kokoro TTS     │  │
-│  │ Voicemail · IVR · Recording    │  │
-│  └────┬──────────────────┬────────┘  │
-└───────┤──────────────────┤───────────┘
-        │                  │
- ┌──────┴──────┐    ┌──────┴──────┐
- │ SIP Devices │    │ SIP Trunk   │
- │ (HT801 etc) │    │ Providers   │
- └─────────────┘    └─────────────┘
+```mermaid
+flowchart TB
+    Browser["🌐 Browser Softphone<br/>(WebRTC via WebSocket signaling)"]
+    Devices["📞 SIP Devices<br/>(HT801, desk phones, ATAs)"]
+    Trunks["☎️ SIP Trunk Providers<br/>(sipgate, easybell, …)"]
+
+    subgraph Router["siprouter"]
+        direction TB
+        subgraph TS["TypeScript Control Plane"]
+            TSBits["Config · WebRTC Signaling<br/>REST API · Web Dashboard<br/>Voicebox Manager · TTS Cache"]
+        end
+        subgraph Rust["Rust proxy-engine (data plane)"]
+            RustBits["SIP Stack · Dialog SM · Auth<br/>Call Manager · N-Leg Mixer<br/>48kHz f32 Bus · Jitter Buffer<br/>Codec Engine · RTP Port Pool<br/>WebRTC Engine · Kokoro TTS<br/>Voicemail · IVR · Recording"]
+        end
+        TS <-->|"JSON-over-stdio IPC"| Rust
+    end
+
+    Browser <-->|"Opus / WebRTC"| TS
+    Rust <-->|"SIP / RTP"| Devices
+    Rust <-->|"SIP / RTP"| Trunks
 ```
 
 ### 🧠 Key Design Decisions
@@ -71,6 +58,37 @@ siprouter sits between your SIP trunk providers and your endpoints — hardware
 - **Per-Session Codec Isolation** — Each call leg gets its own encoder/decoder/resampler/denoiser state — no cross-call corruption.
 - **SDP Codec Negotiation** — Outbound encoding uses the codec actually negotiated in SDP answers, not just the first offered codec.
 
+### 📲 WebRTC Browser Call Flow
+
+Browser calls are set up in a strict three-step dance — the WebRTC leg cannot be attached at call-creation time because the browser's session ID is only known once the SDP offer arrives:
+
+```mermaid
+sequenceDiagram
+    participant B as Browser
+    participant TS as TypeScript (sipproxy.ts)
+    participant R as Rust proxy-engine
+    participant P as SIP Provider
+
+    B->>TS: POST /api/call
+    TS->>R: make_call (pending call, no WebRTC leg yet)
+    R-->>TS: call_created
+    TS-->>B: webrtc-incoming (callId)
+
+    B->>TS: webrtc-offer (sessionId, SDP)
+    TS->>R: handle_webrtc_offer
+    R-->>TS: webrtc-answer (SDP)
+    TS-->>B: webrtc-answer
+    Note over R: Standalone WebRTC session<br/>(not yet attached to call)
+
+    B->>TS: webrtc_link (callId + sessionId)
+    TS->>R: link session → call
+    R->>R: wire WebRTC leg through mixer
+    R->>P: SIP INVITE
+    P-->>R: 200 OK + SDP
+    R-->>TS: call_answered
+    Note over B,P: Bidirectional Opus ↔ codec-transcoded<br/>audio flows through the mixer
+```
+
 ---
 
 ## 🚀 Getting Started
@@ -80,7 +98,6 @@ siprouter sits between your SIP trunk providers and your endpoints — hardware
 - **Node.js** ≥ 20 with `tsx` globally available
 - **pnpm** for package management
 - **Rust** toolchain (for building the proxy engine)
-- **espeak-ng** (optional, for TTS fallback)
 
 ### Install & Build
 
@@ -131,24 +148,41 @@ Create `.nogit/config.json`:
   "routing": {
     "routes": [
       {
-        "id": "inbound-default",
-        "name": "Ring all devices",
-        "priority": 100,
-        "direction": "inbound",
-        "match": {},
+        "id": "inbound-main-did",
+        "name": "Main DID",
+        "priority": 200,
+        "enabled": true,
+        "match": {
+          "direction": "inbound",
+          "sourceProvider": "my-trunk",
+          "numberPattern": "+49421219694"
+        },
         "action": {
           "targets": ["desk-phone"],
           "ringBrowsers": true,
-          "voicemailBox": "main",
-          "noAnswerTimeout": 25
+          "voicemailBox": "main"
+        }
+      },
+      {
+        "id": "inbound-support-did",
+        "name": "Support DID",
+        "priority": 190,
+        "enabled": true,
+        "match": {
+          "direction": "inbound",
+          "sourceProvider": "my-trunk",
+          "numberPattern": "+49421219695"
+        },
+        "action": {
+          "ivrMenuId": "support-menu"
         }
       },
       {
         "id": "outbound-default",
         "name": "Route via trunk",
         "priority": 100,
-        "direction": "outbound",
-        "match": {},
+        "enabled": true,
+        "match": { "direction": "outbound" },
         "action": { "provider": "my-trunk" }
       }
     ]
@@ -170,9 +204,11 @@ Create `.nogit/config.json`:
 }
 ```
 
+Inbound number ownership is explicit: add one inbound route per DID (or DID prefix) and scope it with `sourceProvider` when a provider delivers multiple external numbers.
+
 ### TTS Setup (Optional)
 
-For neural announcements and voicemail greetings, download the Kokoro TTS model:
+For neural voicemail greetings and IVR prompts, download the Kokoro TTS model:
 
 ```bash
 mkdir -p .nogit/tts
@@ -182,7 +218,7 @@ curl -L -o .nogit/tts/voices.bin \
   https://github.com/mzdk100/kokoro/releases/download/V1.0/voices.bin
 ```
 
-Without the model files, TTS falls back to `espeak-ng`. Without either, announcements are skipped — everything else works fine.
+Without the model files, TTS prompts (IVR menus, voicemail greetings) are skipped — everything else works fine.
 
 ### Run
 
@@ -209,7 +245,6 @@ siprouter/
 │   ├── frontend.ts                # Web dashboard HTTP/WS server + REST API
 │   ├── webrtcbridge.ts            # WebRTC signaling layer
 │   ├── registrar.ts               # Browser softphone registration
-│   ├── announcement.ts            # TTS announcement generator (espeak-ng / Kokoro)
 │   ├── voicebox.ts                # Voicemail box management
 │   └── call/
 │       └── prompt-cache.ts        # Named audio prompt WAV management
@@ -246,9 +281,17 @@ The `proxy-engine` binary handles all real-time audio processing with a **48kHz
 
 ### Audio Pipeline
 
-```
-Inbound:   Wire RTP → Jitter Buffer → Decode → Resample to 48kHz → Denoise (RNNoise) → Mix Bus
-Outbound:  Mix Bus → Mix-Minus → Resample to codec rate → Encode → Wire RTP
+```mermaid
+flowchart LR
+    subgraph Inbound["Inbound path (per leg)"]
+        direction LR
+        IN_RTP["Wire RTP"] --> IN_JB["Jitter Buffer"] --> IN_DEC["Decode"] --> IN_RS["Resample → 48 kHz"] --> IN_DN["Denoise (RNNoise)"] --> IN_BUS["Mix Bus"]
+    end
+
+    subgraph Outbound["Outbound path (per leg)"]
+        direction LR
+        OUT_BUS["Mix Bus"] --> OUT_MM["Mix-Minus"] --> OUT_RS["Resample → codec rate"] --> OUT_ENC["Encode"] --> OUT_RTP["Wire RTP"]
+    end
 ```
 
 - **Adaptive jitter buffer** — per-leg `BTreeMap`-based buffer keyed by RTP sequence number. Delivers exactly one frame per 20ms mixer tick in sequence order. Adaptive target depth starts at 3 frames (60ms) and adjusts between 2–6 frames based on observed network jitter. Handles hold/resume by detecting large forward sequence jumps and resetting cleanly.
@@ -262,13 +305,12 @@ Outbound:  Mix Bus → Mix-Minus → Resample to codec rate → Encode → Wire
 
 ## 🗣️ Neural TTS
 
-Announcements and voicemail greetings are synthesized using [Kokoro TTS](https://github.com/mzdk100/kokoro) — an 82M parameter neural model running via ONNX Runtime directly in the Rust process:
+Voicemail greetings and IVR prompts are synthesized using [Kokoro TTS](https://github.com/mzdk100/kokoro) — an 82M parameter neural model running via ONNX Runtime directly in the Rust process:
 
 - **24 kHz, 16-bit mono** output
 - **25+ voice presets** — American/British, male/female (e.g., `af_bella`, `am_adam`, `bf_emma`, `bm_george`)
 - **~800ms** synthesis time for a 3-second phrase
 - Lazy-loaded on first use — no startup cost if TTS is unused
-- Falls back to `espeak-ng` if the ONNX model is not available
 
 ---
 

rust/.cargo/config.toml

@@ -0,0 +1,30 @@
+# Cross-compile configuration for the proxy-engine crate.
+#
+# tsrust builds for both x86_64-unknown-linux-gnu and aarch64-unknown-linux-gnu
+# from an x86_64 host. Without this config, cargo invokes the host `cc` to
+# link aarch64 objects and fails with
+#   rust-lld: error: <obj.o> is incompatible with elf64-x86-64
+#
+# Required Debian/Ubuntu packages for the aarch64 target to work:
+#   sudo apt install gcc-aarch64-linux-gnu g++-aarch64-linux-gnu \
+#                    libc6-dev-arm64-cross libstdc++6-arm64-cross
+#
+# The `libstdc++.so` dev symlink (needed by the -lstdc++ flag that the
+# kokoro-tts/ort build scripts emit) is provided by this repo at
+# ./crosslibs/aarch64/libstdc++.so, pointing at the versioned shared
+# library installed by `libstdc++6-arm64-cross`. This avoids requiring
+# the `libstdc++-13-dev-arm64-cross` package, which is not always
+# installed alongside the runtime.
+
+[target.aarch64-unknown-linux-gnu]
+linker = "aarch64-linux-gnu-gcc"
+rustflags = ["-C", "link-arg=-L.cargo/crosslibs/aarch64"]
+
+# Tell cc-rs-based build scripts (ring, zstd-sys, audiopus_sys, ort-sys) to
+# use the aarch64 cross toolchain when compiling C sources for the aarch64
+# target. Without these, they'd default to the host `cc` and produce x86_64
+# objects that the aarch64 linker then rejects.
+[env]
+CC_aarch64_unknown_linux_gnu = "aarch64-linux-gnu-gcc"
+CXX_aarch64_unknown_linux_gnu = "aarch64-linux-gnu-g++"
+AR_aarch64_unknown_linux_gnu = "aarch64-linux-gnu-ar"

rust/.cargo/crosslibs/aarch64/libstdc++.so

@@ -0,0 +1 @@
+/usr/aarch64-linux-gnu/lib/libstdc++.so.6

rust/Cargo.lock

@@ -532,6 +532,15 @@ dependencies = [
  "cc",
 ]
 
+[[package]]
+name = "cmudict-fast"
+version = "0.8.0"
+source = "registry+https://github.com/rust-lang/crates.io-index"
+checksum = "2c9f73004e928ed46c3e7fd7406d2b12c8674153295f08af084b49860276dc02"
+dependencies = [
+ "thiserror",
+]
+
 [[package]]
 name = "codec-lib"
 version = "0.1.0"
@@ -1730,6 +1739,7 @@ dependencies = [
  "bincode 2.0.1",
  "cc",
  "chinese-number",
+ "cmudict-fast",
  "futures",
  "jieba-rs",
  "log",

rust/crates/codec-lib/src/lib.rs

@@ -115,9 +115,8 @@ pub struct TranscodeState {
 impl TranscodeState {
     /// Create a new transcoding session with fresh codec state.
     pub fn new() -> Result<Self, String> {
-        let mut opus_enc =
-            OpusEncoder::new(SampleRate::Hz48000, Channels::Mono, Application::Voip)
-                .map_err(|e| format!("opus encoder: {e}"))?;
+        let mut opus_enc = OpusEncoder::new(SampleRate::Hz48000, Channels::Mono, Application::Voip)
+            .map_err(|e| format!("opus encoder: {e}"))?;
         opus_enc
             .set_complexity(5)
             .map_err(|e| format!("opus set_complexity: {e}"))?;
@@ -160,14 +159,9 @@ impl TranscodeState {
         let key = (from_rate, to_rate, canonical_chunk);
 
         if !self.resamplers.contains_key(&key) {
-            let r = FftFixedIn::<f64>::new(
-                from_rate as usize,
-                to_rate as usize,
-                canonical_chunk,
-                1,
-                1,
-            )
-            .map_err(|e| format!("resampler {from_rate}->{to_rate}: {e}"))?;
+            let r =
+                FftFixedIn::<f64>::new(from_rate as usize, to_rate as usize, canonical_chunk, 1, 1)
+                    .map_err(|e| format!("resampler {from_rate}->{to_rate}: {e}"))?;
             self.resamplers.insert(key, r);
         }
         let resampler = self.resamplers.get_mut(&key).unwrap();
@@ -284,8 +278,7 @@ impl TranscodeState {
         match pt {
             PT_OPUS => {
                 let mut pcm = vec![0i16; 5760]; // up to 120ms at 48kHz
-                let packet =
-                    OpusPacket::try_from(data).map_err(|e| format!("opus packet: {e}"))?;
+                let packet = OpusPacket::try_from(data).map_err(|e| format!("opus packet: {e}"))?;
                 let out =
                     MutSignals::try_from(&mut pcm[..]).map_err(|e| format!("opus signals: {e}"))?;
                 let n: usize = self
@@ -343,8 +336,7 @@ impl TranscodeState {
         match pt {
             PT_OPUS => {
                 let mut pcm = vec![0.0f32; 5760]; // up to 120ms at 48kHz
-                let packet =
-                    OpusPacket::try_from(data).map_err(|e| format!("opus packet: {e}"))?;
+                let packet = OpusPacket::try_from(data).map_err(|e| format!("opus packet: {e}"))?;
                 let out =
                     MutSignals::try_from(&mut pcm[..]).map_err(|e| format!("opus signals: {e}"))?;
                 let n: usize = self
@@ -368,8 +360,8 @@ impl TranscodeState {
     /// Returns f32 PCM at 48kHz. `frame_size` should be 960 for 20ms.
     pub fn opus_plc(&mut self, frame_size: usize) -> Result<Vec<f32>, String> {
         let mut pcm = vec![0.0f32; frame_size];
-        let out = MutSignals::try_from(&mut pcm[..])
-            .map_err(|e| format!("opus plc signals: {e}"))?;
+        let out =
+            MutSignals::try_from(&mut pcm[..]).map_err(|e| format!("opus plc signals: {e}"))?;
         let n: usize = self
             .opus_dec
             .decode_float(None::<OpusPacket<'_>>, out, false)
@@ -425,14 +417,9 @@ impl TranscodeState {
         let key = (from_rate, to_rate, canonical_chunk);
 
         if !self.resamplers_f32.contains_key(&key) {
-            let r = FftFixedIn::<f32>::new(
-                from_rate as usize,
-                to_rate as usize,
-                canonical_chunk,
-                1,
-                1,
-            )
-            .map_err(|e| format!("resampler f32 {from_rate}->{to_rate}: {e}"))?;
+            let r =
+                FftFixedIn::<f32>::new(from_rate as usize, to_rate as usize, canonical_chunk, 1, 1)
+                    .map_err(|e| format!("resampler f32 {from_rate}->{to_rate}: {e}"))?;
             self.resamplers_f32.insert(key, r);
         }
         let resampler = self.resamplers_f32.get_mut(&key).unwrap();
@@ -508,8 +495,10 @@ mod tests {
             let encoded = mulaw_encode(sample);
             let decoded = mulaw_decode(encoded);
             // µ-law is lossy; verify the decoded value is close.
-            assert!((sample as i32 - decoded as i32).abs() < 1000,
-                "µ-law roundtrip failed for {sample}: got {decoded}");
+            assert!(
+                (sample as i32 - decoded as i32).abs() < 1000,
+                "µ-law roundtrip failed for {sample}: got {decoded}"
+            );
         }
     }
 
@@ -518,8 +507,10 @@ mod tests {
         for sample in [-32768i16, -1000, -1, 0, 1, 1000, 32767] {
             let encoded = alaw_encode(sample);
             let decoded = alaw_decode(encoded);
-            assert!((sample as i32 - decoded as i32).abs() < 1000,
-                "A-law roundtrip failed for {sample}: got {decoded}");
+            assert!(
+                (sample as i32 - decoded as i32).abs() < 1000,
+                "A-law roundtrip failed for {sample}: got {decoded}"
+            );
         }
     }
 
@@ -543,7 +534,9 @@ mod tests {
     fn pcmu_to_pcma_roundtrip() {
         let mut st = TranscodeState::new().unwrap();
         // 160 bytes = 20ms of PCMU at 8kHz
-        let pcmu_data: Vec<u8> = (0..160).map(|i| mulaw_encode((i as i16 * 200) - 16000)).collect();
+        let pcmu_data: Vec<u8> = (0..160)
+            .map(|i| mulaw_encode((i as i16 * 200) - 16000))
+            .collect();
         let pcma = st.transcode(&pcmu_data, PT_PCMU, PT_PCMA, None).unwrap();
         assert_eq!(pcma.len(), 160); // Same frame size
         let back = st.transcode(&pcma, PT_PCMA, PT_PCMU, None).unwrap();

rust/crates/proxy-engine/Cargo.toml

@@ -19,7 +19,7 @@ regex-lite = "0.1"
 webrtc = "0.8"
 rand = "0.8"
 hound = "3.5"
-kokoro-tts = { version = "0.3", default-features = false }
+kokoro-tts = { version = "0.3", default-features = false, features = ["use-cmudict"] }
 ort = { version = "=2.0.0-rc.11", default-features = false, features = [
     "std", "download-binaries", "copy-dylibs", "ndarray",
     "tls-native-vendored"

rust/crates/proxy-engine/src/audio_player.rs

@@ -36,10 +36,7 @@ pub async fn play_wav_file(
 
     // Read all samples as i16.
     let samples: Vec<i16> = if spec.bits_per_sample == 16 {
-        reader
-            .samples::<i16>()
-            .filter_map(|s| s.ok())
-            .collect()
+        reader.samples::<i16>().filter_map(|s| s.ok()).collect()
     } else if spec.bits_per_sample == 32 && spec.sample_format == hound::SampleFormat::Float {
         reader
             .samples::<f32>()
@@ -199,10 +196,7 @@ pub fn load_prompt_pcm_frames(wav_path: &str) -> Result<Vec<Vec<f32>>, String> {
             .map(|s| s as f32 / 32768.0)
             .collect()
     } else if spec.bits_per_sample == 32 && spec.sample_format == hound::SampleFormat::Float {
-        reader
-            .samples::<f32>()
-            .filter_map(|s| s.ok())
-            .collect()
+        reader.samples::<f32>().filter_map(|s| s.ok()).collect()
     } else {
         return Err(format!(
             "unsupported WAV format: {}bit {:?}",
@@ -214,14 +208,23 @@ pub fn load_prompt_pcm_frames(wav_path: &str) -> Result<Vec<Vec<f32>>, String> {
         return Ok(vec![]);
     }
 
+    pcm_to_mix_frames(&samples, wav_rate)
+}
+
+/// Convert PCM samples at an arbitrary rate into 48kHz 20ms mixer frames.
+pub fn pcm_to_mix_frames(samples: &[f32], sample_rate: u32) -> Result<Vec<Vec<f32>>, String> {
+    if samples.is_empty() {
+        return Ok(vec![]);
+    }
+
     // Resample to MIX_RATE (48kHz) if needed.
-    let resampled = if wav_rate != MIX_RATE {
+    let resampled = if sample_rate != MIX_RATE {
         let mut transcoder = TranscodeState::new().map_err(|e| format!("codec init: {e}"))?;
         transcoder
-            .resample_f32(&samples, wav_rate, MIX_RATE)
+            .resample_f32(samples, sample_rate, MIX_RATE)
             .map_err(|e| format!("resample: {e}"))?
     } else {
-        samples
+        samples.to_vec()
     };
 
     // Split into MIX_FRAME_SIZE (960) sample frames.

rust/crates/proxy-engine/src/call.rs

@@ -23,16 +23,22 @@ pub enum CallState {
     Ringing,
     Connected,
     Voicemail,
+    Ivr,
     Terminated,
 }
 
 impl CallState {
+    /// Wire-format string for events/dashboards. Not currently emitted —
+    /// call state changes flow as typed events (`call_answered`, etc.) —
+    /// but kept for future status-snapshot work.
+    #[allow(dead_code)]
     pub fn as_str(&self) -> &'static str {
         match self {
             Self::SettingUp => "setting-up",
             Self::Ringing => "ringing",
             Self::Connected => "connected",
             Self::Voicemail => "voicemail",
+            Self::Ivr => "ivr",
             Self::Terminated => "terminated",
         }
     }
@@ -45,6 +51,8 @@ pub enum CallDirection {
 }
 
 impl CallDirection {
+    /// Wire-format string. See CallState::as_str.
+    #[allow(dead_code)]
     pub fn as_str(&self) -> &'static str {
         match self {
             Self::Inbound => "inbound",
@@ -59,8 +67,13 @@ pub enum LegKind {
     SipProvider,
     SipDevice,
     WebRtc,
-    Media, // voicemail playback, IVR, recording
-    Tool,  // observer leg for recording, transcription, etc.
+    /// Voicemail playback, IVR prompt playback, recording — not yet wired up
+    /// as a distinct leg kind (those paths currently use the mixer's role
+    /// system instead). Kept behind allow so adding a real media leg later
+    /// doesn't require re-introducing the variant.
+    #[allow(dead_code)]
+    Media,
+    Tool, // observer leg for recording, transcription, etc.
 }
 
 impl LegKind {
@@ -107,11 +120,22 @@ pub struct LegInfo {
     /// For SIP legs: the SIP Call-ID for message routing.
     pub sip_call_id: Option<String>,
     /// For WebRTC legs: the session ID in WebRtcEngine.
+    ///
+    /// Populated at leg creation but not yet consumed by the hub —
+    /// WebRTC session lookup currently goes through the session registry
+    /// directly. Kept for introspection/debugging.
+    #[allow(dead_code)]
     pub webrtc_session_id: Option<String>,
     /// The RTP socket allocated for this leg.
     pub rtp_socket: Option<Arc<UdpSocket>>,
     /// The RTP port number.
     pub rtp_port: u16,
+    /// Public IP to advertise in SDP/Record-Route when THIS leg is the
+    /// destination of a rewrite. Populated only for provider legs; `None`
+    /// for LAN SIP devices, WebRTC browsers, media, and tool legs (which
+    /// are reachable via `lan_ip`). See `route_passthrough_message` for
+    /// the per-destination advertise-IP logic.
+    pub public_ip: Option<String>,
     /// The remote media endpoint (learned from SDP or address learning).
     pub remote_media: Option<SocketAddr>,
     /// SIP signaling address (provider or device).
@@ -124,14 +148,21 @@ pub struct LegInfo {
 
 /// A multiparty call with N legs and a central mixer.
 pub struct Call {
+    // Duplicated from the HashMap key in CallManager. Kept for future
+    // status-snapshot work.
+    #[allow(dead_code)]
     pub id: String,
     pub state: CallState,
+    // Populated at call creation but not currently consumed — dashboard
+    // pull snapshots are gone (push events only).
+    #[allow(dead_code)]
     pub direction: CallDirection,
     pub created_at: Instant,
 
     // Metadata.
     pub caller_number: Option<String>,
     pub callee_number: Option<String>,
+    #[allow(dead_code)]
     pub provider_id: String,
 
     /// Original INVITE from the device (for device-originated outbound calls).
@@ -211,42 +242,4 @@ impl Call {
             handle.abort();
         }
     }
-
-    /// Produce a JSON status snapshot for the dashboard.
-    pub fn to_status_json(&self) -> serde_json::Value {
-        let legs: Vec<serde_json::Value> = self
-            .legs
-            .values()
-            .filter(|l| l.state != LegState::Terminated)
-            .map(|l| {
-                let metadata: serde_json::Value = if l.metadata.is_empty() {
-                    serde_json::json!({})
-                } else {
-                    serde_json::Value::Object(
-                        l.metadata.iter().map(|(k, v)| (k.clone(), v.clone())).collect(),
-                    )
-                };
-                serde_json::json!({
-                    "id": l.id,
-                    "type": l.kind.as_str(),
-                    "state": l.state.as_str(),
-                    "codec": sip_proto::helpers::codec_name(l.codec_pt),
-                    "rtpPort": l.rtp_port,
-                    "remoteMedia": l.remote_media.map(|a| format!("{}:{}", a.ip(), a.port())),
-                    "metadata": metadata,
-                })
-            })
-            .collect();
-
-        serde_json::json!({
-            "id": self.id,
-            "state": self.state.as_str(),
-            "direction": self.direction.as_str(),
-            "callerNumber": self.caller_number,
-            "calleeNumber": self.callee_number,
-            "providerUsed": self.provider_id,
-            "duration": self.duration_secs(),
-            "legs": legs,
-        })
-    }
 }

rust/crates/proxy-engine/src/config.rs

@@ -4,6 +4,7 @@
 //! proxy engine via the `configure` command. These types mirror the TS interfaces.
 
 use serde::Deserialize;
+use sip_proto::message::SipMessage;
 use std::net::SocketAddr;
 
 /// Network endpoint.
@@ -30,6 +31,11 @@ impl Endpoint {
 }
 
 /// Provider quirks for codec/protocol workarounds.
+//
+// Deserialized from provider config for TS parity. Early-media silence
+// injection and related workarounds are not yet ported to the Rust engine,
+// so every field is populated by serde but not yet consumed.
+#[allow(dead_code)]
 #[derive(Debug, Clone, Deserialize)]
 pub struct Quirks {
     #[serde(rename = "earlyMediaSilence")]
@@ -44,6 +50,9 @@ pub struct Quirks {
 #[derive(Debug, Clone, Deserialize)]
 pub struct ProviderConfig {
     pub id: String,
+    // UI label — populated by serde for parity with the TS config, not
+    // consumed at runtime.
+    #[allow(dead_code)]
     #[serde(rename = "displayName")]
     pub display_name: String,
     pub domain: String,
@@ -54,6 +63,8 @@ pub struct ProviderConfig {
     #[serde(rename = "registerIntervalSec")]
     pub register_interval_sec: u32,
     pub codecs: Vec<u8>,
+    // Workaround knobs populated by serde but not yet acted upon — see Quirks.
+    #[allow(dead_code)]
     pub quirks: Quirks,
 }
 
@@ -84,6 +95,10 @@ pub struct RouteMatch {
 
 /// Route action.
 #[derive(Debug, Clone, Deserialize)]
+// Several fields (voicemail_box, ivr_menu_id, no_answer_timeout) are read
+// by resolve_inbound_route but not yet honored downstream — see the
+// multi-target TODO in CallManager::create_inbound_call.
+#[allow(dead_code)]
 pub struct RouteAction {
     pub targets: Option<Vec<String>>,
     #[serde(rename = "ringBrowsers")]
@@ -106,7 +121,11 @@ pub struct RouteAction {
 /// A routing rule.
 #[derive(Debug, Clone, Deserialize)]
 pub struct Route {
+    // `id` and `name` are UI identifiers, populated by serde but not
+    // consumed by the resolvers.
+    #[allow(dead_code)]
     pub id: String,
+    #[allow(dead_code)]
     pub name: String,
     pub priority: i32,
     pub enabled: bool,
@@ -141,6 +160,10 @@ pub struct AppConfig {
     pub providers: Vec<ProviderConfig>,
     pub devices: Vec<DeviceConfig>,
     pub routing: RoutingConfig,
+    #[serde(default)]
+    pub voiceboxes: Vec<VoiceboxConfig>,
+    #[serde(default)]
+    pub ivr: Option<IvrConfig>,
 }
 
 #[derive(Debug, Clone, Deserialize)]
@@ -148,12 +171,158 @@ pub struct RoutingConfig {
     pub routes: Vec<Route>,
 }
 
+// ---------------------------------------------------------------------------
+// Voicebox config
+// ---------------------------------------------------------------------------
+
+#[allow(dead_code)]
+#[derive(Debug, Clone, Deserialize)]
+pub struct VoiceboxConfig {
+    pub id: String,
+    #[serde(default)]
+    pub enabled: bool,
+    #[serde(rename = "greetingText")]
+    pub greeting_text: Option<String>,
+    #[serde(rename = "greetingVoice")]
+    pub greeting_voice: Option<String>,
+    #[serde(rename = "greetingWavPath")]
+    pub greeting_wav_path: Option<String>,
+    #[serde(rename = "maxRecordingSec")]
+    pub max_recording_sec: Option<u32>,
+}
+
+// ---------------------------------------------------------------------------
+// IVR config
+// ---------------------------------------------------------------------------
+
+#[allow(dead_code)]
+#[derive(Debug, Clone, Deserialize)]
+pub struct IvrConfig {
+    pub enabled: bool,
+    pub menus: Vec<IvrMenuConfig>,
+    #[serde(rename = "entryMenuId")]
+    pub entry_menu_id: String,
+}
+
+#[derive(Debug, Clone, Deserialize)]
+pub struct IvrMenuConfig {
+    pub id: String,
+    #[serde(rename = "promptText")]
+    pub prompt_text: String,
+    #[serde(rename = "promptVoice")]
+    pub prompt_voice: Option<String>,
+    pub entries: Vec<IvrMenuEntry>,
+    #[serde(rename = "timeoutSec")]
+    pub timeout_sec: Option<u32>,
+}
+
+#[allow(dead_code)]
+#[derive(Debug, Clone, Deserialize)]
+pub struct IvrMenuEntry {
+    pub digit: String,
+    pub action: String,
+    pub target: Option<String>,
+}
+
 // ---------------------------------------------------------------------------
 // Pattern matching (ported from ts/config.ts)
 // ---------------------------------------------------------------------------
 
+/// Extract the URI user part and normalize phone-like identities for routing.
+///
+/// This keeps inbound route matching stable across provider-specific URI shapes,
+/// e.g. `sip:+49 421 219694@trunk.example` and `sip:0049421219694@trunk.example`
+/// both normalize to `+49421219694`.
+pub fn normalize_routing_identity(value: &str) -> String {
+    let extracted = SipMessage::extract_uri_user(value).unwrap_or(value).trim();
+    if extracted.is_empty() {
+        return String::new();
+    }
+
+    let mut digits = String::new();
+    let mut saw_plus = false;
+
+    for (idx, ch) in extracted.chars().enumerate() {
+        if ch.is_ascii_digit() {
+            digits.push(ch);
+            continue;
+        }
+
+        if ch == '+' && idx == 0 {
+            saw_plus = true;
+            continue;
+        }
+
+        if matches!(ch, ' ' | '\t' | '-' | '.' | '/' | '(' | ')') {
+            continue;
+        }
+
+        return extracted.to_string();
+    }
+
+    if digits.is_empty() {
+        return extracted.to_string();
+    }
+    if saw_plus {
+        return format!("+{digits}");
+    }
+    if digits.starts_with("00") && digits.len() > 2 {
+        return format!("+{}", &digits[2..]);
+    }
+
+    digits
+}
+
+fn parse_numeric_range_value(value: &str) -> Option<(bool, &str)> {
+    let trimmed = value.trim();
+    if trimmed.is_empty() {
+        return None;
+    }
+
+    let (has_plus, digits) = if let Some(rest) = trimmed.strip_prefix('+') {
+        (true, rest)
+    } else {
+        (false, trimmed)
+    };
+
+    if digits.is_empty() || !digits.chars().all(|c| c.is_ascii_digit()) {
+        return None;
+    }
+
+    Some((has_plus, digits))
+}
+
+fn matches_numeric_range_pattern(pattern: &str, value: &str) -> bool {
+    let Some((start, end)) = pattern.split_once("..") else {
+        return false;
+    };
+
+    let Some((start_plus, start_digits)) = parse_numeric_range_value(start) else {
+        return false;
+    };
+    let Some((end_plus, end_digits)) = parse_numeric_range_value(end) else {
+        return false;
+    };
+    let Some((value_plus, value_digits)) = parse_numeric_range_value(value) else {
+        return false;
+    };
+
+    if start_plus != end_plus || value_plus != start_plus {
+        return false;
+    }
+    if start_digits.len() != end_digits.len() || value_digits.len() != start_digits.len() {
+        return false;
+    }
+    if start_digits > end_digits {
+        return false;
+    }
+
+    value_digits >= start_digits && value_digits <= end_digits
+}
+
 /// Test a value against a pattern string.
 /// - None/empty: matches everything (wildcard)
+/// - `start..end`: numeric range match
 /// - Trailing '*': prefix match
 /// - Starts with '/': regex match
 /// - Otherwise: exact match
@@ -169,6 +338,10 @@ pub fn matches_pattern(pattern: Option<&str>, value: &str) -> bool {
         return value.starts_with(&pattern[..pattern.len() - 1]);
     }
 
+    if matches_numeric_range_pattern(pattern, value) {
+        return true;
+    }
+
     // Regex match: "/^\\+49/" or "/pattern/i"
     if pattern.starts_with('/') {
         if let Some(last_slash) = pattern[1..].rfind('/') {
@@ -192,10 +365,18 @@ pub fn matches_pattern(pattern: Option<&str>, value: &str) -> bool {
 /// Result of resolving an outbound route.
 pub struct OutboundRouteResult {
     pub provider: ProviderConfig,
+    // TODO: prefix rewriting is unfinished — this is computed but the
+    // caller ignores it and uses the raw dialed number.
+    #[allow(dead_code)]
     pub transformed_number: String,
 }
 
 /// Result of resolving an inbound route.
+//
+// `device_ids` and `ring_browsers` are consumed by create_inbound_call.
+// The remaining fields (voicemail_box, ivr_menu_id, no_answer_timeout)
+// are resolved but not yet acted upon — see the multi-target TODO.
+#[allow(dead_code)]
 pub struct InboundRouteResult {
     pub device_ids: Vec<String>,
     pub ring_browsers: bool,
@@ -280,7 +461,7 @@ impl AppConfig {
         provider_id: &str,
         called_number: &str,
         caller_number: &str,
-    ) -> InboundRouteResult {
+    ) -> Option<InboundRouteResult> {
         let mut routes: Vec<&Route> = self
             .routing
             .routes
@@ -304,22 +485,170 @@ impl AppConfig {
                 continue;
             }
 
-            return InboundRouteResult {
+            return Some(InboundRouteResult {
                 device_ids: route.action.targets.clone().unwrap_or_default(),
                 ring_browsers: route.action.ring_browsers.unwrap_or(false),
                 voicemail_box: route.action.voicemail_box.clone(),
                 ivr_menu_id: route.action.ivr_menu_id.clone(),
                 no_answer_timeout: route.action.no_answer_timeout,
-            };
+            });
         }
 
-        // Fallback: ring all devices + browsers.
-        InboundRouteResult {
-            device_ids: vec![],
-            ring_browsers: true,
-            voicemail_box: None,
-            ivr_menu_id: None,
-            no_answer_timeout: None,
-        }
+        None
+    }
+}
+
+#[cfg(test)]
+mod tests {
+    use super::*;
+
+    fn test_app_config(routes: Vec<Route>) -> AppConfig {
+        AppConfig {
+            proxy: ProxyConfig {
+                lan_ip: "127.0.0.1".to_string(),
+                lan_port: 5070,
+                public_ip_seed: None,
+                rtp_port_range: RtpPortRange {
+                    min: 20_000,
+                    max: 20_100,
+                },
+            },
+            providers: vec![ProviderConfig {
+                id: "provider-a".to_string(),
+                display_name: "Provider A".to_string(),
+                domain: "example.com".to_string(),
+                outbound_proxy: Endpoint {
+                    address: "example.com".to_string(),
+                    port: 5060,
+                },
+                username: "user".to_string(),
+                password: "pass".to_string(),
+                register_interval_sec: 300,
+                codecs: vec![9],
+                quirks: Quirks {
+                    early_media_silence: false,
+                    silence_payload_type: None,
+                    silence_max_packets: None,
+                },
+            }],
+            devices: vec![DeviceConfig {
+                id: "desk".to_string(),
+                display_name: "Desk".to_string(),
+                expected_address: "127.0.0.1".to_string(),
+                extension: "100".to_string(),
+            }],
+            routing: RoutingConfig { routes },
+            voiceboxes: vec![],
+            ivr: None,
+        }
+    }
+
+    #[test]
+    fn normalize_routing_identity_extracts_uri_user_and_phone_number() {
+        assert_eq!(
+            normalize_routing_identity("sip:0049 421 219694@voip.easybell.de"),
+            "+49421219694"
+        );
+        assert_eq!(
+            normalize_routing_identity("<tel:+49 (421) 219694>"),
+            "+49421219694"
+        );
+        assert_eq!(normalize_routing_identity("sip:100@pbx.local"), "100");
+        assert_eq!(normalize_routing_identity("sip:alice@pbx.local"), "alice");
+    }
+
+    #[test]
+    fn resolve_inbound_route_requires_explicit_match() {
+        let cfg = test_app_config(vec![]);
+        assert!(cfg
+            .resolve_inbound_route("provider-a", "+49421219694", "+491701234567")
+            .is_none());
+    }
+
+    #[test]
+    fn resolve_inbound_route_matches_per_number_on_shared_provider() {
+        let cfg = test_app_config(vec![
+            Route {
+                id: "main".to_string(),
+                name: "Main DID".to_string(),
+                priority: 200,
+                enabled: true,
+                match_criteria: RouteMatch {
+                    direction: "inbound".to_string(),
+                    number_pattern: Some("+49421219694".to_string()),
+                    caller_pattern: None,
+                    source_provider: Some("provider-a".to_string()),
+                    source_device: None,
+                },
+                action: RouteAction {
+                    targets: Some(vec!["desk".to_string()]),
+                    ring_browsers: Some(true),
+                    voicemail_box: None,
+                    ivr_menu_id: None,
+                    no_answer_timeout: None,
+                    provider: None,
+                    failover_providers: None,
+                    strip_prefix: None,
+                    prepend_prefix: None,
+                },
+            },
+            Route {
+                id: "support".to_string(),
+                name: "Support DID".to_string(),
+                priority: 100,
+                enabled: true,
+                match_criteria: RouteMatch {
+                    direction: "inbound".to_string(),
+                    number_pattern: Some("+49421219695".to_string()),
+                    caller_pattern: None,
+                    source_provider: Some("provider-a".to_string()),
+                    source_device: None,
+                },
+                action: RouteAction {
+                    targets: None,
+                    ring_browsers: Some(false),
+                    voicemail_box: Some("support-box".to_string()),
+                    ivr_menu_id: None,
+                    no_answer_timeout: Some(20),
+                    provider: None,
+                    failover_providers: None,
+                    strip_prefix: None,
+                    prepend_prefix: None,
+                },
+            },
+        ]);
+
+        let main = cfg
+            .resolve_inbound_route("provider-a", "+49421219694", "+491701234567")
+            .expect("main DID should match");
+        assert_eq!(main.device_ids, vec!["desk".to_string()]);
+        assert!(main.ring_browsers);
+
+        let support = cfg
+            .resolve_inbound_route("provider-a", "+49421219695", "+491701234567")
+            .expect("support DID should match");
+        assert_eq!(support.voicemail_box.as_deref(), Some("support-box"));
+        assert_eq!(support.no_answer_timeout, Some(20));
+        assert!(!support.ring_browsers);
+    }
+
+    #[test]
+    fn matches_pattern_supports_numeric_ranges() {
+        assert!(matches_pattern(
+            Some("042116767546..042116767548"),
+            "042116767547"
+        ));
+        assert!(!matches_pattern(
+            Some("042116767546..042116767548"),
+            "042116767549"
+        ));
+        assert!(matches_pattern(
+            Some("+4942116767546..+4942116767548"),
+            "+4942116767547"
+        ));
+        assert!(!matches_pattern(
+            Some("+4942116767546..+4942116767548"),
+            "042116767547"
+        ));
     }
 }

rust/crates/proxy-engine/src/dtmf.rs

@@ -1,200 +0,0 @@
-//! DTMF detection — parses RFC 2833 telephone-event RTP packets.
-//!
-//! Deduplicates repeated packets (same digit sent multiple times with
-//! increasing duration) and fires once per detected digit.
-//!
-//! Ported from ts/call/dtmf-detector.ts.
-
-use crate::ipc::{emit_event, OutTx};
-
-/// RFC 2833 event ID → character mapping.
-const EVENT_CHARS: &[char] = &[
-    '0', '1', '2', '3', '4', '5', '6', '7', '8', '9', '*', '#', 'A', 'B', 'C', 'D',
-];
-
-/// Safety timeout: report digit if no End packet arrives within this many ms.
-const SAFETY_TIMEOUT_MS: u64 = 200;
-
-/// DTMF detector for a single RTP stream.
-pub struct DtmfDetector {
-    /// Negotiated telephone-event payload type (default 101).
-    telephone_event_pt: u8,
-    /// Clock rate for duration calculation (default 8000 Hz).
-    clock_rate: u32,
-    /// Call ID for event emission.
-    call_id: String,
-
-    // Deduplication state.
-    current_event_id: Option<u8>,
-    current_event_ts: Option<u32>,
-    current_event_reported: bool,
-    current_event_duration: u16,
-
-    out_tx: OutTx,
-}
-
-impl DtmfDetector {
-    pub fn new(call_id: String, out_tx: OutTx) -> Self {
-        Self {
-            telephone_event_pt: 101,
-            clock_rate: 8000,
-            call_id,
-            current_event_id: None,
-            current_event_ts: None,
-            current_event_reported: false,
-            current_event_duration: 0,
-            out_tx,
-        }
-    }
-
-    /// Feed an RTP packet. Checks PT; ignores non-DTMF packets.
-    /// Returns Some(digit_char) if a digit was detected.
-    pub fn process_rtp(&mut self, data: &[u8]) -> Option<char> {
-        if data.len() < 16 {
-            return None; // 12-byte header + 4-byte telephone-event minimum
-        }
-
-        let pt = data[1] & 0x7F;
-        if pt != self.telephone_event_pt {
-            return None;
-        }
-
-        let marker = (data[1] & 0x80) != 0;
-        let rtp_timestamp = u32::from_be_bytes([data[4], data[5], data[6], data[7]]);
-
-        // Parse telephone-event payload.
-        let event_id = data[12];
-        let end_bit = (data[13] & 0x80) != 0;
-        let duration = u16::from_be_bytes([data[14], data[15]]);
-
-        if event_id as usize >= EVENT_CHARS.len() {
-            return None;
-        }
-
-        // Detect new event.
-        let is_new = marker
-            || self.current_event_id != Some(event_id)
-            || self.current_event_ts != Some(rtp_timestamp);
-
-        if is_new {
-            // Report pending unreported event.
-            let pending = self.report_pending();
-
-            self.current_event_id = Some(event_id);
-            self.current_event_ts = Some(rtp_timestamp);
-            self.current_event_reported = false;
-            self.current_event_duration = duration;
-
-            if pending.is_some() {
-                return pending;
-            }
-        }
-
-        if duration > self.current_event_duration {
-            self.current_event_duration = duration;
-        }
-
-        // Report on End bit (first time only).
-        if end_bit && !self.current_event_reported {
-            self.current_event_reported = true;
-            let digit = EVENT_CHARS[event_id as usize];
-            let duration_ms = (self.current_event_duration as f64 / self.clock_rate as f64) * 1000.0;
-
-            emit_event(
-                &self.out_tx,
-                "dtmf_digit",
-                serde_json::json!({
-                    "call_id": self.call_id,
-                    "digit": digit.to_string(),
-                    "duration_ms": duration_ms.round() as u32,
-                    "source": "rfc2833",
-                }),
-            );
-
-            return Some(digit);
-        }
-
-        None
-    }
-
-    /// Report a pending unreported event.
-    fn report_pending(&mut self) -> Option<char> {
-        if let Some(event_id) = self.current_event_id {
-            if !self.current_event_reported && (event_id as usize) < EVENT_CHARS.len() {
-                self.current_event_reported = true;
-                let digit = EVENT_CHARS[event_id as usize];
-                let duration_ms =
-                    (self.current_event_duration as f64 / self.clock_rate as f64) * 1000.0;
-
-                emit_event(
-                    &self.out_tx,
-                    "dtmf_digit",
-                    serde_json::json!({
-                        "call_id": self.call_id,
-                        "digit": digit.to_string(),
-                        "duration_ms": duration_ms.round() as u32,
-                        "source": "rfc2833",
-                    }),
-                );
-
-                return Some(digit);
-            }
-        }
-        None
-    }
-
-    /// Process a SIP INFO message body for DTMF.
-    pub fn process_sip_info(&mut self, content_type: &str, body: &str) -> Option<char> {
-        let ct = content_type.to_ascii_lowercase();
-
-        if ct.contains("application/dtmf-relay") {
-            // Format: "Signal= 5\r\nDuration= 160\r\n"
-            let signal = body
-                .lines()
-                .find(|l| l.to_ascii_lowercase().starts_with("signal"))
-                .and_then(|l| l.split('=').nth(1))
-                .map(|s| s.trim().to_string())?;
-
-            if signal.len() != 1 {
-                return None;
-            }
-            let digit = signal.chars().next()?.to_ascii_uppercase();
-            if !"0123456789*#ABCD".contains(digit) {
-                return None;
-            }
-
-            emit_event(
-                &self.out_tx,
-                "dtmf_digit",
-                serde_json::json!({
-                    "call_id": self.call_id,
-                    "digit": digit.to_string(),
-                    "source": "sip-info",
-                }),
-            );
-
-            return Some(digit);
-        }
-
-        if ct.contains("application/dtmf") {
-            let digit = body.trim().chars().next()?.to_ascii_uppercase();
-            if !"0123456789*#ABCD".contains(digit) {
-                return None;
-            }
-
-            emit_event(
-                &self.out_tx,
-                "dtmf_digit",
-                serde_json::json!({
-                    "call_id": self.call_id,
-                    "digit": digit.to_string(),
-                    "source": "sip-info",
-                }),
-            );
-
-            return Some(digit);
-        }
-
-        None
-    }
-}

rust/crates/proxy-engine/src/ipc.rs

@@ -19,7 +19,13 @@ pub struct Command {
 }
 
 /// Send a response to a command.
-pub fn respond(tx: &OutTx, id: &str, success: bool, result: Option<serde_json::Value>, error: Option<&str>) {
+pub fn respond(
+    tx: &OutTx,
+    id: &str,
+    success: bool,
+    result: Option<serde_json::Value>,
+    error: Option<&str>,
+) {
     let mut resp = serde_json::json!({ "id": id, "success": success });
     if let Some(r) = result {
         resp["result"] = r;

rust/crates/proxy-engine/src/leg_io.rs

@@ -63,7 +63,8 @@ pub fn spawn_sip_inbound(
                         if offset + 4 > n {
                             continue; // Malformed: extension header truncated.
                         }
-                        let ext_len = u16::from_be_bytes([buf[offset + 2], buf[offset + 3]]) as usize;
+                        let ext_len =
+                            u16::from_be_bytes([buf[offset + 2], buf[offset + 3]]) as usize;
                         offset += 4 + ext_len * 4;
                     }
                     if offset >= n {
@@ -74,7 +75,17 @@ pub fn spawn_sip_inbound(
                     if payload.is_empty() {
                         continue;
                     }
-                    if inbound_tx.send(RtpPacket { payload, payload_type: pt, marker, seq, timestamp }).await.is_err() {
+                    if inbound_tx
+                        .send(RtpPacket {
+                            payload,
+                            payload_type: pt,
+                            marker,
+                            seq,
+                            timestamp,
+                        })
+                        .await
+                        .is_err()
+                    {
                         break; // Channel closed — leg removed.
                     }
                 }

rust/crates/proxy-engine/src/mixer.rs

@@ -7,7 +7,8 @@
 //! All encoding/decoding happens at leg boundaries. Per-leg inbound denoising at 48kHz.
 //!
 //! The mixer runs a 20ms tick loop:
-//! 1. Drain inbound channels, decode to f32, resample to 48kHz, denoise per-leg
+//! 1. Drain inbound channels, reorder RTP, decode variable-duration packets to 48kHz,
+//!    and queue them in per-leg PCM buffers
 //! 2. Compute total mix (sum of all **participant** legs' f32 PCM as f64)
 //! 3. For each participant leg: mix-minus = total - own, resample to leg codec rate, encode, send
 //! 4. For each isolated leg: play prompt frame or silence, check DTMF
@@ -16,11 +17,12 @@
 
 use crate::ipc::{emit_event, OutTx};
 use crate::jitter_buffer::{JitterBuffer, JitterResult};
-use crate::rtp::{build_rtp_header, rtp_clock_increment};
+use crate::rtp::{build_rtp_header, rtp_clock_increment, rtp_clock_rate};
+use crate::tts::TtsStreamMessage;
 use codec_lib::{codec_sample_rate, new_denoiser, TranscodeState};
 use nnnoiseless::DenoiseState;
 use std::collections::{HashMap, VecDeque};
-use tokio::sync::{mpsc, oneshot};
+use tokio::sync::{mpsc, oneshot, watch};
 use tokio::task::JoinHandle;
 use tokio::time::{self, Duration, MissedTickBehavior};
 
@@ -29,6 +31,12 @@ use tokio::time::{self, Duration, MissedTickBehavior};
 const MIX_RATE: u32 = 48000;
 /// Samples per 20ms frame at the mixing rate.
 const MIX_FRAME_SIZE: usize = 960; // 48000 * 0.020
+/// Safety cap for how much timestamp-derived gap fill we synthesize at once.
+const MAX_GAP_FILL_SAMPLES: usize = MIX_FRAME_SIZE * 6; // 120ms
+/// Bound how many decode / concealment steps a leg can consume in one tick.
+const MAX_PACKET_STEPS_PER_TICK: usize = 24;
+/// Report the first output drop immediately, then every N drops.
+const DROP_REPORT_INTERVAL: u64 = 50;
 
 /// A raw RTP payload received from a leg (no RTP header).
 pub struct RtpPacket {
@@ -57,6 +65,12 @@ enum LegRole {
 struct IsolationState {
     /// PCM frames at MIX_RATE (960 samples each, 48kHz f32) queued for playback.
     prompt_frames: VecDeque<Vec<f32>>,
+    /// Live TTS frames arrive here while playback is already in progress.
+    prompt_stream_rx: Option<mpsc::Receiver<TtsStreamMessage>>,
+    /// Cancels the background TTS producer when the interaction ends early.
+    prompt_cancel_tx: Option<watch::Sender<bool>>,
+    /// Whether the live prompt stream has ended.
+    prompt_stream_finished: bool,
     /// Digits that complete the interaction (e.g., ['1', '2']).
     expected_digits: Vec<char>,
     /// Ticks remaining before timeout (decremented each tick after prompt ends).
@@ -105,6 +119,7 @@ struct ToolLegSlot {
     #[allow(dead_code)]
     tool_type: ToolType,
     audio_tx: mpsc::Sender<ToolAudioBatch>,
+    dropped_batches: u64,
 }
 
 // ---------------------------------------------------------------------------
@@ -132,12 +147,14 @@ pub enum MixerCommand {
         leg_id: String,
         /// PCM frames at MIX_RATE (48kHz f32), each 960 samples.
         prompt_pcm_frames: Vec<Vec<f32>>,
+        /// Optional live prompt stream. Frames are appended as they are synthesized.
+        prompt_stream_rx: Option<mpsc::Receiver<TtsStreamMessage>>,
+        /// Optional cancellation handle for the live prompt stream.
+        prompt_cancel_tx: Option<watch::Sender<bool>>,
         expected_digits: Vec<char>,
         timeout_ms: u32,
         result_tx: oneshot::Sender<InteractionResult>,
     },
-    /// Cancel an in-progress interaction (e.g., leg being removed).
-    CancelInteraction { leg_id: String },
 
     /// Add a tool leg that receives per-source unmerged audio.
     AddToolLeg {
@@ -161,8 +178,15 @@ struct MixerLegSlot {
     denoiser: Box<DenoiseState<'static>>,
     inbound_rx: mpsc::Receiver<RtpPacket>,
     outbound_tx: mpsc::Sender<Vec<u8>>,
+    /// Decoded PCM waiting for playout. Variable-duration RTP packets are
+    /// decoded into this FIFO; the mixer consumes exactly one 20ms frame per tick.
+    pcm_buffer: VecDeque<f32>,
     /// Last decoded+denoised PCM frame at MIX_RATE (960 samples, 48kHz f32).
     last_pcm_frame: Vec<f32>,
+    /// Next RTP timestamp expected from the inbound stream.
+    expected_rtp_timestamp: Option<u32>,
+    /// Best-effort estimate of packet duration in RTP clock units.
+    estimated_packet_ts: u32,
     /// Number of consecutive ticks with no inbound packet.
     silent_ticks: u32,
     /// Per-leg jitter buffer for packet reordering and timing.
@@ -171,15 +195,302 @@ struct MixerLegSlot {
     rtp_seq: u16,
     rtp_ts: u32,
     rtp_ssrc: u32,
+    /// Dropped outbound frames for this leg (queue full / closed).
+    outbound_drops: u64,
     /// Current role of this leg in the mixer.
     role: LegRole,
 }
 
+fn mix_samples_to_rtp_ts(codec_pt: u8, mix_samples: usize) -> u32 {
+    let clock_rate = rtp_clock_rate(codec_pt).max(1) as u64;
+    (((mix_samples as u64 * clock_rate) + (MIX_RATE as u64 / 2)) / MIX_RATE as u64) as u32
+}
+
+fn rtp_ts_to_mix_samples(codec_pt: u8, rtp_ts: u32) -> usize {
+    let clock_rate = rtp_clock_rate(codec_pt).max(1) as u64;
+    (((rtp_ts as u64 * MIX_RATE as u64) + (clock_rate / 2)) / clock_rate) as usize
+}
+
+fn is_forward_rtp_delta(delta: u32) -> bool {
+    delta > 0 && delta < 0x8000_0000
+}
+
+fn should_emit_drop_event(total_drops: u64) -> bool {
+    total_drops == 1 || total_drops % DROP_REPORT_INTERVAL == 0
+}
+
+fn emit_output_drop_event(
+    out_tx: &OutTx,
+    call_id: &str,
+    leg_id: Option<&str>,
+    tool_leg_id: Option<&str>,
+    stream: &str,
+    reason: &str,
+    total_drops: u64,
+) {
+    if !should_emit_drop_event(total_drops) {
+        return;
+    }
+
+    emit_event(
+        out_tx,
+        "mixer_output_drop",
+        serde_json::json!({
+            "call_id": call_id,
+            "leg_id": leg_id,
+            "tool_leg_id": tool_leg_id,
+            "stream": stream,
+            "reason": reason,
+            "total_drops": total_drops,
+        }),
+    );
+}
+
+fn fade_concealment_from_last_frame(slot: &mut MixerLegSlot, samples: usize, decay: f32) {
+    let mut template = if slot.last_pcm_frame.is_empty() {
+        vec![0.0f32; MIX_FRAME_SIZE]
+    } else {
+        slot.last_pcm_frame.clone()
+    };
+
+    let mut remaining = samples;
+    while remaining > 0 {
+        for sample in &mut template {
+            *sample *= decay;
+        }
+        let take = remaining.min(template.len());
+        slot.pcm_buffer.extend(template.iter().take(take).copied());
+        remaining -= take;
+    }
+}
+
+fn append_packet_loss_concealment(slot: &mut MixerLegSlot, samples: usize) {
+    let mut remaining = samples.max(1);
+    while remaining > 0 {
+        let chunk = remaining.min(MIX_FRAME_SIZE);
+        if slot.codec_pt == codec_lib::PT_OPUS {
+            match slot.transcoder.opus_plc(chunk) {
+                Ok(mut pcm) => {
+                    pcm.resize(chunk, 0.0);
+                    slot.pcm_buffer.extend(pcm);
+                }
+                Err(_) => fade_concealment_from_last_frame(slot, chunk, 0.8),
+            }
+        } else {
+            fade_concealment_from_last_frame(slot, chunk, 0.85);
+        }
+        remaining -= chunk;
+    }
+}
+
+fn decode_packet_to_mix_pcm(slot: &mut MixerLegSlot, pkt: &RtpPacket) -> Option<Vec<f32>> {
+    let (pcm, rate) = slot
+        .transcoder
+        .decode_to_f32(&pkt.payload, pkt.payload_type)
+        .ok()?;
+
+    let pcm_48k = if rate == MIX_RATE {
+        pcm
+    } else {
+        slot.transcoder
+            .resample_f32(&pcm, rate, MIX_RATE)
+            .unwrap_or_else(|_| vec![0.0f32; MIX_FRAME_SIZE])
+    };
+
+    let processed = if slot.codec_pt != codec_lib::PT_OPUS {
+        TranscodeState::denoise_f32(&mut slot.denoiser, &pcm_48k)
+    } else {
+        pcm_48k
+    };
+
+    Some(processed)
+}
+
+fn queue_inbound_packet(slot: &mut MixerLegSlot, pkt: RtpPacket) {
+    if let Some(pcm_48k) = decode_packet_to_mix_pcm(slot, &pkt) {
+        if pcm_48k.is_empty() {
+            return;
+        }
+
+        if let Some(expected_ts) = slot.expected_rtp_timestamp {
+            let gap_ts = pkt.timestamp.wrapping_sub(expected_ts);
+            if is_forward_rtp_delta(gap_ts) {
+                let gap_samples = rtp_ts_to_mix_samples(slot.codec_pt, gap_ts);
+                if gap_samples <= MAX_GAP_FILL_SAMPLES {
+                    append_packet_loss_concealment(slot, gap_samples);
+                } else {
+                    slot.pcm_buffer.clear();
+                }
+            }
+        }
+
+        let packet_ts = mix_samples_to_rtp_ts(slot.codec_pt, pcm_48k.len());
+        if packet_ts > 0 {
+            slot.estimated_packet_ts = packet_ts;
+            slot.expected_rtp_timestamp = Some(pkt.timestamp.wrapping_add(packet_ts));
+        }
+        slot.pcm_buffer.extend(pcm_48k);
+    }
+}
+
+fn fill_leg_playout_buffer(slot: &mut MixerLegSlot) {
+    let mut steps = 0usize;
+    while slot.pcm_buffer.len() < MIX_FRAME_SIZE && steps < MAX_PACKET_STEPS_PER_TICK {
+        steps += 1;
+        match slot.jitter.consume() {
+            JitterResult::Packet(pkt) => queue_inbound_packet(slot, pkt),
+            JitterResult::Missing => {
+                let conceal_ts = slot
+                    .estimated_packet_ts
+                    .max(rtp_clock_increment(slot.codec_pt));
+                let conceal_samples =
+                    rtp_ts_to_mix_samples(slot.codec_pt, conceal_ts).clamp(1, MAX_GAP_FILL_SAMPLES);
+                append_packet_loss_concealment(slot, conceal_samples);
+                if let Some(expected_ts) = slot.expected_rtp_timestamp {
+                    slot.expected_rtp_timestamp = Some(expected_ts.wrapping_add(conceal_ts));
+                }
+            }
+            JitterResult::Filling => break,
+        }
+    }
+}
+
+fn take_mix_frame(slot: &mut MixerLegSlot) -> Vec<f32> {
+    let mut frame = Vec::with_capacity(MIX_FRAME_SIZE);
+    while frame.len() < MIX_FRAME_SIZE {
+        if let Some(sample) = slot.pcm_buffer.pop_front() {
+            frame.push(sample);
+        } else {
+            frame.push(0.0);
+        }
+    }
+    frame
+}
+
+fn soft_limit_sample(sample: f32) -> f32 {
+    const KNEE: f32 = 0.85;
+
+    let abs = sample.abs();
+    if abs <= KNEE {
+        sample
+    } else {
+        let excess = abs - KNEE;
+        let compressed = KNEE + (excess / (1.0 + (excess / (1.0 - KNEE))));
+        sample.signum() * compressed.min(1.0)
+    }
+}
+
+fn try_send_leg_output(
+    out_tx: &OutTx,
+    call_id: &str,
+    leg_id: &str,
+    slot: &mut MixerLegSlot,
+    rtp: Vec<u8>,
+    stream: &str,
+) {
+    let reason = match slot.outbound_tx.try_send(rtp) {
+        Ok(()) => return,
+        Err(mpsc::error::TrySendError::Full(_)) => "full",
+        Err(mpsc::error::TrySendError::Closed(_)) => "closed",
+    };
+
+    slot.outbound_drops += 1;
+    emit_output_drop_event(
+        out_tx,
+        call_id,
+        Some(leg_id),
+        None,
+        stream,
+        reason,
+        slot.outbound_drops,
+    );
+}
+
+fn try_send_tool_output(
+    out_tx: &OutTx,
+    call_id: &str,
+    tool_leg_id: &str,
+    tool: &mut ToolLegSlot,
+    batch: ToolAudioBatch,
+) {
+    let reason = match tool.audio_tx.try_send(batch) {
+        Ok(()) => return,
+        Err(mpsc::error::TrySendError::Full(_)) => "full",
+        Err(mpsc::error::TrySendError::Closed(_)) => "closed",
+    };
+
+    tool.dropped_batches += 1;
+    emit_output_drop_event(
+        out_tx,
+        call_id,
+        None,
+        Some(tool_leg_id),
+        "tool-batch",
+        reason,
+        tool.dropped_batches,
+    );
+}
+
+fn cancel_prompt_producer(state: &mut IsolationState) {
+    if let Some(cancel_tx) = state.prompt_cancel_tx.take() {
+        let _ = cancel_tx.send(true);
+    }
+}
+
+fn cancel_isolated_interaction(state: &mut IsolationState) {
+    cancel_prompt_producer(state);
+    if let Some(tx) = state.result_tx.take() {
+        let _ = tx.send(InteractionResult::Cancelled);
+    }
+}
+
+fn drain_prompt_stream(
+    out_tx: &OutTx,
+    call_id: &str,
+    leg_id: &str,
+    state: &mut IsolationState,
+) {
+    loop {
+        let Some(mut stream_rx) = state.prompt_stream_rx.take() else {
+            return;
+        };
+
+        match stream_rx.try_recv() {
+            Ok(TtsStreamMessage::Frames(frames)) => {
+                state.prompt_frames.extend(frames);
+                state.prompt_stream_rx = Some(stream_rx);
+            }
+            Ok(TtsStreamMessage::Finished) => {
+                state.prompt_stream_finished = true;
+                return;
+            }
+            Ok(TtsStreamMessage::Failed(error)) => {
+                emit_event(
+                    out_tx,
+                    "mixer_error",
+                    serde_json::json!({
+                        "call_id": call_id,
+                        "leg_id": leg_id,
+                        "error": format!("tts stream failed: {error}"),
+                    }),
+                );
+                state.prompt_stream_finished = true;
+                return;
+            }
+            Err(mpsc::error::TryRecvError::Empty) => {
+                state.prompt_stream_rx = Some(stream_rx);
+                return;
+            }
+            Err(mpsc::error::TryRecvError::Disconnected) => {
+                state.prompt_stream_finished = true;
+                return;
+            }
+        }
+    }
+}
+
 /// Spawn the mixer task for a call. Returns the command sender and task handle.
-pub fn spawn_mixer(
-    call_id: String,
-    out_tx: OutTx,
-) -> (mpsc::Sender<MixerCommand>, JoinHandle<()>) {
+pub fn spawn_mixer(call_id: String, out_tx: OutTx) -> (mpsc::Sender<MixerCommand>, JoinHandle<()>) {
     let (cmd_tx, cmd_rx) = mpsc::channel::<MixerCommand>(32);
 
     let handle = tokio::spawn(async move {
@@ -190,11 +501,7 @@ pub fn spawn_mixer(
 }
 
 /// The 20ms mixing loop.
-async fn mixer_loop(
-    call_id: String,
-    mut cmd_rx: mpsc::Receiver<MixerCommand>,
-    out_tx: OutTx,
-) {
+async fn mixer_loop(call_id: String, mut cmd_rx: mpsc::Receiver<MixerCommand>, out_tx: OutTx) {
     let mut legs: HashMap<String, MixerLegSlot> = HashMap::new();
     let mut tool_legs: HashMap<String, ToolLegSlot> = HashMap::new();
     let mut interval = time::interval(Duration::from_millis(20));
@@ -235,11 +542,15 @@ async fn mixer_loop(
                             denoiser: new_denoiser(),
                             inbound_rx,
                             outbound_tx,
+                            pcm_buffer: VecDeque::new(),
                             last_pcm_frame: vec![0.0f32; MIX_FRAME_SIZE],
+                            expected_rtp_timestamp: None,
+                            estimated_packet_ts: rtp_clock_increment(codec_pt),
                             silent_ticks: 0,
                             rtp_seq: 0,
                             rtp_ts: 0,
                             rtp_ssrc: rand::random(),
+                            outbound_drops: 0,
                             role: LegRole::Participant,
                             jitter: JitterBuffer::new(),
                         },
@@ -249,9 +560,7 @@ async fn mixer_loop(
                     // If the leg is isolated, send Cancelled before dropping.
                     if let Some(slot) = legs.get_mut(&leg_id) {
                         if let LegRole::Isolated(ref mut state) = slot.role {
-                            if let Some(tx) = state.result_tx.take() {
-                                let _ = tx.send(InteractionResult::Cancelled);
-                            }
+                            cancel_isolated_interaction(state);
                         }
                     }
                     legs.remove(&leg_id);
@@ -261,9 +570,7 @@ async fn mixer_loop(
                     // Cancel all outstanding interactions before shutting down.
                     for slot in legs.values_mut() {
                         if let LegRole::Isolated(ref mut state) = slot.role {
-                            if let Some(tx) = state.result_tx.take() {
-                                let _ = tx.send(InteractionResult::Cancelled);
-                            }
+                            cancel_isolated_interaction(state);
                         }
                     }
                     return;
@@ -271,6 +578,8 @@ async fn mixer_loop(
                 Ok(MixerCommand::StartInteraction {
                     leg_id,
                     prompt_pcm_frames,
+                    prompt_stream_rx,
+                    prompt_cancel_tx,
                     expected_digits,
                     timeout_ms,
                     result_tx,
@@ -278,13 +587,14 @@ async fn mixer_loop(
                     if let Some(slot) = legs.get_mut(&leg_id) {
                         // Cancel any existing interaction first.
                         if let LegRole::Isolated(ref mut old_state) = slot.role {
-                            if let Some(tx) = old_state.result_tx.take() {
-                                let _ = tx.send(InteractionResult::Cancelled);
-                            }
+                            cancel_isolated_interaction(old_state);
                         }
                         let timeout_ticks = timeout_ms / 20;
                         slot.role = LegRole::Isolated(IsolationState {
                             prompt_frames: VecDeque::from(prompt_pcm_frames),
+                            prompt_stream_rx,
+                            prompt_cancel_tx,
+                            prompt_stream_finished: false,
                             expected_digits,
                             timeout_ticks_remaining: timeout_ticks,
                             prompt_done: false,
@@ -292,17 +602,10 @@ async fn mixer_loop(
                         });
                     } else {
                         // Leg not found — immediately cancel.
-                        let _ = result_tx.send(InteractionResult::Cancelled);
-                    }
-                }
-                Ok(MixerCommand::CancelInteraction { leg_id }) => {
-                    if let Some(slot) = legs.get_mut(&leg_id) {
-                        if let LegRole::Isolated(ref mut state) = slot.role {
-                            if let Some(tx) = state.result_tx.take() {
-                                let _ = tx.send(InteractionResult::Cancelled);
-                            }
+                        if let Some(cancel_tx) = prompt_cancel_tx {
+                            let _ = cancel_tx.send(true);
                         }
-                        slot.role = LegRole::Participant;
+                        let _ = result_tx.send(InteractionResult::Cancelled);
                     }
                 }
                 Ok(MixerCommand::AddToolLeg {
@@ -310,7 +613,14 @@ async fn mixer_loop(
                     tool_type,
                     audio_tx,
                 }) => {
-                    tool_legs.insert(leg_id, ToolLegSlot { tool_type, audio_tx });
+                    tool_legs.insert(
+                        leg_id,
+                        ToolLegSlot {
+                            tool_type,
+                            audio_tx,
+                            dropped_batches: 0,
+                        },
+                    );
                 }
                 Ok(MixerCommand::RemoveToolLeg { leg_id }) => {
                     tool_legs.remove(&leg_id);
@@ -351,54 +661,11 @@ async fn mixer_loop(
                 }
             }
 
-            // Step 2b: Consume exactly one frame from the jitter buffer.
-            match slot.jitter.consume() {
-                JitterResult::Packet(pkt) => {
-                    match slot.transcoder.decode_to_f32(&pkt.payload, pkt.payload_type) {
-                        Ok((pcm, rate)) => {
-                            let pcm_48k = if rate == MIX_RATE {
-                                pcm
-                            } else {
-                                slot.transcoder
-                                    .resample_f32(&pcm, rate, MIX_RATE)
-                                    .unwrap_or_else(|_| vec![0.0f32; MIX_FRAME_SIZE])
-                            };
-                            let processed = if slot.codec_pt != codec_lib::PT_OPUS {
-                                TranscodeState::denoise_f32(&mut slot.denoiser, &pcm_48k)
-                            } else {
-                                pcm_48k
-                            };
-                            let mut frame = processed;
-                            frame.resize(MIX_FRAME_SIZE, 0.0);
-                            slot.last_pcm_frame = frame;
-                        }
-                        Err(_) => {}
-                    }
-                }
-                JitterResult::Missing => {
-                    // Invoke Opus PLC or fade for non-Opus codecs.
-                    if slot.codec_pt == codec_lib::PT_OPUS {
-                        match slot.transcoder.opus_plc(MIX_FRAME_SIZE) {
-                            Ok(pcm) => {
-                                slot.last_pcm_frame = pcm;
-                            }
-                            Err(_) => {
-                                for s in slot.last_pcm_frame.iter_mut() {
-                                    *s *= 0.8;
-                                }
-                            }
-                        }
-                    } else {
-                        // Non-Opus: fade last frame toward silence.
-                        for s in slot.last_pcm_frame.iter_mut() {
-                            *s *= 0.85;
-                        }
-                    }
-                }
-                JitterResult::Filling => {
-                    slot.last_pcm_frame = vec![0.0f32; MIX_FRAME_SIZE];
-                }
-            }
+            // Step 2b: Decode enough RTP to cover one 20ms playout frame.
+            // Variable-duration packets (10ms, 20ms, 60ms, ...) accumulate in
+            // the per-leg PCM FIFO; we pop exactly one 20ms frame below.
+            fill_leg_playout_buffer(slot);
+            slot.last_pcm_frame = take_mix_frame(slot);
 
             // Run jitter adaptation + prune stale packets.
             slot.jitter.adapt();
@@ -412,6 +679,9 @@ async fn mixer_loop(
             }
             if slot.silent_ticks > 150 {
                 slot.last_pcm_frame = vec![0.0f32; MIX_FRAME_SIZE];
+                slot.pcm_buffer.clear();
+                slot.expected_rtp_timestamp = None;
+                slot.estimated_packet_ts = rtp_clock_increment(slot.codec_pt);
             }
         }
 
@@ -434,12 +704,12 @@ async fn mixer_loop(
         for (lid, slot) in legs.iter_mut() {
             match &mut slot.role {
                 LegRole::Participant => {
-                    // Mix-minus: total minus this leg's own contribution, clamped to [-1.0, 1.0].
+                    // Mix-minus: total minus this leg's own contribution.
+                    // Apply a light soft limiter instead of hard clipping the sum.
                     let mut mix_minus = Vec::with_capacity(MIX_FRAME_SIZE);
                     for i in 0..MIX_FRAME_SIZE {
-                        let sample =
-                            (total_mix[i] - slot.last_pcm_frame[i] as f64) as f32;
-                        mix_minus.push(sample.clamp(-1.0, 1.0));
+                        let sample = (total_mix[i] - slot.last_pcm_frame[i] as f64) as f32;
+                        mix_minus.push(soft_limit_sample(sample));
                     }
 
                     // Resample from 48kHz to the leg's codec native rate.
@@ -453,11 +723,10 @@ async fn mixer_loop(
                     };
 
                     // Encode to the leg's codec (f32 → i16 → codec inside encode_from_f32).
-                    let encoded =
-                        match slot.transcoder.encode_from_f32(&resampled, slot.codec_pt) {
-                            Ok(e) if !e.is_empty() => e,
-                            _ => continue,
-                        };
+                    let encoded = match slot.transcoder.encode_from_f32(&resampled, slot.codec_pt) {
+                        Ok(e) if !e.is_empty() => e,
+                        _ => continue,
+                    };
 
                     // Build RTP packet with header.
                     let header =
@@ -468,10 +737,11 @@ async fn mixer_loop(
                     slot.rtp_seq = slot.rtp_seq.wrapping_add(1);
                     slot.rtp_ts = slot.rtp_ts.wrapping_add(rtp_clock_increment(slot.codec_pt));
 
-                    // Non-blocking send — drop frame if channel is full.
-                    let _ = slot.outbound_tx.try_send(rtp);
+                    try_send_leg_output(&out_tx, &call_id, lid, slot, rtp, "participant-audio");
                 }
                 LegRole::Isolated(state) => {
+                    drain_prompt_stream(&out_tx, &call_id, lid, state);
+
                     // Check for DTMF digit from this leg.
                     let mut matched_digit: Option<char> = None;
                     for (src_lid, dtmf_pkt) in &dtmf_forward {
@@ -495,12 +765,14 @@ async fn mixer_loop(
 
                     if let Some(digit) = matched_digit {
                         // Interaction complete — digit matched.
-                        completed_interactions
-                            .push((lid.clone(), InteractionResult::Digit(digit)));
+                        completed_interactions.push((lid.clone(), InteractionResult::Digit(digit)));
                     } else {
-                        // Play prompt frame or silence.
+                        // Play prompt frame, wait for live TTS, or move to timeout once the
+                        // prompt stream has fully drained.
                         let pcm_frame = if let Some(frame) = state.prompt_frames.pop_front() {
                             frame
+                        } else if !state.prompt_stream_finished {
+                            vec![0.0f32; MIX_FRAME_SIZE]
                         } else {
                             state.prompt_done = true;
                             vec![0.0f32; MIX_FRAME_SIZE]
@@ -516,6 +788,7 @@ async fn mixer_loop(
                                 .unwrap_or_default()
                         };
 
+                        let mut prompt_rtp: Option<Vec<u8>> = None;
                         if let Ok(encoded) =
                             slot.transcoder.encode_from_f32(&resampled, slot.codec_pt)
                         {
@@ -529,10 +802,9 @@ async fn mixer_loop(
                                 let mut rtp = header.to_vec();
                                 rtp.extend_from_slice(&encoded);
                                 slot.rtp_seq = slot.rtp_seq.wrapping_add(1);
-                                slot.rtp_ts = slot
-                                    .rtp_ts
-                                    .wrapping_add(rtp_clock_increment(slot.codec_pt));
-                                let _ = slot.outbound_tx.try_send(rtp);
+                                slot.rtp_ts =
+                                    slot.rtp_ts.wrapping_add(rtp_clock_increment(slot.codec_pt));
+                                prompt_rtp = Some(rtp);
                             }
                         }
 
@@ -545,6 +817,17 @@ async fn mixer_loop(
                                 state.timeout_ticks_remaining -= 1;
                             }
                         }
+
+                        if let Some(rtp) = prompt_rtp {
+                            try_send_leg_output(
+                                &out_tx,
+                                &call_id,
+                                lid,
+                                slot,
+                                rtp,
+                                "isolated-prompt",
+                            );
+                        }
                     }
                 }
             }
@@ -554,6 +837,7 @@ async fn mixer_loop(
         for (lid, result) in completed_interactions {
             if let Some(slot) = legs.get_mut(&lid) {
                 if let LegRole::Isolated(ref mut state) = slot.role {
+                    cancel_prompt_producer(state);
                     if let Some(tx) = state.result_tx.take() {
                         let _ = tx.send(result);
                     }
@@ -574,7 +858,7 @@ async fn mixer_loop(
                 })
                 .collect();
 
-            for tool in tool_legs.values() {
+            for (tool_leg_id, tool) in tool_legs.iter_mut() {
                 let batch = ToolAudioBatch {
                     sources: sources
                         .iter()
@@ -584,8 +868,7 @@ async fn mixer_loop(
                         })
                         .collect(),
                 };
-                // Non-blocking send — drop batch if tool can't keep up.
-                let _ = tool.audio_tx.try_send(batch);
+                try_send_tool_output(&out_tx, &call_id, tool_leg_id, tool, batch);
             }
         }
 
@@ -618,7 +901,7 @@ async fn mixer_loop(
                 rtp_out.extend_from_slice(&dtmf_pkt.payload);
                 target_slot.rtp_seq = target_slot.rtp_seq.wrapping_add(1);
                 // Don't increment rtp_ts for DTMF — it shares timestamp context with audio.
-                let _ = target_slot.outbound_tx.try_send(rtp_out);
+                try_send_leg_output(&out_tx, &call_id, target_lid, target_slot, rtp_out, "dtmf");
             }
         }
     }

rust/crates/proxy-engine/src/provider.rs

@@ -267,11 +267,7 @@ impl ProviderManager {
 
     /// Try to handle a SIP response as a provider registration response.
     /// Returns true if consumed.
-    pub async fn handle_response(
-        &self,
-        msg: &SipMessage,
-        socket: &UdpSocket,
-    ) -> bool {
+    pub async fn handle_response(&self, msg: &SipMessage, socket: &UdpSocket) -> bool {
         for ps_arc in &self.providers {
             let mut ps = ps_arc.lock().await;
             let was_registered = ps.is_registered;
@@ -322,7 +318,10 @@ impl ProviderManager {
     }
 
     /// Find a provider by its config ID (e.g. "easybell").
-    pub async fn find_by_provider_id(&self, provider_id: &str) -> Option<Arc<Mutex<ProviderState>>> {
+    pub async fn find_by_provider_id(
+        &self,
+        provider_id: &str,
+    ) -> Option<Arc<Mutex<ProviderState>>> {
         for ps_arc in &self.providers {
             let ps = ps_arc.lock().await;
             if ps.config.id == provider_id {
@@ -331,17 +330,6 @@ impl ProviderManager {
         }
         None
     }
-
-    /// Check if a provider is currently registered.
-    pub async fn is_registered(&self, provider_id: &str) -> bool {
-        for ps_arc in &self.providers {
-            let ps = ps_arc.lock().await;
-            if ps.config.id == provider_id {
-                return ps.is_registered;
-            }
-        }
-        false
-    }
 }
 
 /// Registration loop for a single provider.

rust/crates/proxy-engine/src/recorder.rs

@@ -25,8 +25,7 @@ impl Recorder {
     ) -> Result<Self, String> {
         // Ensure parent directory exists.
         if let Some(parent) = Path::new(file_path).parent() {
-            std::fs::create_dir_all(parent)
-                .map_err(|e| format!("create dir: {e}"))?;
+            std::fs::create_dir_all(parent).map_err(|e| format!("create dir: {e}"))?;
         }
 
         let sample_rate = 8000u32; // Record at 8kHz (standard telephony)
@@ -57,10 +56,13 @@ impl Recorder {
 
     /// Create a recorder that writes raw PCM at a given sample rate.
     /// Used by tool legs that already have decoded PCM (no RTP processing needed).
-    pub fn new_pcm(file_path: &str, sample_rate: u32, max_duration_ms: Option<u64>) -> Result<Self, String> {
+    pub fn new_pcm(
+        file_path: &str,
+        sample_rate: u32,
+        max_duration_ms: Option<u64>,
+    ) -> Result<Self, String> {
         if let Some(parent) = Path::new(file_path).parent() {
-            std::fs::create_dir_all(parent)
-                .map_err(|e| format!("create dir: {e}"))?;
+            std::fs::create_dir_all(parent).map_err(|e| format!("create dir: {e}"))?;
         }
 
         let spec = hound::WavSpec {
@@ -178,5 +180,8 @@ impl Recorder {
 pub struct RecordingResult {
     pub file_path: String,
     pub duration_ms: u64,
+    // Running-sample total kept for parity with the TS recorder; not yet
+    // surfaced through any event or dashboard field.
+    #[allow(dead_code)]
     pub total_samples: u64,
 }

rust/crates/proxy-engine/src/registrar.rs

@@ -19,11 +19,19 @@ const MAX_EXPIRES: u32 = 300;
 #[derive(Debug, Clone)]
 pub struct RegisteredDevice {
     pub device_id: String,
+    // These fields are populated at REGISTER time for logging/debugging but are
+    // not read back — device identity flows via the `device_registered` push
+    // event, not via struct queries. Kept behind allow(dead_code) because
+    // removing them would churn handle_register for no runtime benefit.
+    #[allow(dead_code)]
     pub display_name: String,
+    #[allow(dead_code)]
     pub extension: String,
     pub contact_addr: SocketAddr,
+    #[allow(dead_code)]
     pub registered_at: Instant,
     pub expires_at: Instant,
+    #[allow(dead_code)]
     pub aor: String,
 }
 
@@ -52,18 +60,17 @@ impl Registrar {
 
     /// Try to handle a SIP REGISTER from a device.
     /// Returns Some(response_bytes) if handled, None if not a known device.
-    pub fn handle_register(
-        &mut self,
-        msg: &SipMessage,
-        from_addr: SocketAddr,
-    ) -> Option<Vec<u8>> {
+    pub fn handle_register(&mut self, msg: &SipMessage, from_addr: SocketAddr) -> Option<Vec<u8>> {
         if msg.method() != Some("REGISTER") {
             return None;
         }
 
         // Find the device by matching the source IP against expectedAddress.
         let from_ip = from_addr.ip().to_string();
-        let device = self.devices.iter().find(|d| d.expected_address == from_ip)?;
+        let device = self
+            .devices
+            .iter()
+            .find(|d| d.expected_address == from_ip)?;
 
         let from_header = msg.get_header("From").unwrap_or("");
         let aor = SipMessage::extract_uri(from_header)
@@ -71,9 +78,7 @@ impl Registrar {
             .unwrap_or_else(|| format!("sip:{}@{}", device.extension, from_ip));
 
         let expires_header = msg.get_header("Expires");
-        let requested: u32 = expires_header
-            .and_then(|s| s.parse().ok())
-            .unwrap_or(3600);
+        let requested: u32 = expires_header.and_then(|s| s.parse().ok()).unwrap_or(3600);
         let expires = requested.min(MAX_EXPIRES);
 
         let entry = RegisteredDevice {
@@ -114,10 +119,7 @@ impl Registrar {
             Some(ResponseOptions {
                 to_tag: Some(generate_tag()),
                 contact: Some(contact),
-                extra_headers: Some(vec![(
-                    "Expires".to_string(),
-                    expires.to_string(),
-                )]),
+                extra_headers: Some(vec![("Expires".to_string(), expires.to_string())]),
                 ..Default::default()
             }),
         );
@@ -134,38 +136,11 @@ impl Registrar {
         Some(entry.contact_addr)
     }
 
-    /// Check if a source address belongs to a known device.
-    pub fn is_known_device_address(&self, addr: &str) -> bool {
-        self.devices.iter().any(|d| d.expected_address == addr)
-    }
-
     /// Find a registered device by its source IP address.
     pub fn find_by_address(&self, addr: &SocketAddr) -> Option<&RegisteredDevice> {
         let ip = addr.ip().to_string();
-        self.registered.values().find(|e| {
-            e.contact_addr.ip().to_string() == ip && Instant::now() <= e.expires_at
-        })
-    }
-
-    /// Get all device statuses for the dashboard.
-    pub fn get_all_statuses(&self) -> Vec<serde_json::Value> {
-        let now = Instant::now();
-        let mut result = Vec::new();
-
-        for dc in &self.devices {
-            let reg = self.registered.get(&dc.id);
-            let connected = reg.map(|r| now <= r.expires_at).unwrap_or(false);
-            result.push(serde_json::json!({
-                "id": dc.id,
-                "displayName": dc.display_name,
-                "address": reg.filter(|_| connected).map(|r| r.contact_addr.ip().to_string()),
-                "port": reg.filter(|_| connected).map(|r| r.contact_addr.port()),
-                "aor": reg.map(|r| r.aor.as_str()).unwrap_or(""),
-                "connected": connected,
-                "isBrowser": false,
-            }));
-        }
-
-        result
+        self.registered
+            .values()
+            .find(|e| e.contact_addr.ip().to_string() == ip && Instant::now() <= e.expires_at)
     }
 }

rust/crates/proxy-engine/src/rtp.rs

@@ -1,17 +1,19 @@
-//! RTP port pool and media forwarding.
+//! RTP port pool for media sockets.
 //!
-//! Manages a pool of even-numbered UDP ports for RTP media.
-//! Each port gets a bound tokio UdpSocket. Supports:
-//! - Direct forwarding (SIP-to-SIP, no transcoding)
-//! - Transcoding forwarding (via codec-lib, e.g. G.722 ↔ Opus)
-//! - Silence generation
-//! - NAT priming
+//! Manages a pool of even-numbered UDP ports for RTP media. `allocate()`
+//! hands back an `Arc<UdpSocket>` to the caller (stored on the owning
+//! `LegInfo`), while the pool itself keeps only a `Weak<UdpSocket>`. When
+//! the call terminates and `LegInfo` is dropped, the strong refcount
+//! reaches zero, the socket is closed, and `allocate()` prunes the dead
+//! weak ref the next time it scans that slot — so the port automatically
+//! becomes available for reuse without any explicit `release()` plumbing.
 //!
-//! Ported from ts/call/rtp-port-pool.ts + sip-leg.ts RTP handling.
+//! This fixes the previous leak where the pool held `Arc<UdpSocket>` and
+//! `release()` was never called, eventually exhausting the port range and
+//! causing "503 Service Unavailable" on new calls.
 
 use std::collections::HashMap;
-use std::net::SocketAddr;
-use std::sync::Arc;
+use std::sync::{Arc, Weak};
 use tokio::net::UdpSocket;
 
 /// A single RTP port allocation.
@@ -24,7 +26,7 @@ pub struct RtpAllocation {
 pub struct RtpPortPool {
     min: u16,
     max: u16,
-    allocated: HashMap<u16, Arc<UdpSocket>>,
+    allocated: HashMap<u16, Weak<UdpSocket>>,
 }
 
 impl RtpPortPool {
@@ -41,11 +43,19 @@ impl RtpPortPool {
     pub async fn allocate(&mut self) -> Option<RtpAllocation> {
         let mut port = self.min;
         while port < self.max {
+            // Prune a dead weak ref at this slot: if the last strong Arc
+            // (held by the owning LegInfo) was dropped when the call ended,
+            // the socket is already closed and the slot is free again.
+            if let Some(weak) = self.allocated.get(&port) {
+                if weak.strong_count() == 0 {
+                    self.allocated.remove(&port);
+                }
+            }
             if !self.allocated.contains_key(&port) {
                 match UdpSocket::bind(format!("0.0.0.0:{port}")).await {
                     Ok(sock) => {
                         let sock = Arc::new(sock);
-                        self.allocated.insert(port, sock.clone());
+                        self.allocated.insert(port, Arc::downgrade(&sock));
                         return Some(RtpAllocation { port, socket: sock });
                     }
                     Err(_) => {
@@ -57,83 +67,6 @@ impl RtpPortPool {
         }
         None // Pool exhausted.
     }
-
-    /// Release a port back to the pool.
-    pub fn release(&mut self, port: u16) {
-        self.allocated.remove(&port);
-        // Socket is dropped when the last Arc reference goes away.
-    }
-
-    pub fn size(&self) -> usize {
-        self.allocated.len()
-    }
-
-    pub fn capacity(&self) -> usize {
-        ((self.max - self.min) / 2) as usize
-    }
-}
-
-/// An active RTP relay between two endpoints.
-/// Receives on `local_socket` and forwards to `remote_addr`.
-pub struct RtpRelay {
-    pub local_port: u16,
-    pub local_socket: Arc<UdpSocket>,
-    pub remote_addr: Option<SocketAddr>,
-    /// If set, transcode packets using this codec session before forwarding.
-    pub transcode: Option<TranscodeConfig>,
-    /// Packets received counter.
-    pub pkt_received: u64,
-    /// Packets sent counter.
-    pub pkt_sent: u64,
-}
-
-pub struct TranscodeConfig {
-    pub from_pt: u8,
-    pub to_pt: u8,
-    pub session_id: String,
-}
-
-impl RtpRelay {
-    pub fn new(port: u16, socket: Arc<UdpSocket>) -> Self {
-        Self {
-            local_port: port,
-            local_socket: socket,
-            remote_addr: None,
-            transcode: None,
-            pkt_received: 0,
-            pkt_sent: 0,
-        }
-    }
-
-    pub fn set_remote(&mut self, addr: SocketAddr) {
-        self.remote_addr = Some(addr);
-    }
-}
-
-/// Send a 1-byte NAT priming packet to open a pinhole.
-pub async fn prime_nat(socket: &UdpSocket, remote: SocketAddr) {
-    let _ = socket.send_to(&[0u8], remote).await;
-}
-
-/// Build an RTP silence frame for PCMU (payload type 0).
-pub fn silence_frame_pcmu() -> Vec<u8> {
-    // 12-byte RTP header + 160 bytes of µ-law silence (0xFF)
-    let mut frame = vec![0u8; 172];
-    frame[0] = 0x80; // V=2
-    frame[1] = 0; // PT=0 (PCMU)
-    // seq, timestamp, ssrc left as 0 — caller should set these
-    frame[12..].fill(0xFF); // µ-law silence
-    frame
-}
-
-/// Build an RTP silence frame for G.722 (payload type 9).
-pub fn silence_frame_g722() -> Vec<u8> {
-    // 12-byte RTP header + 160 bytes of G.722 silence
-    let mut frame = vec![0u8; 172];
-    frame[0] = 0x80; // V=2
-    frame[1] = 9; // PT=9 (G.722)
-    // G.722 silence: all zeros is valid silence
-    frame
 }
 
 /// Build an RTP header with the given parameters.
@@ -149,10 +82,15 @@ pub fn build_rtp_header(pt: u8, seq: u16, timestamp: u32, ssrc: u32) -> [u8; 12]
 
 /// Get the RTP clock increment per 20ms frame for a payload type.
 pub fn rtp_clock_increment(pt: u8) -> u32 {
+    rtp_clock_rate(pt) / 50
+}
+
+/// Get the RTP clock rate for a payload type.
+pub fn rtp_clock_rate(pt: u8) -> u32 {
     match pt {
-        9 => 160,   // G.722: 8000 Hz clock rate (despite 16kHz audio) × 0.02s
-        0 | 8 => 160, // PCMU/PCMA: 8000 × 0.02
-        111 => 960, // Opus: 48000 × 0.02
-        _ => 160,
+        9 => 8000,     // G.722 uses an 8kHz RTP clock despite 16kHz audio.
+        0 | 8 => 8000, // PCMU/PCMA
+        111 => 48000,  // Opus
+        _ => 8000,
     }
 }

rust/crates/proxy-engine/src/sip_leg.rs

@@ -16,7 +16,6 @@ use sip_proto::helpers::{
 };
 use sip_proto::message::{RequestOptions, SipMessage};
 use std::net::SocketAddr;
-use std::sync::Arc;
 use tokio::net::UdpSocket;
 
 /// State of a SIP leg.
@@ -40,6 +39,9 @@ pub struct SipLegConfig {
     /// SIP target endpoint (provider outbound proxy or device address).
     pub sip_target: SocketAddr,
     /// Provider credentials (for 407 auth).
+    // username is carried for parity with the provider config but digest auth
+    // rebuilds the username from the registered AOR, so this slot is never read.
+    #[allow(dead_code)]
     pub username: Option<String>,
     pub password: Option<String>,
     pub registered_aor: Option<String>,
@@ -51,6 +53,10 @@ pub struct SipLegConfig {
 
 /// A SIP leg with full dialog management.
 pub struct SipLeg {
+    // Leg identity is tracked via the enclosing LegInfo's key in the call's
+    // leg map; SipLeg itself never reads this field back. Kept to preserve
+    // the (id, config) constructor shape used by the call manager.
+    #[allow(dead_code)]
     pub id: String,
     pub state: LegState,
     pub config: SipLegConfig,
@@ -122,17 +128,24 @@ impl SipLeg {
                 max_forwards: Some(70),
                 body: Some(sdp),
                 content_type: Some("application/sdp".to_string()),
-                extra_headers: Some(vec![
-                    ("User-Agent".to_string(), "SipRouter/1.0".to_string()),
-                ]),
+                extra_headers: Some(vec![(
+                    "User-Agent".to_string(),
+                    "SipRouter/1.0".to_string(),
+                )]),
             },
         );
 
-        self.dialog = Some(SipDialog::from_uac_invite(&invite, ip, self.config.lan_port));
+        self.dialog = Some(SipDialog::from_uac_invite(
+            &invite,
+            ip,
+            self.config.lan_port,
+        ));
         self.invite = Some(invite.clone());
         self.state = LegState::Inviting;
 
-        let _ = socket.send_to(&invite.serialize(), self.config.sip_target).await;
+        let _ = socket
+            .send_to(&invite.serialize(), self.config.sip_target)
+            .await;
     }
 
     /// Handle an incoming SIP message routed to this leg.
@@ -411,11 +424,6 @@ impl SipLeg {
         dialog.terminate();
         Some(msg.serialize())
     }
-
-    /// Get the SIP Call-ID for routing.
-    pub fn sip_call_id(&self) -> Option<&str> {
-        self.dialog.as_ref().map(|d| d.call_id.as_str())
-    }
 }
 
 /// Actions produced by the SipLeg message handler.
@@ -442,10 +450,7 @@ pub enum SipLegAction {
 /// Build an ACK for a non-2xx response (same transaction as the INVITE).
 fn build_non_2xx_ack(original_invite: &SipMessage, response: &SipMessage) -> SipMessage {
     let via = original_invite.get_header("Via").unwrap_or("").to_string();
-    let from = original_invite
-        .get_header("From")
-        .unwrap_or("")
-        .to_string();
+    let from = original_invite.get_header("From").unwrap_or("").to_string();
     let to = response.get_header("To").unwrap_or("").to_string();
     let call_id = original_invite.call_id().to_string();
     let cseq_num: u32 = original_invite

rust/crates/proxy-engine/src/sip_transport.rs

@@ -27,27 +27,9 @@ impl SipTransport {
         self.socket.clone()
     }
 
-    /// Send a raw SIP message to a destination.
-    pub async fn send_to(&self, data: &[u8], dest: SocketAddr) -> Result<usize, String> {
-        self.socket
-            .send_to(data, dest)
-            .await
-            .map_err(|e| format!("send to {dest}: {e}"))
-    }
-
-    /// Send a raw SIP message to an address:port pair.
-    pub async fn send_to_addr(&self, data: &[u8], addr: &str, port: u16) -> Result<usize, String> {
-        let dest: SocketAddr = format!("{addr}:{port}")
-            .parse()
-            .map_err(|e| format!("bad address {addr}:{port}: {e}"))?;
-        self.send_to(data, dest).await
-    }
-
     /// Spawn the UDP receive loop. Calls the handler for every received packet.
-    pub fn spawn_receiver<F>(
-        &self,
-        handler: F,
-    ) where
+    pub fn spawn_receiver<F>(&self, handler: F)
+    where
         F: Fn(&[u8], SocketAddr) + Send + 'static,
     {
         let socket = self.socket.clone();

rust/crates/proxy-engine/src/tool_leg.rs

@@ -51,7 +51,8 @@ pub fn spawn_recording_tool(
                 });
 
                 // Convert f32 [-1.0, 1.0] to i16 for WAV writing.
-                let pcm_i16: Vec<i16> = source.pcm_48k
+                let pcm_i16: Vec<i16> = source
+                    .pcm_48k
                     .iter()
                     .map(|&s| (s * 32767.0).round().clamp(-32768.0, 32767.0) as i16)
                     .collect();

rust/crates/proxy-engine/src/tts.rs

@@ -1,15 +1,49 @@
 //! Text-to-speech engine — synthesizes text to WAV files using Kokoro neural TTS.
 //!
 //! The model is loaded lazily on first use. If the model/voices files are not
-//! present, the generate command returns an error and the TS side falls back
-//! to espeak-ng.
+//! present, the generate command returns an error and the caller skips the prompt.
+//!
+//! Caching is handled internally via a `.meta` sidecar file next to each WAV.
+//! When `cacheable` is true, the engine checks whether the existing WAV was
+//! generated from the same text+voice; if so it returns immediately (cache hit).
+//! Callers never need to check for cached files — that is entirely this module's
+//! responsibility.
 
+use crate::audio_player::pcm_to_mix_frames;
 use kokoro_tts::{KokoroTts, Voice};
 use std::path::Path;
+use std::sync::Arc;
+use tokio::sync::{mpsc, watch};
+
+pub const DEFAULT_MODEL_PATH: &str = ".nogit/tts/kokoro-v1.0.onnx";
+pub const DEFAULT_VOICES_PATH: &str = ".nogit/tts/voices.bin";
+const TTS_OUTPUT_RATE: u32 = 24000;
+const MAX_CHUNK_CHARS: usize = 220;
+const MIN_CHUNK_CHARS: usize = 80;
+
+pub enum TtsStreamMessage {
+    Frames(Vec<Vec<f32>>),
+    Finished,
+    Failed(String),
+}
+
+pub struct TtsLivePrompt {
+    pub initial_frames: Vec<Vec<f32>>,
+    pub stream_rx: mpsc::Receiver<TtsStreamMessage>,
+    pub cancel_tx: watch::Sender<bool>,
+}
+
+#[derive(Clone)]
+pub struct TtsPromptRequest {
+    pub model_path: String,
+    pub voices_path: String,
+    pub voice_name: String,
+    pub text: String,
+}
 
 /// Wraps the Kokoro TTS engine with lazy model loading.
 pub struct TtsEngine {
-    tts: Option<KokoroTts>,
+    tts: Option<Arc<KokoroTts>>,
     /// Path that was used to load the current model (for cache invalidation).
     loaded_model_path: String,
     loaded_voices_path: String,
@@ -24,6 +58,69 @@ impl TtsEngine {
         }
     }
 
+    async fn ensure_loaded(
+        &mut self,
+        model_path: &str,
+        voices_path: &str,
+    ) -> Result<Arc<KokoroTts>, String> {
+        if !Path::new(model_path).exists() {
+            return Err(format!("model not found: {model_path}"));
+        }
+        if !Path::new(voices_path).exists() {
+            return Err(format!("voices not found: {voices_path}"));
+        }
+
+        if self.tts.is_none()
+            || self.loaded_model_path != model_path
+            || self.loaded_voices_path != voices_path
+        {
+            eprintln!("[tts] loading model: {model_path}");
+            let tts = Arc::new(
+                KokoroTts::new(model_path, voices_path)
+                    .await
+                    .map_err(|e| format!("model load failed: {e:?}"))?,
+            );
+            self.tts = Some(tts);
+            self.loaded_model_path = model_path.to_string();
+            self.loaded_voices_path = voices_path.to_string();
+        }
+
+        Ok(self.tts.as_ref().unwrap().clone())
+    }
+
+    pub async fn start_live_prompt(
+        &mut self,
+        request: TtsPromptRequest,
+    ) -> Result<TtsLivePrompt, String> {
+        if request.text.trim().is_empty() {
+            return Err("empty text".into());
+        }
+
+        let tts = self
+            .ensure_loaded(&request.model_path, &request.voices_path)
+            .await?;
+        let voice = select_voice(&request.voice_name);
+        let chunks = chunk_text(&request.text);
+        if chunks.is_empty() {
+            return Err("empty text".into());
+        }
+
+        let initial_frames = synth_text_to_mix_frames(&tts, chunks[0].as_str(), voice).await?;
+        let remaining_chunks: Vec<String> = chunks.into_iter().skip(1).collect();
+        let (stream_tx, stream_rx) = mpsc::channel(8);
+        let (cancel_tx, cancel_rx) = watch::channel(false);
+
+        tokio::spawn(async move {
+            stream_live_prompt_chunks(tts, voice, remaining_chunks, stream_tx, cancel_rx).await;
+        });
+
+        Ok(TtsLivePrompt {
+            initial_frames,
+            stream_rx,
+            cancel_tx,
+        })
+    }
+
     /// Generate a WAV file from text.
     ///
     /// Params (from IPC JSON):
@@ -32,52 +129,64 @@ impl TtsEngine {
     ///   - `voice`: voice name (e.g. "af_bella")
     ///   - `text`: text to synthesize
     ///   - `output`: output WAV file path
-    pub async fn generate(&mut self, params: &serde_json::Value) -> Result<serde_json::Value, String> {
-        let model_path = params.get("model").and_then(|v| v.as_str())
+    ///   - `cacheable`: if true, skip synthesis when the output WAV already
+    ///     matches the same text+voice (checked via a `.meta` sidecar file)
+    pub async fn generate(
+        &mut self,
+        params: &serde_json::Value,
+    ) -> Result<serde_json::Value, String> {
+        let model_path = params
+            .get("model")
+            .and_then(|v| v.as_str())
             .ok_or("missing 'model' param")?;
-        let voices_path = params.get("voices").and_then(|v| v.as_str())
+        let voices_path = params
+            .get("voices")
+            .and_then(|v| v.as_str())
             .ok_or("missing 'voices' param")?;
-        let voice_name = params.get("voice").and_then(|v| v.as_str())
+        let voice_name = params
+            .get("voice")
+            .and_then(|v| v.as_str())
             .unwrap_or("af_bella");
-        let text = params.get("text").and_then(|v| v.as_str())
+        let text = params
+            .get("text")
+            .and_then(|v| v.as_str())
             .ok_or("missing 'text' param")?;
-        let output_path = params.get("output").and_then(|v| v.as_str())
+        let output_path = params
+            .get("output")
+            .and_then(|v| v.as_str())
             .ok_or("missing 'output' param")?;
+        let cacheable = params
+            .get("cacheable")
+            .and_then(|v| v.as_bool())
+            .unwrap_or(false);
 
         if text.is_empty() {
             return Err("empty text".into());
         }
 
-        // Check that model/voices files exist.
-        if !Path::new(model_path).exists() {
-            return Err(format!("model not found: {model_path}"));
-        }
-        if !Path::new(voices_path).exists() {
-            return Err(format!("voices not found: {voices_path}"));
+        // Cache check: if cacheable and the sidecar matches, return immediately.
+        if cacheable && self.is_cache_hit(output_path, text, voice_name) {
+            eprintln!("[tts] cache hit: {output_path}");
+            return Ok(serde_json::json!({ "output": output_path }));
         }
 
-        // Lazy-load or reload if paths changed.
-        if self.tts.is_none()
-            || self.loaded_model_path != model_path
-            || self.loaded_voices_path != voices_path
-        {
-            eprintln!("[tts] loading model: {model_path}");
-            let tts = KokoroTts::new(model_path, voices_path)
-                .await
-                .map_err(|e| format!("model load failed: {e:?}"))?;
-            self.tts = Some(tts);
-            self.loaded_model_path = model_path.to_string();
-            self.loaded_voices_path = voices_path.to_string();
+        // Ensure parent directory exists.
+        if let Some(parent) = Path::new(output_path).parent() {
+            let _ = std::fs::create_dir_all(parent);
         }
 
-        let tts = self.tts.as_ref().unwrap();
+        let tts = self.ensure_loaded(model_path, voices_path).await?;
         let voice = select_voice(voice_name);
 
-        eprintln!("[tts] synthesizing voice '{voice_name}': \"{text}\"");
-        let (samples, duration) = tts.synth(text, voice)
+        eprintln!("[tts] synthesizing WAV voice '{voice_name}' to {output_path}");
+        let (samples, duration) = tts
+            .synth(text, voice)
             .await
             .map_err(|e| format!("synthesis failed: {e:?}"))?;
-        eprintln!("[tts] synthesized {} samples in {duration:?}", samples.len());
+        eprintln!(
+            "[tts] synthesized {} samples in {duration:?}",
+            samples.len()
+        );
 
         // Write 24kHz 16-bit mono WAV.
         let spec = hound::WavSpec {
@@ -91,13 +200,149 @@ impl TtsEngine {
             .map_err(|e| format!("WAV create failed: {e}"))?;
         for &sample in &samples {
             let s16 = (sample * 32767.0).round().clamp(-32768.0, 32767.0) as i16;
-            writer.write_sample(s16).map_err(|e| format!("WAV write: {e}"))?;
+            writer
+                .write_sample(s16)
+                .map_err(|e| format!("WAV write: {e}"))?;
+        }
+        writer
+            .finalize()
+            .map_err(|e| format!("WAV finalize: {e}"))?;
+
+        // Write sidecar for future cache checks.
+        if cacheable {
+            self.write_cache_meta(output_path, text, voice_name);
         }
-        writer.finalize().map_err(|e| format!("WAV finalize: {e}"))?;
 
         eprintln!("[tts] wrote {output_path}");
         Ok(serde_json::json!({ "output": output_path }))
     }
+
+    // -----------------------------------------------------------------------
+    // Cache helpers
+    // -----------------------------------------------------------------------
+
+    /// Check if the WAV + sidecar on disk match the given text+voice.
+    fn is_cache_hit(&self, output_path: &str, text: &str, voice: &str) -> bool {
+        let meta_path = format!("{output_path}.meta");
+        if !Path::new(output_path).exists() || !Path::new(&meta_path).exists() {
+            return false;
+        }
+        match std::fs::read_to_string(&meta_path) {
+            Ok(contents) => contents == Self::cache_key(text, voice),
+            Err(_) => false,
+        }
+    }
+
+    /// Write the sidecar `.meta` file next to the WAV.
+    fn write_cache_meta(&self, output_path: &str, text: &str, voice: &str) {
+        let meta_path = format!("{output_path}.meta");
+        let _ = std::fs::write(&meta_path, Self::cache_key(text, voice));
+    }
+
+    /// Build the cache key from text + voice.
+    fn cache_key(text: &str, voice: &str) -> String {
+        format!("{}\0{}", text, voice)
+    }
+}
+
+async fn synth_text_to_mix_frames(
+    tts: &Arc<KokoroTts>,
+    text: &str,
+    voice: Voice,
+) -> Result<Vec<Vec<f32>>, String> {
+    let (samples, duration) = tts
+        .synth(text, voice)
+        .await
+        .map_err(|e| format!("synthesis failed: {e:?}"))?;
+    eprintln!(
+        "[tts] synthesized chunk ({} chars, {} samples) in {duration:?}",
+        text.chars().count(),
+        samples.len()
+    );
+    pcm_to_mix_frames(&samples, TTS_OUTPUT_RATE)
+}
+
+async fn stream_live_prompt_chunks(
+    tts: Arc<KokoroTts>,
+    voice: Voice,
+    chunks: Vec<String>,
+    stream_tx: mpsc::Sender<TtsStreamMessage>,
+    mut cancel_rx: watch::Receiver<bool>,
+) {
+    for chunk in chunks {
+        if *cancel_rx.borrow() {
+            break;
+        }
+
+        match synth_text_to_mix_frames(&tts, &chunk, voice).await {
+            Ok(frames) => {
+                if *cancel_rx.borrow() {
+                    break;
+                }
+                if stream_tx.send(TtsStreamMessage::Frames(frames)).await.is_err() {
+                    return;
+                }
+            }
+            Err(error) => {
+                let _ = stream_tx.send(TtsStreamMessage::Failed(error)).await;
+                return;
+            }
+        }
+
+        if cancel_rx.has_changed().unwrap_or(false) && *cancel_rx.borrow_and_update() {
+            break;
+        }
+    }
+
+    let _ = stream_tx.send(TtsStreamMessage::Finished).await;
+}
+
+fn chunk_text(text: &str) -> Vec<String> {
+    let mut chunks = Vec::new();
+    let mut current = String::new();
+
+    for ch in text.chars() {
+        current.push(ch);
+
+        let len = current.chars().count();
+        let hard_split = len >= MAX_CHUNK_CHARS && (ch.is_whitespace() || is_soft_boundary(ch));
+        let natural_split = len >= MIN_CHUNK_CHARS && is_sentence_boundary(ch);
+
+        if natural_split || hard_split {
+            push_chunk(&mut chunks, &mut current);
+        }
+    }
+
+    push_chunk(&mut chunks, &mut current);
+
+    if chunks.len() >= 2 {
+        let last_len = chunks.last().unwrap().chars().count();
+        if last_len < (MIN_CHUNK_CHARS / 2) {
+            let tail = chunks.pop().unwrap();
+            if let Some(prev) = chunks.last_mut() {
+                prev.push(' ');
+                prev.push_str(tail.trim());
+            }
+        }
+    }
+
+    chunks
+}
+
+fn push_chunk(chunks: &mut Vec<String>, current: &mut String) {
+    let trimmed = current.trim();
+    if !trimmed.is_empty() {
+        chunks.push(trimmed.to_string());
+    }
+    current.clear();
+}
+
+fn is_sentence_boundary(ch: char) -> bool {
+    matches!(ch, '.' | '!' | '?' | '\n' | ';' | ':')
+}
+
+fn is_soft_boundary(ch: char) -> bool {
+    matches!(ch, ',' | ';' | ':' | ')' | ']' | '\n')
 }
 
 /// Map voice name string to Kokoro Voice enum variant.

rust/crates/proxy-engine/src/voicemail.rs

@@ -128,8 +128,8 @@ async fn record_from_socket(
                     break; // Max duration reached.
                 }
             }
-            Ok(Err(_)) => break,     // Socket error (closed).
-            Err(_) => break,          // Timeout (max duration + grace).
+            Ok(Err(_)) => break, // Socket error (closed).
+            Err(_) => break,     // Timeout (max duration + grace).
         }
     }
 

rust/crates/proxy-engine/src/webrtc_engine.rs

@@ -58,9 +58,7 @@ impl WebRtcEngine {
             .register_default_codecs()
             .map_err(|e| format!("register codecs: {e}"))?;
 
-        let api = APIBuilder::new()
-            .with_media_engine(media_engine)
-            .build();
+        let api = APIBuilder::new().with_media_engine(media_engine).build();
 
         let config = RTCConfiguration {
             ice_servers: vec![],
@@ -91,8 +89,7 @@ impl WebRtcEngine {
             .map_err(|e| format!("add track: {e}"))?;
 
         // Shared mixer channel sender (populated when linked to a call).
-        let mixer_tx: Arc<Mutex<Option<mpsc::Sender<RtpPacket>>>> =
-            Arc::new(Mutex::new(None));
+        let mixer_tx: Arc<Mutex<Option<mpsc::Sender<RtpPacket>>>> = Arc::new(Mutex::new(None));
 
         // ICE candidate handler.
         let out_tx_ice = self.out_tx.clone();
@@ -256,7 +253,11 @@ impl WebRtcEngine {
 
     pub async fn close_session(&mut self, session_id: &str) -> Result<(), String> {
         if let Some(session) = self.sessions.remove(session_id) {
-            session.pc.close().await.map_err(|e| format!("close: {e}"))?;
+            session
+                .pc
+                .close()
+                .await
+                .map_err(|e| format!("close: {e}"))?;
         }
         Ok(())
     }

rust/crates/sip-proto/src/dialog.rs

@@ -51,9 +51,7 @@ impl SipDialog {
                 .map(|s| s.to_string())
                 .unwrap_or_else(generate_tag),
             remote_tag: None,
-            local_uri: SipMessage::extract_uri(from)
-                .unwrap_or("")
-                .to_string(),
+            local_uri: SipMessage::extract_uri(from).unwrap_or("").to_string(),
             remote_uri: SipMessage::extract_uri(to).unwrap_or("").to_string(),
             local_cseq,
             remote_cseq: 0,
@@ -181,10 +179,7 @@ impl SipDialog {
                 format!("<{}>{remote_tag_str}", self.remote_uri),
             ),
             ("Call-ID".to_string(), self.call_id.clone()),
-            (
-                "CSeq".to_string(),
-                format!("{} {method}", self.local_cseq),
-            ),
+            ("CSeq".to_string(), format!("{} {method}", self.local_cseq)),
             ("Max-Forwards".to_string(), "70".to_string()),
         ];
 
@@ -243,10 +238,7 @@ impl SipDialog {
                 format!("<{}>{remote_tag_str}", self.remote_uri),
             ),
             ("Call-ID".to_string(), self.call_id.clone()),
-            (
-                "CSeq".to_string(),
-                format!("{} ACK", self.local_cseq),
-            ),
+            ("CSeq".to_string(), format!("{} ACK", self.local_cseq)),
             ("Max-Forwards".to_string(), "70".to_string()),
         ];
 
@@ -271,10 +263,7 @@ impl SipDialog {
             ("From".to_string(), from),
             ("To".to_string(), to),
             ("Call-ID".to_string(), self.call_id.clone()),
-            (
-                "CSeq".to_string(),
-                format!("{} CANCEL", self.local_cseq),
-            ),
+            ("CSeq".to_string(), format!("{} CANCEL", self.local_cseq)),
             ("Max-Forwards".to_string(), "70".to_string()),
             ("Content-Length".to_string(), "0".to_string()),
         ];
@@ -284,11 +273,7 @@ impl SipDialog {
             .unwrap_or(&self.remote_target)
             .to_string();
 
-        SipMessage::new(
-            format!("CANCEL {ruri} SIP/2.0"),
-            headers,
-            String::new(),
-        )
+        SipMessage::new(format!("CANCEL {ruri} SIP/2.0"), headers, String::new())
     }
 
     /// Transition the dialog to terminated state.

rust/crates/sip-proto/src/helpers.rs

@@ -27,7 +27,9 @@ pub fn generate_branch() -> String {
 
 fn random_hex(bytes: usize) -> String {
     let mut rng = rand::thread_rng();
-    (0..bytes).map(|_| format!("{:02x}", rng.gen::<u8>())).collect()
+    (0..bytes)
+        .map(|_| format!("{:02x}", rng.gen::<u8>()))
+        .collect()
 }
 
 // ---- Codec registry --------------------------------------------------------
@@ -142,7 +144,9 @@ pub fn parse_digest_challenge(header: &str) -> Option<DigestChallenge> {
                 return Some(after[1..1 + end].to_string());
             }
             // Unquoted value.
-            let end = after.find(|c: char| c == ',' || c.is_whitespace()).unwrap_or(after.len());
+            let end = after
+                .find(|c: char| c == ',' || c.is_whitespace())
+                .unwrap_or(after.len());
             return Some(after[..end].to_string());
         }
         None
@@ -241,11 +245,7 @@ pub struct MwiResult {
     pub extra_headers: Vec<(String, String)>,
 }
 
-pub fn build_mwi_body(
-    new_messages: u32,
-    old_messages: u32,
-    account_uri: &str,
-) -> MwiResult {
+pub fn build_mwi_body(new_messages: u32, old_messages: u32, account_uri: &str) -> MwiResult {
     let waiting = if new_messages > 0 { "yes" } else { "no" };
     let body = format!(
         "Messages-Waiting: {waiting}\r\n\

rust/crates/sip-proto/src/lib.rs

@@ -4,9 +4,9 @@
 //! SDP handling, Digest authentication, and URI rewriting.
 //! Ported from the TypeScript `ts/sip/` library.
 
-pub mod message;
 pub mod dialog;
 pub mod helpers;
+pub mod message;
 pub mod rewrite;
 
 /// Network endpoint (address + port + optional negotiated codec).

rust/crates/sip-proto/src/message.rs

@@ -14,7 +14,11 @@ pub struct SipMessage {
 
 impl SipMessage {
     pub fn new(start_line: String, headers: Vec<(String, String)>, body: String) -> Self {
-        Self { start_line, headers, body }
+        Self {
+            start_line,
+            headers,
+            body,
+        }
     }
 
     // ---- Parsing -----------------------------------------------------------
@@ -175,7 +179,8 @@ impl SipMessage {
 
     /// Inserts a header at the top of the header list.
     pub fn prepend_header(&mut self, name: &str, value: &str) -> &mut Self {
-        self.headers.insert(0, (name.to_string(), value.to_string()));
+        self.headers
+            .insert(0, (name.to_string(), value.to_string()));
         self
     }
 
@@ -233,10 +238,7 @@ impl SipMessage {
             .to_display_name
             .map(|d| format!("\"{d}\" "))
             .unwrap_or_default();
-        let to_tag_str = opts
-            .to_tag
-            .map(|t| format!(";tag={t}"))
-            .unwrap_or_default();
+        let to_tag_str = opts.to_tag.map(|t| format!(";tag={t}")).unwrap_or_default();
 
         let mut headers = vec![
             (
@@ -364,7 +366,43 @@ impl SipMessage {
                 .find(|c: char| c == ';' || c == '>')
                 .unwrap_or(trimmed.len());
             let result = &trimmed[..end];
-            if result.is_empty() { None } else { Some(result) }
+            if result.is_empty() {
+                None
+            } else {
+                Some(result)
+            }
+        }
+    }
+
+    /// Extract the user part from a SIP/TEL URI or header value.
+    pub fn extract_uri_user(uri_or_header_value: &str) -> Option<&str> {
+        let raw = Self::extract_uri(uri_or_header_value).unwrap_or(uri_or_header_value);
+        let raw = raw.trim();
+        if raw.is_empty() {
+            return None;
+        }
+
+        let user_part = if raw
+            .get(..5)
+            .is_some_and(|prefix| prefix.eq_ignore_ascii_case("sips:"))
+        {
+            &raw[5..]
+        } else if raw.get(..4).is_some_and(|prefix| {
+            prefix.eq_ignore_ascii_case("sip:") || prefix.eq_ignore_ascii_case("tel:")
+        }) {
+            &raw[4..]
+        } else {
+            raw
+        };
+
+        let end = user_part
+            .find(|c: char| matches!(c, '@' | ';' | '?' | '>'))
+            .unwrap_or(user_part.len());
+        let result = &user_part[..end];
+        if result.is_empty() {
+            None
+        } else {
+            Some(result)
         }
     }
 }
@@ -506,6 +544,19 @@ mod tests {
             SipMessage::extract_uri("\"Name\" <sip:user@host>;tag=abc"),
             Some("sip:user@host")
         );
+        assert_eq!(
+            SipMessage::extract_uri_user("\"Name\" <sip:+49 421 219694@host>;tag=abc"),
+            Some("+49 421 219694")
+        );
+        assert_eq!(
+            SipMessage::extract_uri_user("sip:0049421219694@voip.easybell.de"),
+            Some("0049421219694")
+        );
+        assert_eq!(
+            SipMessage::extract_uri_user("tel:+49421219694;phone-context=example.com"),
+            Some("+49421219694")
+        );
+        assert_eq!(SipMessage::extract_uri_user("SIP:user@host"), Some("user"));
     }
 
     #[test]
@@ -535,7 +586,10 @@ mod tests {
         );
         assert_eq!(invite.method(), Some("INVITE"));
         assert_eq!(invite.call_id(), "test-123");
-        assert!(invite.get_header("Via").unwrap().contains("192.168.1.1:5070"));
+        assert!(invite
+            .get_header("Via")
+            .unwrap()
+            .contains("192.168.1.1:5070"));
 
         let response = SipMessage::create_response(
             200,

rust/crates/sip-proto/src/rewrite.rs

@@ -92,7 +92,11 @@ pub fn rewrite_sdp(body: &str, ip: &str, port: u16) -> (String, Option<Endpoint>
         .collect();
 
     let original = match (orig_addr, orig_port) {
-        (Some(a), Some(p)) => Some(Endpoint { address: a, port: p, codec_pt: None }),
+        (Some(a), Some(p)) => Some(Endpoint {
+            address: a,
+            port: p,
+            codec_pt: None,
+        }),
         _ => None,
     };
 

ts/00_commitinfo_data.ts

@@ -3,6 +3,6 @@
  */
 export const commitinfo = {
   name: 'siprouter',
-  version: '1.19.2',
+  version: '1.25.0',
   description: 'undefined'
 }

ts/announcement.ts

@@ -1,137 +0,0 @@
-/**
- * TTS announcement module — generates announcement WAV files at startup.
- *
- * Engine priority: espeak-ng (formant TTS, fast) → Kokoro neural TTS via
- * proxy-engine → disabled.
- *
- * The generated WAV is left on disk for Rust's audio_player / start_interaction
- * to play during calls. No encoding or RTP playback happens in TypeScript.
- */
-
-import { execSync } from 'node:child_process';
-import fs from 'node:fs';
-import path from 'node:path';
-import { sendProxyCommand, isProxyReady } from './proxybridge.ts';
-
-// ---------------------------------------------------------------------------
-// State
-// ---------------------------------------------------------------------------
-
-const TTS_DIR = path.join(process.cwd(), '.nogit', 'tts');
-const ANNOUNCEMENT_TEXT = "Hello. I'm connecting your call now.";
-const CACHE_WAV = path.join(TTS_DIR, 'announcement.wav');
-
-// Kokoro fallback constants.
-const KOKORO_MODEL = 'kokoro-v1.0.onnx';
-const KOKORO_VOICES = 'voices.bin';
-const KOKORO_VOICE = 'af_bella';
-
-// ---------------------------------------------------------------------------
-// TTS generators
-// ---------------------------------------------------------------------------
-
-/** Check if espeak-ng is available on the system. */
-function isEspeakAvailable(): boolean {
-  try {
-    execSync('which espeak-ng', { stdio: 'pipe' });
-    return true;
-  } catch {
-    return false;
-  }
-}
-
-/** Generate announcement WAV via espeak-ng (primary engine). */
-function generateViaEspeak(wavPath: string, text: string, log: (msg: string) => void): boolean {
-  log('[tts] generating announcement audio via espeak-ng...');
-  try {
-    execSync(
-      `espeak-ng -v en-us -s 150 -w "${wavPath}" "${text}"`,
-      { timeout: 10000, stdio: 'pipe' },
-    );
-    log('[tts] espeak-ng WAV generated');
-    return true;
-  } catch (e: any) {
-    log(`[tts] espeak-ng failed: ${e.message}`);
-    return false;
-  }
-}
-
-/** Generate announcement WAV via Kokoro TTS (fallback, runs inside proxy-engine). */
-async function generateViaKokoro(wavPath: string, text: string, log: (msg: string) => void): Promise<boolean> {
-  const modelPath = path.join(TTS_DIR, KOKORO_MODEL);
-  const voicesPath = path.join(TTS_DIR, KOKORO_VOICES);
-
-  if (!fs.existsSync(modelPath) || !fs.existsSync(voicesPath)) {
-    log('[tts] Kokoro model/voices not found — Kokoro fallback unavailable');
-    return false;
-  }
-
-  if (!isProxyReady()) {
-    log('[tts] proxy-engine not ready — Kokoro fallback unavailable');
-    return false;
-  }
-
-  log('[tts] generating announcement audio via Kokoro TTS (fallback)...');
-  try {
-    await sendProxyCommand('generate_tts', {
-      model: modelPath,
-      voices: voicesPath,
-      voice: KOKORO_VOICE,
-      text,
-      output: wavPath,
-    });
-    log('[tts] Kokoro WAV generated (via proxy-engine)');
-    return true;
-  } catch (e: any) {
-    log(`[tts] Kokoro failed: ${e.message}`);
-    return false;
-  }
-}
-
-// ---------------------------------------------------------------------------
-// Initialization
-// ---------------------------------------------------------------------------
-
-/**
- * Pre-generate the announcement WAV file.
- * Must be called after the proxy engine is initialized.
- *
- * Engine priority: espeak-ng → Kokoro → disabled.
- */
-export async function initAnnouncement(log: (msg: string) => void): Promise<boolean> {
-  fs.mkdirSync(TTS_DIR, { recursive: true });
-
-  try {
-    if (!fs.existsSync(CACHE_WAV)) {
-      let generated = false;
-
-      // Try espeak-ng first.
-      if (isEspeakAvailable()) {
-        generated = generateViaEspeak(CACHE_WAV, ANNOUNCEMENT_TEXT, log);
-      } else {
-        log('[tts] espeak-ng not installed — trying Kokoro fallback');
-      }
-
-      // Fall back to Kokoro (via proxy-engine).
-      if (!generated) {
-        generated = await generateViaKokoro(CACHE_WAV, ANNOUNCEMENT_TEXT, log);
-      }
-
-      if (!generated) {
-        log('[tts] no TTS engine available — announcements disabled');
-        return false;
-      }
-    }
-
-    log('[tts] announcement WAV ready');
-    return true;
-  } catch (e: any) {
-    log(`[tts] init error: ${e.message}`);
-    return false;
-  }
-}
-
-/** Get the path to the cached announcement WAV, or null if not generated. */
-export function getAnnouncementWavPath(): string | null {
-  return fs.existsSync(CACHE_WAV) ? CACHE_WAV : null;
-}

ts/call/prompt-cache.ts

@@ -1,275 +0,0 @@
-/**
- * PromptCache — manages named audio prompt WAV files for IVR and voicemail.
- *
- * Generates WAV files via espeak-ng (primary) or Kokoro TTS through the
- * proxy-engine (fallback). Also supports loading pre-existing WAV files
- * and programmatic tone generation.
- *
- * All audio playback happens in Rust (audio_player / start_interaction).
- * This module only manages WAV files on disk.
- */
-
-import { execSync } from 'node:child_process';
-import fs from 'node:fs';
-import path from 'node:path';
-import { Buffer } from 'node:buffer';
-import { sendProxyCommand, isProxyReady } from '../proxybridge.ts';
-
-// ---------------------------------------------------------------------------
-// Types
-// ---------------------------------------------------------------------------
-
-/** A cached prompt — just a WAV file path and metadata. */
-export interface ICachedPrompt {
-  /** Unique prompt identifier. */
-  id: string;
-  /** Path to the WAV file on disk. */
-  wavPath: string;
-  /** Total duration in milliseconds (approximate, from WAV header). */
-  durationMs: number;
-}
-
-// ---------------------------------------------------------------------------
-// TTS helpers
-// ---------------------------------------------------------------------------
-
-const TTS_DIR = path.join(process.cwd(), '.nogit', 'tts');
-
-/** Check if espeak-ng is available. */
-function isEspeakAvailable(): boolean {
-  try {
-    execSync('which espeak-ng', { stdio: 'pipe' });
-    return true;
-  } catch {
-    return false;
-  }
-}
-
-/** Generate WAV via espeak-ng. */
-function generateViaEspeak(wavPath: string, text: string): boolean {
-  try {
-    execSync(
-      `espeak-ng -v en-us -s 150 -w "${wavPath}" "${text}"`,
-      { timeout: 10000, stdio: 'pipe' },
-    );
-    return true;
-  } catch {
-    return false;
-  }
-}
-
-/** Generate WAV via Kokoro TTS (runs inside proxy-engine). */
-async function generateViaKokoro(wavPath: string, text: string, voice: string): Promise<boolean> {
-  const modelPath = path.join(TTS_DIR, 'kokoro-v1.0.onnx');
-  const voicesPath = path.join(TTS_DIR, 'voices.bin');
-  if (!fs.existsSync(modelPath) || !fs.existsSync(voicesPath)) return false;
-  if (!isProxyReady()) return false;
-
-  try {
-    await sendProxyCommand('generate_tts', {
-      model: modelPath,
-      voices: voicesPath,
-      voice,
-      text,
-      output: wavPath,
-    });
-    return true;
-  } catch {
-    return false;
-  }
-}
-
-/** Read a WAV file's duration from its header. */
-function getWavDurationMs(wavPath: string): number {
-  try {
-    const wav = fs.readFileSync(wavPath);
-    if (wav.length < 44) return 0;
-    if (wav.toString('ascii', 0, 4) !== 'RIFF') return 0;
-
-    let sampleRate = 16000;
-    let dataSize = 0;
-    let bitsPerSample = 16;
-    let channels = 1;
-    let offset = 12;
-
-    while (offset < wav.length - 8) {
-      const chunkId = wav.toString('ascii', offset, offset + 4);
-      const chunkSize = wav.readUInt32LE(offset + 4);
-      if (chunkId === 'fmt ') {
-        channels = wav.readUInt16LE(offset + 10);
-        sampleRate = wav.readUInt32LE(offset + 12);
-        bitsPerSample = wav.readUInt16LE(offset + 22);
-      }
-      if (chunkId === 'data') {
-        dataSize = chunkSize;
-      }
-      offset += 8 + chunkSize;
-      if (offset % 2 !== 0) offset++;
-    }
-
-    const bytesPerSample = (bitsPerSample / 8) * channels;
-    const totalSamples = bytesPerSample > 0 ? dataSize / bytesPerSample : 0;
-    return sampleRate > 0 ? Math.round((totalSamples / sampleRate) * 1000) : 0;
-  } catch {
-    return 0;
-  }
-}
-
-// ---------------------------------------------------------------------------
-// PromptCache
-// ---------------------------------------------------------------------------
-
-export class PromptCache {
-  private prompts = new Map<string, ICachedPrompt>();
-  private log: (msg: string) => void;
-  private espeakAvailable: boolean | null = null;
-
-  constructor(log: (msg: string) => void) {
-    this.log = log;
-  }
-
-  // -------------------------------------------------------------------------
-  // Public API
-  // -------------------------------------------------------------------------
-
-  /** Get a cached prompt by ID. */
-  get(id: string): ICachedPrompt | null {
-    return this.prompts.get(id) ?? null;
-  }
-
-  /** Check if a prompt is cached. */
-  has(id: string): boolean {
-    return this.prompts.has(id);
-  }
-
-  /** List all cached prompt IDs. */
-  listIds(): string[] {
-    return [...this.prompts.keys()];
-  }
-
-  /**
-   * Generate a TTS prompt WAV and cache its path.
-   * Uses espeak-ng (primary) or Kokoro (fallback).
-   */
-  async generatePrompt(id: string, text: string, voice = 'af_bella'): Promise<ICachedPrompt | null> {
-    fs.mkdirSync(TTS_DIR, { recursive: true });
-    const wavPath = path.join(TTS_DIR, `prompt-${id}.wav`);
-
-    // Check espeak availability once.
-    if (this.espeakAvailable === null) {
-      this.espeakAvailable = isEspeakAvailable();
-    }
-
-    // Generate WAV if not already on disk.
-    if (!fs.existsSync(wavPath)) {
-      let generated = false;
-      if (this.espeakAvailable) {
-        generated = generateViaEspeak(wavPath, text);
-      }
-      if (!generated) {
-        generated = await generateViaKokoro(wavPath, text, voice);
-      }
-      if (!generated) {
-        this.log(`[prompt-cache] failed to generate TTS for "${id}"`);
-        return null;
-      }
-      this.log(`[prompt-cache] generated WAV for "${id}"`);
-    }
-
-    return this.registerWav(id, wavPath);
-  }
-
-  /**
-   * Load a pre-existing WAV file as a prompt.
-   */
-  async loadWavPrompt(id: string, wavPath: string): Promise<ICachedPrompt | null> {
-    if (!fs.existsSync(wavPath)) {
-      this.log(`[prompt-cache] WAV not found: ${wavPath}`);
-      return null;
-    }
-    return this.registerWav(id, wavPath);
-  }
-
-  /**
-   * Generate a beep tone WAV and cache it.
-   */
-  async generateBeep(
-    id: string,
-    freqHz = 1000,
-    durationMs = 500,
-    amplitude = 8000,
-  ): Promise<ICachedPrompt | null> {
-    fs.mkdirSync(TTS_DIR, { recursive: true });
-    const wavPath = path.join(TTS_DIR, `prompt-${id}.wav`);
-
-    if (!fs.existsSync(wavPath)) {
-      // Generate 16kHz 16-bit mono sine wave WAV.
-      const sampleRate = 16000;
-      const totalSamples = Math.floor((sampleRate * durationMs) / 1000);
-      const pcm = Buffer.alloc(totalSamples * 2);
-
-      for (let i = 0; i < totalSamples; i++) {
-        const t = i / sampleRate;
-        const fadeLen = Math.floor(sampleRate * 0.01); // 10ms fade
-        let envelope = 1.0;
-        if (i < fadeLen) envelope = i / fadeLen;
-        else if (i > totalSamples - fadeLen) envelope = (totalSamples - i) / fadeLen;
-
-        const sample = Math.round(Math.sin(2 * Math.PI * freqHz * t) * amplitude * envelope);
-        pcm.writeInt16LE(Math.max(-32768, Math.min(32767, sample)), i * 2);
-      }
-
-      // Write WAV file.
-      const headerSize = 44;
-      const dataSize = pcm.length;
-      const wav = Buffer.alloc(headerSize + dataSize);
-
-      // RIFF header
-      wav.write('RIFF', 0);
-      wav.writeUInt32LE(36 + dataSize, 4);
-      wav.write('WAVE', 8);
-
-      // fmt chunk
-      wav.write('fmt ', 12);
-      wav.writeUInt32LE(16, 16);        // chunk size
-      wav.writeUInt16LE(1, 20);         // PCM format
-      wav.writeUInt16LE(1, 22);         // mono
-      wav.writeUInt32LE(sampleRate, 24);
-      wav.writeUInt32LE(sampleRate * 2, 28); // byte rate
-      wav.writeUInt16LE(2, 32);         // block align
-      wav.writeUInt16LE(16, 34);        // bits per sample
-
-      // data chunk
-      wav.write('data', 36);
-      wav.writeUInt32LE(dataSize, 40);
-      pcm.copy(wav, 44);
-
-      fs.writeFileSync(wavPath, wav);
-      this.log(`[prompt-cache] beep WAV generated for "${id}"`);
-    }
-
-    return this.registerWav(id, wavPath);
-  }
-
-  /** Remove a prompt from the cache. */
-  remove(id: string): void {
-    this.prompts.delete(id);
-  }
-
-  /** Clear all cached prompts. */
-  clear(): void {
-    this.prompts.clear();
-  }
-
-  // -------------------------------------------------------------------------
-  // Internal
-  // -------------------------------------------------------------------------
-
-  private registerWav(id: string, wavPath: string): ICachedPrompt {
-    const durationMs = getWavDurationMs(wavPath);
-    const prompt: ICachedPrompt = { id, wavPath, durationMs };
-    this.prompts.set(id, prompt);
-    this.log(`[prompt-cache] cached "${id}": ${wavPath} (${(durationMs / 1000).toFixed(1)}s)`);
-    return prompt;
-  }
-}

ts/config.ts

@@ -8,6 +8,7 @@
 
 import fs from 'node:fs';
 import path from 'node:path';
+import type { IVoiceboxConfig } from './voicebox.js';
 
 // ---------------------------------------------------------------------------
 // Shared types (previously in ts/sip/types.ts, now inlined)
@@ -47,6 +48,24 @@ export interface IDeviceConfig {
   extension: string;
 }
 
+export type TIncomingNumberMode = 'single' | 'range' | 'regex';
+
+export interface IIncomingNumberConfig {
+  id: string;
+  label: string;
+  providerId?: string;
+  mode: TIncomingNumberMode;
+  countryCode?: string;
+  areaCode?: string;
+  localNumber?: string;
+  rangeEnd?: string;
+  pattern?: string;
+
+  // Legacy persisted fields kept for migration compatibility.
+  number?: string;
+  rangeStart?: string;
+}
+
 // ---------------------------------------------------------------------------
 // Match/Action routing model
 // ---------------------------------------------------------------------------
@@ -61,8 +80,11 @@ export interface ISipRouteMatch {
   direction: 'inbound' | 'outbound';
 
   /**
-   * Match the dialed/called number (To/Request-URI for inbound DID, dialed digits for outbound).
-   * Supports: exact string, prefix with trailing '*' (e.g. "+4930*"), or regex ("/^\\+49/").
+   * Match the normalized called number.
+   *
+   * Inbound: matches the provider-delivered DID / Request-URI user part.
+   * Outbound: matches the normalized dialed digits.
+   * Supports: exact string, numeric range `start..end`, prefix with trailing '*' (e.g. "+4930*"), or regex ("/^\\+49/").
    */
   numberPattern?: string;
 
@@ -88,13 +110,13 @@ export interface ISipRouteAction {
 
   // --- Inbound actions (IVR / voicemail) ---
 
-  /** Route directly to a voicemail box (skip ringing devices). */
+  /** Voicemail fallback for matched inbound routes. */
   voicemailBox?: string;
 
   /** Route to an IVR menu by menu ID (skip ringing devices). */
   ivrMenuId?: string;
 
-  /** Override no-answer timeout (seconds) before routing to voicemail. */
+  /** Reserved for future no-answer handling. */
   noAnswerTimeout?: number;
 
   // --- Outbound actions (provider selection) ---
@@ -160,24 +182,13 @@ export interface IContact {
 // Voicebox configuration
 // ---------------------------------------------------------------------------
 
-export interface IVoiceboxConfig {
-  /** Unique ID — typically matches device ID or extension. */
-  id: string;
-  /** Whether this voicebox is active. */
-  enabled: boolean;
-  /** Custom TTS greeting text. */
-  greetingText?: string;
-  /** TTS voice ID (default 'af_bella'). */
-  greetingVoice?: string;
-  /** Path to uploaded WAV greeting (overrides TTS). */
-  greetingWavPath?: string;
-  /** Seconds to wait before routing to voicemail (default 25). */
-  noAnswerTimeoutSec?: number;
-  /** Maximum recording duration in seconds (default 120). */
-  maxRecordingSec?: number;
-  /** Maximum stored messages per box (default 50). */
-  maxMessages?: number;
-}
+// Canonical definition lives in voicebox.ts (imported at the top of this
+// file) — re-exported here so consumers can import everything from a
+// single config module without pulling in the voicebox implementation.
+// This used to be a duplicated interface and caused
+// "number | undefined is not assignable to number" type errors when
+// passing config.voiceboxes into VoiceboxManager.init().
+export type { IVoiceboxConfig };
 
 // ---------------------------------------------------------------------------
 // IVR configuration
@@ -241,6 +252,7 @@ export interface IAppConfig {
   proxy: IProxyConfig;
   providers: IProviderConfig[];
   devices: IDeviceConfig[];
+  incomingNumbers?: IIncomingNumberConfig[];
   routing: IRoutingConfig;
   contacts: IContact[];
   voiceboxes?: IVoiceboxConfig[];
@@ -295,6 +307,14 @@ export function loadConfig(): IAppConfig {
     d.extension ??= '100';
   }
 
+  cfg.incomingNumbers ??= [];
+  for (const incoming of cfg.incomingNumbers) {
+    if (!incoming.id) incoming.id = `incoming-${Date.now()}`;
+    incoming.label ??= incoming.id;
+    incoming.mode ??= incoming.pattern ? 'regex' : incoming.rangeStart || incoming.rangeEnd ? 'range' : 'single';
+    incoming.countryCode ??= incoming.mode === 'regex' ? undefined : '+49';
+  }
+
   cfg.routing ??= { routes: [] };
   cfg.routing.routes ??= [];
 

ts/frontend.ts

@@ -14,12 +14,36 @@ import { WebSocketServer, WebSocket } from 'ws';
 import { handleWebRtcSignaling } from './webrtcbridge.ts';
 import type { VoiceboxManager } from './voicebox.ts';
 
-// CallManager was previously used for WebRTC call handling. Now replaced by Rust proxy-engine.
-// Kept as `any` type for backward compat with the function signature until full WebRTC port.
-type CallManager = any;
-
 const CONFIG_PATH = path.join(process.cwd(), '.nogit', 'config.json');
 
+interface IHandleRequestContext {
+  getStatus: () => unknown;
+  log: (msg: string) => void;
+  onStartCall: (number: string, deviceId?: string, providerId?: string) => { id: string } | null;
+  onHangupCall: (callId: string) => boolean;
+  onConfigSaved?: () => void | Promise<void>;
+  voiceboxManager?: VoiceboxManager;
+}
+
+interface IWebUiOptions extends IHandleRequestContext {
+  port: number;
+  onWebRtcOffer?: (sessionId: string, sdp: string, ws: WebSocket) => Promise<void>;
+  onWebRtcIce?: (sessionId: string, candidate: unknown) => Promise<void>;
+  onWebRtcClose?: (sessionId: string) => Promise<void>;
+  onWebRtcAccept?: (callId: string, sessionId: string) => void;
+}
+
+interface IWebRtcSocketMessage {
+  type?: string;
+  sessionId?: string;
+  callId?: string;
+  sdp?: string;
+  candidate?: unknown;
+  userAgent?: string;
+  _remoteIp?: string | null;
+  [key: string]: unknown;
+}
+
 // ---------------------------------------------------------------------------
 // WebSocket broadcast
 // ---------------------------------------------------------------------------
@@ -82,14 +106,9 @@ function loadStaticFiles(): void {
 async function handleRequest(
   req: http.IncomingMessage,
   res: http.ServerResponse,
-  getStatus: () => unknown,
-  log: (msg: string) => void,
-  onStartCall: (number: string, deviceId?: string, providerId?: string) => { id: string } | null,
-  onHangupCall: (callId: string) => boolean,
-  onConfigSaved?: () => void,
-  callManager?: CallManager,
-  voiceboxManager?: VoiceboxManager,
+  context: IHandleRequestContext,
 ): Promise<void> {
+  const { getStatus, log, onStartCall, onHangupCall, onConfigSaved, voiceboxManager } = context;
   const url = new URL(req.url || '/', `http://${req.headers.host || 'localhost'}`);
   const method = req.method || 'GET';
 
@@ -247,6 +266,7 @@ async function handleRequest(
           if (existing && ud.displayName !== undefined) existing.displayName = ud.displayName;
         }
       }
+      if (updates.incomingNumbers !== undefined) cfg.incomingNumbers = updates.incomingNumbers;
       if (updates.routing) {
         if (updates.routing.routes) {
           cfg.routing.routes = updates.routing.routes;
@@ -258,7 +278,7 @@ async function handleRequest(
 
       fs.writeFileSync(CONFIG_PATH, JSON.stringify(cfg, null, 2) + '\n');
       log('[config] updated config.json');
-      onConfigSaved?.();
+      await onConfigSaved?.();
       return sendJson(res, { ok: true });
     } catch (e: any) {
       return sendJson(res, { ok: false, error: e.message }, 400);
@@ -339,21 +359,21 @@ async function handleRequest(
 // ---------------------------------------------------------------------------
 
 export function initWebUi(
-  getStatus: () => unknown,
-  log: (msg: string) => void,
-  onStartCall: (number: string, deviceId?: string, providerId?: string) => { id: string } | null,
-  onHangupCall: (callId: string) => boolean,
-  onConfigSaved?: () => void,
-  callManager?: CallManager,
-  voiceboxManager?: VoiceboxManager,
-  /** WebRTC signaling handlers — forwarded to Rust proxy-engine. */
-  onWebRtcOffer?: (sessionId: string, sdp: string, ws: WebSocket) => Promise<void>,
-  onWebRtcIce?: (sessionId: string, candidate: any) => Promise<void>,
-  onWebRtcClose?: (sessionId: string) => Promise<void>,
-  /** Called when browser sends webrtc-accept (callId + sessionId linking). */
-  onWebRtcAccept?: (callId: string, sessionId: string) => void,
+  options: IWebUiOptions,
 ): void {
-  const WEB_PORT = 3060;
+  const {
+    port,
+    getStatus,
+    log,
+    onStartCall,
+    onHangupCall,
+    onConfigSaved,
+    voiceboxManager,
+    onWebRtcOffer,
+    onWebRtcIce,
+    onWebRtcClose,
+    onWebRtcAccept,
+  } = options;
 
   loadStaticFiles();
 
@@ -367,12 +387,12 @@ export function initWebUi(
     const cert = fs.readFileSync(certPath, 'utf8');
     const key = fs.readFileSync(keyPath, 'utf8');
     server = https.createServer({ cert, key }, (req, res) =>
-      handleRequest(req, res, getStatus, log, onStartCall, onHangupCall, onConfigSaved, callManager, voiceboxManager).catch(() => { res.writeHead(500); res.end(); }),
+      handleRequest(req, res, { getStatus, log, onStartCall, onHangupCall, onConfigSaved, voiceboxManager }).catch(() => { res.writeHead(500); res.end(); }),
     );
     useTls = true;
   } catch {
     server = http.createServer((req, res) =>
-      handleRequest(req, res, getStatus, log, onStartCall, onHangupCall, onConfigSaved, callManager, voiceboxManager).catch(() => { res.writeHead(500); res.end(); }),
+      handleRequest(req, res, { getStatus, log, onStartCall, onHangupCall, onConfigSaved, voiceboxManager }).catch(() => { res.writeHead(500); res.end(); }),
     );
   }
 
@@ -386,12 +406,12 @@ export function initWebUi(
 
     socket.on('message', (raw) => {
       try {
-        const msg = JSON.parse(raw.toString());
+        const msg = JSON.parse(raw.toString()) as IWebRtcSocketMessage;
         if (msg.type === 'webrtc-offer' && msg.sessionId) {
           // Forward to Rust proxy-engine for WebRTC handling.
-          if (onWebRtcOffer) {
+          if (onWebRtcOffer && typeof msg.sdp === 'string') {
             log(`[webrtc-ws] offer msg keys: ${Object.keys(msg).join(',')}, sdp type: ${typeof msg.sdp}, sdp len: ${msg.sdp?.length || 0}`);
-            onWebRtcOffer(msg.sessionId, msg.sdp, socket as any).catch((e: any) =>
+            onWebRtcOffer(msg.sessionId, msg.sdp, socket).catch((e: any) =>
               log(`[webrtc] offer error: ${e.message}`));
           }
         } else if (msg.type === 'webrtc-ice' && msg.sessionId) {
@@ -409,7 +429,7 @@ export function initWebUi(
           }
         } else if (msg.type?.startsWith('webrtc-')) {
           msg._remoteIp = remoteIp;
-          handleWebRtcSignaling(socket as any, msg);
+          handleWebRtcSignaling(socket, msg);
         }
       } catch { /* ignore */ }
     });
@@ -418,8 +438,8 @@ export function initWebUi(
     socket.on('error', () => wsClients.delete(socket));
   });
 
-  server.listen(WEB_PORT, '0.0.0.0', () => {
-    log(`web ui listening on ${useTls ? 'https' : 'http'}://0.0.0.0:${WEB_PORT}`);
+  server.listen(port, '0.0.0.0', () => {
+    log(`web ui listening on ${useTls ? 'https' : 'http'}://0.0.0.0:${port}`);
   });
 
   setInterval(() => broadcastWs('status', getStatus()), 1000);

ts/proxybridge.ts

@@ -4,13 +4,36 @@
  * The proxy-engine handles ALL SIP protocol mechanics. TypeScript only:
  * - Sends configuration
  * - Receives high-level events (incoming_call, call_ended, etc.)
- * - Sends high-level commands (hangup, make_call, play_audio)
+ * - Sends high-level commands (hangup, make_call, add_leg, webrtc_offer)
  *
  * No raw SIP ever touches TypeScript.
  */
 
 import path from 'node:path';
 import { RustBridge } from '@push.rocks/smartrust';
+import type { TProxyEventMap } from './shared/proxy-events.ts';
+export type {
+  ICallAnsweredEvent,
+  ICallEndedEvent,
+  ICallRingingEvent,
+  IDeviceRegisteredEvent,
+  IIncomingCallEvent,
+  ILegAddedEvent,
+  ILegRemovedEvent,
+  ILegStateChangedEvent,
+  IOutboundCallEvent,
+  IOutboundCallStartedEvent,
+  IProviderRegisteredEvent,
+  IRecordingDoneEvent,
+  ISipUnhandledEvent,
+  IVoicemailErrorEvent,
+  IVoicemailStartedEvent,
+  IWebRtcAudioRxEvent,
+  IWebRtcIceCandidateEvent,
+  IWebRtcStateEvent,
+  IWebRtcTrackEvent,
+  TProxyEventMap,
+} from './shared/proxy-events.ts';
 
 // ---------------------------------------------------------------------------
 // Command type map for smartrust
@@ -29,18 +52,14 @@ type TProxyCommands = {
     params: { number: string; device_id?: string; provider_id?: string };
     result: { call_id: string };
   };
-  play_audio: {
-    params: { call_id: string; leg_id?: string; file_path: string; codec?: number };
-    result: Record<string, never>;
+  add_leg: {
+    params: { call_id: string; number: string; provider_id?: string };
+    result: { leg_id: string };
   };
-  start_recording: {
-    params: { call_id: string; file_path: string; max_duration_ms?: number };
+  remove_leg: {
+    params: { call_id: string; leg_id: string };
     result: Record<string, never>;
   };
-  stop_recording: {
-    params: { call_id: string };
-    result: { file_path: string; duration_ms: number };
-  };
   add_device_leg: {
     params: { call_id: string; device_id: string };
     result: { leg_id: string };
@@ -63,6 +82,19 @@ type TProxyCommands = {
     };
     result: { result: 'digit' | 'timeout' | 'cancelled'; digit?: string };
   };
+  start_tts_interaction: {
+    params: {
+      call_id: string;
+      leg_id: string;
+      text: string;
+      voice?: string;
+      model?: string;
+      voices?: string;
+      expected_digits: string;
+      timeout_ms: number;
+    };
+    result: { result: 'digit' | 'timeout' | 'cancelled'; digit?: string };
+  };
   add_tool_leg: {
     params: {
       call_id: string;
@@ -80,50 +112,39 @@ type TProxyCommands = {
     result: Record<string, never>;
   };
   generate_tts: {
-    params: { model: string; voices: string; voice: string; text: string; output: string };
+    params: { model: string; voices: string; voice: string; text: string; output: string; cacheable?: boolean };
     result: { output: string };
   };
+  // WebRTC signaling — bridged from the browser via the TS control plane.
+  webrtc_offer: {
+    params: { session_id: string; sdp: string };
+    result: { sdp: string };
+  };
+  webrtc_ice: {
+    params: {
+      session_id: string;
+      candidate: string;
+      sdp_mid?: string;
+      sdp_mline_index?: number;
+    };
+    result: Record<string, never>;
+  };
+  webrtc_link: {
+    params: {
+      session_id: string;
+      call_id: string;
+      provider_media_addr: string;
+      provider_media_port: number;
+      sip_pt?: number;
+    };
+    result: Record<string, never>;
+  };
+  webrtc_close: {
+    params: { session_id: string };
+    result: Record<string, never>;
+  };
 };
 
-// ---------------------------------------------------------------------------
-// Event types from Rust
-// ---------------------------------------------------------------------------
-
-export interface IIncomingCallEvent {
-  call_id: string;
-  from_uri: string;
-  to_number: string;
-  provider_id: string;
-}
-
-export interface IOutboundCallEvent {
-  call_id: string;
-  from_device: string | null;
-  to_number: string;
-}
-
-export interface ICallEndedEvent {
-  call_id: string;
-  reason: string;
-  duration: number;
-  from_side?: string;
-}
-
-export interface IProviderRegisteredEvent {
-  provider_id: string;
-  registered: boolean;
-  public_ip: string | null;
-}
-
-export interface IDeviceRegisteredEvent {
-  device_id: string;
-  display_name: string;
-  address: string;
-  port: number;
-  aor: string;
-  expires: number;
-}
-
 // ---------------------------------------------------------------------------
 // Bridge singleton
 // ---------------------------------------------------------------------------
@@ -132,10 +153,34 @@ let bridge: RustBridge<TProxyCommands> | null = null;
 let initialized = false;
 let logFn: ((msg: string) => void) | undefined;
 
+type TWebRtcIceCandidate = {
+  candidate?: string;
+  sdpMid?: string;
+  sdpMLineIndex?: number;
+} | string;
+
+function errorMessage(error: unknown): string {
+  return error instanceof Error ? error.message : String(error);
+}
+
 function buildLocalPaths(): string[] {
   const root = process.cwd();
+  // Map Node's process.arch to tsrust's friendly target name.
+  // tsrust writes multi-target binaries as <bin>_<os>_<arch>,
+  // e.g. proxy-engine_linux_amd64 / proxy-engine_linux_arm64.
+  const archSuffix =
+    process.arch === 'arm64' ? 'linux_arm64' :
+    process.arch === 'x64'   ? 'linux_amd64' :
+    null;
+  const multiTarget = archSuffix
+    ? [path.join(root, 'dist_rust', `proxy-engine_${archSuffix}`)]
+    : [];
   return [
+    // 1. Multi-target output matching the running host arch (Docker image, CI, multi-target dev).
+    ...multiTarget,
+    // 2. Single-target (unsuffixed) output — legacy/fallback when tsrust runs without targets.
     path.join(root, 'dist_rust', 'proxy-engine'),
+    // 3. Direct cargo builds for dev iteration.
     path.join(root, 'rust', 'target', 'release', 'proxy-engine'),
     path.join(root, 'rust', 'target', 'debug', 'proxy-engine'),
   ];
@@ -176,8 +221,8 @@ export async function initProxyEngine(log?: (msg: string) => void): Promise<bool
     initialized = true;
     log?.('[proxy-engine] spawned and ready');
     return true;
-  } catch (e: any) {
-    log?.(`[proxy-engine] init error: ${e.message}`);
+  } catch (error: unknown) {
+    log?.(`[proxy-engine] init error: ${errorMessage(error)}`);
     bridge = null;
     return false;
   }
@@ -187,14 +232,14 @@ export async function initProxyEngine(log?: (msg: string) => void): Promise<bool
  * Send the full app config to the proxy engine.
  * This binds the SIP socket, starts provider registrations, etc.
  */
-export async function configureProxyEngine(config: Record<string, unknown>): Promise<boolean> {
+export async function configureProxyEngine(config: TProxyCommands['configure']['params']): Promise<boolean> {
   if (!bridge || !initialized) return false;
   try {
-    const result = await bridge.sendCommand('configure', config as any);
-    logFn?.(`[proxy-engine] configured, SIP bound on ${(result as any)?.bound || '?'}`);
+    const result = await sendProxyCommand('configure', config);
+    logFn?.(`[proxy-engine] configured, SIP bound on ${result.bound || '?'}`);
     return true;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] configure error: ${e.message}`);
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] configure error: ${errorMessage(error)}`);
     return false;
   }
 }
@@ -205,14 +250,14 @@ export async function configureProxyEngine(config: Record<string, unknown>): Pro
 export async function makeCall(number: string, deviceId?: string, providerId?: string): Promise<string | null> {
   if (!bridge || !initialized) return null;
   try {
-    const result = await bridge.sendCommand('make_call', {
+    const result = await sendProxyCommand('make_call', {
       number,
       device_id: deviceId,
       provider_id: providerId,
-    } as any);
-    return (result as any)?.call_id || null;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] make_call error: ${e?.message || e}`);
+    });
+    return result.call_id || null;
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] make_call error: ${errorMessage(error)}`);
     return null;
   }
 }
@@ -223,7 +268,7 @@ export async function makeCall(number: string, deviceId?: string, providerId?: s
 export async function hangupCall(callId: string): Promise<boolean> {
   if (!bridge || !initialized) return false;
   try {
-    await bridge.sendCommand('hangup', { call_id: callId } as any);
+    await sendProxyCommand('hangup', { call_id: callId });
     return true;
   } catch {
     return false;
@@ -236,10 +281,9 @@ export async function hangupCall(callId: string): Promise<boolean> {
 export async function webrtcOffer(sessionId: string, sdp: string): Promise<{ sdp: string } | null> {
   if (!bridge || !initialized) return null;
   try {
-    const result = await bridge.sendCommand('webrtc_offer', { session_id: sessionId, sdp } as any);
-    return result as any;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] webrtc_offer error: ${e?.message || e}`);
+    return await sendProxyCommand('webrtc_offer', { session_id: sessionId, sdp });
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] webrtc_offer error: ${errorMessage(error)}`);
     return null;
   }
 }
@@ -247,15 +291,15 @@ export async function webrtcOffer(sessionId: string, sdp: string): Promise<{ sdp
 /**
  * Forward an ICE candidate to the proxy engine.
  */
-export async function webrtcIce(sessionId: string, candidate: any): Promise<void> {
+export async function webrtcIce(sessionId: string, candidate: TWebRtcIceCandidate): Promise<void> {
   if (!bridge || !initialized) return;
   try {
-    await bridge.sendCommand('webrtc_ice', {
+    await sendProxyCommand('webrtc_ice', {
       session_id: sessionId,
-      candidate: candidate?.candidate || candidate,
-      sdp_mid: candidate?.sdpMid,
-      sdp_mline_index: candidate?.sdpMLineIndex,
-    } as any);
+      candidate: typeof candidate === 'string' ? candidate : candidate.candidate || '',
+      sdp_mid: typeof candidate === 'string' ? undefined : candidate.sdpMid,
+      sdp_mline_index: typeof candidate === 'string' ? undefined : candidate.sdpMLineIndex,
+    });
   } catch { /* ignore */ }
 }
 
@@ -266,16 +310,16 @@ export async function webrtcIce(sessionId: string, candidate: any): Promise<void
 export async function webrtcLink(sessionId: string, callId: string, providerMediaAddr: string, providerMediaPort: number, sipPt: number = 9): Promise<boolean> {
   if (!bridge || !initialized) return false;
   try {
-    await bridge.sendCommand('webrtc_link', {
+    await sendProxyCommand('webrtc_link', {
       session_id: sessionId,
       call_id: callId,
       provider_media_addr: providerMediaAddr,
       provider_media_port: providerMediaPort,
       sip_pt: sipPt,
-    } as any);
+    });
     return true;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] webrtc_link error: ${e?.message || e}`);
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] webrtc_link error: ${errorMessage(error)}`);
     return false;
   }
 }
@@ -286,14 +330,14 @@ export async function webrtcLink(sessionId: string, callId: string, providerMedi
 export async function addLeg(callId: string, number: string, providerId?: string): Promise<string | null> {
   if (!bridge || !initialized) return null;
   try {
-    const result = await bridge.sendCommand('add_leg', {
+    const result = await sendProxyCommand('add_leg', {
       call_id: callId,
       number,
       provider_id: providerId,
-    } as any);
-    return (result as any)?.leg_id || null;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] add_leg error: ${e?.message || e}`);
+    });
+    return result.leg_id || null;
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] add_leg error: ${errorMessage(error)}`);
     return null;
   }
 }
@@ -304,10 +348,10 @@ export async function addLeg(callId: string, number: string, providerId?: string
 export async function removeLeg(callId: string, legId: string): Promise<boolean> {
   if (!bridge || !initialized) return false;
   try {
-    await bridge.sendCommand('remove_leg', { call_id: callId, leg_id: legId } as any);
+    await sendProxyCommand('remove_leg', { call_id: callId, leg_id: legId });
     return true;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] remove_leg error: ${e?.message || e}`);
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] remove_leg error: ${errorMessage(error)}`);
     return false;
   }
 }
@@ -318,7 +362,7 @@ export async function removeLeg(callId: string, legId: string): Promise<boolean>
 export async function webrtcClose(sessionId: string): Promise<void> {
   if (!bridge || !initialized) return;
   try {
-    await bridge.sendCommand('webrtc_close', { session_id: sessionId } as any);
+    await sendProxyCommand('webrtc_close', { session_id: sessionId });
   } catch { /* ignore */ }
 }
 
@@ -332,13 +376,13 @@ export async function webrtcClose(sessionId: string): Promise<void> {
 export async function addDeviceLeg(callId: string, deviceId: string): Promise<string | null> {
   if (!bridge || !initialized) return null;
   try {
-    const result = await bridge.sendCommand('add_device_leg', {
+    const result = await sendProxyCommand('add_device_leg', {
       call_id: callId,
       device_id: deviceId,
-    } as any);
-    return (result as any)?.leg_id || null;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] add_device_leg error: ${e?.message || e}`);
+    });
+    return result.leg_id || null;
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] add_device_leg error: ${errorMessage(error)}`);
     return null;
   }
 }
@@ -353,14 +397,14 @@ export async function transferLeg(
 ): Promise<boolean> {
   if (!bridge || !initialized) return false;
   try {
-    await bridge.sendCommand('transfer_leg', {
+    await sendProxyCommand('transfer_leg', {
       source_call_id: sourceCallId,
       leg_id: legId,
       target_call_id: targetCallId,
-    } as any);
+    });
     return true;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] transfer_leg error: ${e?.message || e}`);
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] transfer_leg error: ${errorMessage(error)}`);
     return false;
   }
 }
@@ -376,15 +420,15 @@ export async function replaceLeg(
 ): Promise<string | null> {
   if (!bridge || !initialized) return null;
   try {
-    const result = await bridge.sendCommand('replace_leg', {
+    const result = await sendProxyCommand('replace_leg', {
       call_id: callId,
       old_leg_id: oldLegId,
       number,
       provider_id: providerId,
-    } as any);
-    return (result as any)?.new_leg_id || null;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] replace_leg error: ${e?.message || e}`);
+    });
+    return result.new_leg_id || null;
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] replace_leg error: ${errorMessage(error)}`);
     return null;
   }
 }
@@ -402,16 +446,49 @@ export async function startInteraction(
 ): Promise<{ result: 'digit' | 'timeout' | 'cancelled'; digit?: string } | null> {
   if (!bridge || !initialized) return null;
   try {
-    const result = await bridge.sendCommand('start_interaction', {
+    return await sendProxyCommand('start_interaction', {
       call_id: callId,
       leg_id: legId,
       prompt_wav: promptWav,
       expected_digits: expectedDigits,
       timeout_ms: timeoutMs,
-    } as any);
-    return result as any;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] start_interaction error: ${e?.message || e}`);
+    });
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] start_interaction error: ${errorMessage(error)}`);
+    return null;
+  }
+}
+
+/**
+ * Start a live TTS interaction on a specific leg. The first chunk is rendered
+ * up front and the rest streams into the mixer while playback is already live.
+ */
+export async function startTtsInteraction(
+  callId: string,
+  legId: string,
+  text: string,
+  expectedDigits: string,
+  timeoutMs: number,
+  options?: {
+    voice?: string;
+    model?: string;
+    voices?: string;
+  },
+): Promise<{ result: 'digit' | 'timeout' | 'cancelled'; digit?: string } | null> {
+  if (!bridge || !initialized) return null;
+  try {
+    return await sendProxyCommand('start_tts_interaction', {
+      call_id: callId,
+      leg_id: legId,
+      text,
+      expected_digits: expectedDigits,
+      timeout_ms: timeoutMs,
+      voice: options?.voice,
+      model: options?.model,
+      voices: options?.voices,
+    });
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] start_tts_interaction error: ${errorMessage(error)}`);
     return null;
   }
 }
@@ -427,14 +504,14 @@ export async function addToolLeg(
 ): Promise<string | null> {
   if (!bridge || !initialized) return null;
   try {
-    const result = await bridge.sendCommand('add_tool_leg', {
+    const result = await sendProxyCommand('add_tool_leg', {
       call_id: callId,
       tool_type: toolType,
       config,
-    } as any);
-    return (result as any)?.tool_leg_id || null;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] add_tool_leg error: ${e?.message || e}`);
+    });
+    return result.tool_leg_id || null;
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] add_tool_leg error: ${errorMessage(error)}`);
     return null;
   }
 }
@@ -445,13 +522,13 @@ export async function addToolLeg(
 export async function removeToolLeg(callId: string, toolLegId: string): Promise<boolean> {
   if (!bridge || !initialized) return false;
   try {
-    await bridge.sendCommand('remove_tool_leg', {
+    await sendProxyCommand('remove_tool_leg', {
       call_id: callId,
       tool_leg_id: toolLegId,
-    } as any);
+    });
     return true;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] remove_tool_leg error: ${e?.message || e}`);
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] remove_tool_leg error: ${errorMessage(error)}`);
     return false;
   }
 }
@@ -467,15 +544,15 @@ export async function setLegMetadata(
 ): Promise<boolean> {
   if (!bridge || !initialized) return false;
   try {
-    await bridge.sendCommand('set_leg_metadata', {
+    await sendProxyCommand('set_leg_metadata', {
       call_id: callId,
       leg_id: legId,
       key,
       value,
-    } as any);
+    });
     return true;
-  } catch (e: any) {
-    logFn?.(`[proxy-engine] set_leg_metadata error: ${e?.message || e}`);
+  } catch (error: unknown) {
+    logFn?.(`[proxy-engine] set_leg_metadata error: ${errorMessage(error)}`);
     return false;
   }
 }
@@ -487,7 +564,7 @@ export async function setLegMetadata(
  * dtmf_digit, recording_done, tool_recording_done, tool_transcription_done,
  * leg_added, leg_removed, sip_unhandled
  */
-export function onProxyEvent(event: string, handler: (data: any) => void): void {
+export function onProxyEvent<K extends keyof TProxyEventMap>(event: K, handler: (data: TProxyEventMap[K]) => void): void {
   if (!bridge) throw new Error('proxy engine not initialized');
   bridge.on(`management:${event}`, handler);
 }
@@ -503,7 +580,7 @@ export async function sendProxyCommand<K extends keyof TProxyCommands>(
   params: TProxyCommands[K]['params'],
 ): Promise<TProxyCommands[K]['result']> {
   if (!bridge || !initialized) throw new Error('proxy engine not initialized');
-  return bridge.sendCommand(method as string, params as any) as any;
+  return bridge.sendCommand(method, params) as Promise<TProxyCommands[K]['result']>;
 }
 
 /** Shut down the proxy engine. */

ts/runtime/proxy-events.ts

@@ -0,0 +1,187 @@
+import { onProxyEvent } from '../proxybridge.ts';
+import type { VoiceboxManager } from '../voicebox.ts';
+import type { StatusStore } from './status-store.ts';
+import type { IProviderMediaInfo, WebRtcLinkManager } from './webrtc-linking.ts';
+
+export interface IRegisterProxyEventHandlersOptions {
+  log: (msg: string) => void;
+  statusStore: StatusStore;
+  voiceboxManager: VoiceboxManager;
+  webRtcLinks: WebRtcLinkManager;
+  getBrowserDeviceIds: () => string[];
+  sendToBrowserDevice: (deviceId: string, data: unknown) => boolean;
+  broadcast: (type: string, data: unknown) => void;
+  onLinkWebRtcSession: (callId: string, sessionId: string, media: IProviderMediaInfo) => void;
+  onCloseWebRtcSession: (sessionId: string) => void;
+}
+
+export function registerProxyEventHandlers(options: IRegisterProxyEventHandlersOptions): void {
+  const {
+    log,
+    statusStore,
+    voiceboxManager,
+    webRtcLinks,
+    getBrowserDeviceIds,
+    sendToBrowserDevice,
+    broadcast,
+    onLinkWebRtcSession,
+    onCloseWebRtcSession,
+  } = options;
+
+  onProxyEvent('provider_registered', (data) => {
+    const previous = statusStore.noteProviderRegistered(data);
+    if (previous) {
+      if (data.registered && !previous.wasRegistered) {
+        log(`[provider:${data.provider_id}] registered (publicIp=${data.public_ip})`);
+      } else if (!data.registered && previous.wasRegistered) {
+        log(`[provider:${data.provider_id}] registration lost`);
+      }
+    }
+    broadcast('registration', { providerId: data.provider_id, registered: data.registered });
+  });
+
+  onProxyEvent('device_registered', (data) => {
+    if (statusStore.noteDeviceRegistered(data)) {
+      log(`[registrar] ${data.display_name} registered from ${data.address}:${data.port}`);
+    }
+  });
+
+  onProxyEvent('incoming_call', (data) => {
+    log(`[call] incoming: ${data.from_uri} -> ${data.to_number} via ${data.provider_id} (${data.call_id})`);
+    statusStore.noteIncomingCall(data);
+
+    if (data.ring_browsers === false) {
+      return;
+    }
+
+    for (const deviceId of getBrowserDeviceIds()) {
+      sendToBrowserDevice(deviceId, {
+        type: 'webrtc-incoming',
+        callId: data.call_id,
+        from: data.from_uri,
+        deviceId,
+      });
+    }
+  });
+
+  onProxyEvent('outbound_device_call', (data) => {
+    log(`[call] outbound: device ${data.from_device} -> ${data.to_number} (${data.call_id})`);
+    statusStore.noteOutboundDeviceCall(data);
+  });
+
+  onProxyEvent('outbound_call_started', (data) => {
+    log(`[call] outbound started: ${data.call_id} -> ${data.number} via ${data.provider_id}`);
+    statusStore.noteOutboundCallStarted(data);
+
+    for (const deviceId of getBrowserDeviceIds()) {
+      sendToBrowserDevice(deviceId, {
+        type: 'webrtc-incoming',
+        callId: data.call_id,
+        from: data.number,
+        deviceId,
+      });
+    }
+  });
+
+  onProxyEvent('call_ringing', (data) => {
+    statusStore.noteCallRinging(data);
+  });
+
+  onProxyEvent('call_answered', (data) => {
+    if (statusStore.noteCallAnswered(data)) {
+      log(`[call] ${data.call_id} connected`);
+    }
+
+    if (!data.provider_media_addr || !data.provider_media_port) {
+      return;
+    }
+
+    const target = webRtcLinks.noteCallAnswered(data.call_id, {
+      addr: data.provider_media_addr,
+      port: data.provider_media_port,
+      sipPt: data.sip_pt ?? 9,
+    });
+
+    if (!target) {
+      log(`[webrtc] media info cached for call=${data.call_id}, waiting for session accept`);
+      return;
+    }
+
+    onLinkWebRtcSession(data.call_id, target.sessionId, target.media);
+  });
+
+  onProxyEvent('call_ended', (data) => {
+    if (statusStore.noteCallEnded(data)) {
+      log(`[call] ${data.call_id} ended: ${data.reason} (${data.duration}s)`);
+    }
+
+    broadcast('webrtc-call-ended', { callId: data.call_id });
+
+    const sessionId = webRtcLinks.cleanupCall(data.call_id);
+    if (sessionId) {
+      onCloseWebRtcSession(sessionId);
+    }
+  });
+
+  onProxyEvent('sip_unhandled', (data) => {
+    log(`[sip] unhandled ${data.method_or_status} Call-ID=${data.call_id?.slice(0, 20)} from=${data.from_addr}:${data.from_port}`);
+  });
+
+  onProxyEvent('leg_added', (data) => {
+    log(`[leg] added: call=${data.call_id} leg=${data.leg_id} kind=${data.kind} state=${data.state}`);
+    statusStore.noteLegAdded(data);
+  });
+
+  onProxyEvent('leg_removed', (data) => {
+    log(`[leg] removed: call=${data.call_id} leg=${data.leg_id}`);
+    statusStore.noteLegRemoved(data);
+  });
+
+  onProxyEvent('leg_state_changed', (data) => {
+    log(`[leg] state: call=${data.call_id} leg=${data.leg_id} -> ${data.state}`);
+    statusStore.noteLegStateChanged(data);
+  });
+
+  onProxyEvent('webrtc_ice_candidate', (data) => {
+    broadcast('webrtc-ice', {
+      sessionId: data.session_id,
+      candidate: {
+        candidate: data.candidate,
+        sdpMid: data.sdp_mid,
+        sdpMLineIndex: data.sdp_mline_index,
+      },
+    });
+  });
+
+  onProxyEvent('webrtc_state', (data) => {
+    log(`[webrtc] session=${data.session_id?.slice(0, 8)} state=${data.state}`);
+  });
+
+  onProxyEvent('webrtc_track', (data) => {
+    log(`[webrtc] session=${data.session_id?.slice(0, 8)} track=${data.kind} codec=${data.codec}`);
+  });
+
+  onProxyEvent('webrtc_audio_rx', (data) => {
+    if (data.packet_count === 1 || data.packet_count === 50) {
+      log(`[webrtc] session=${data.session_id?.slice(0, 8)} browser audio rx #${data.packet_count}`);
+    }
+  });
+
+  onProxyEvent('voicemail_started', (data) => {
+    log(`[voicemail] started for call ${data.call_id} caller=${data.caller_number}`);
+  });
+
+  onProxyEvent('recording_done', (data) => {
+    log(`[voicemail] recording done: ${data.file_path} (${data.duration_ms}ms) caller=${data.caller_number}`);
+    voiceboxManager.addMessage('default', {
+      callerNumber: data.caller_number || 'Unknown',
+      callerName: null,
+      fileName: data.file_path,
+      durationMs: data.duration_ms,
+    });
+  });
+
+  onProxyEvent('voicemail_error', (data) => {
+    log(`[voicemail] error: ${data.error} call=${data.call_id}`);
+  });
+}

ts/runtime/status-store.ts

@@ -0,0 +1,313 @@
+import type { IAppConfig } from '../config.ts';
+import type {
+  ICallAnsweredEvent,
+  ICallEndedEvent,
+  ICallRingingEvent,
+  IDeviceRegisteredEvent,
+  IIncomingCallEvent,
+  ILegAddedEvent,
+  ILegRemovedEvent,
+  ILegStateChangedEvent,
+  IOutboundCallEvent,
+  IOutboundCallStartedEvent,
+  IProviderRegisteredEvent,
+} from '../shared/proxy-events.ts';
+import type {
+  IActiveCall,
+  ICallHistoryEntry,
+  IDeviceStatus,
+  IProviderStatus,
+  IStatusSnapshot,
+  TLegType,
+} from '../shared/status.ts';
+
+const MAX_HISTORY = 100;
+const CODEC_NAMES: Record<number, string> = {
+  0: 'PCMU',
+  8: 'PCMA',
+  9: 'G.722',
+  111: 'Opus',
+};
+
+export class StatusStore {
+  private appConfig: IAppConfig;
+  private providerStatuses = new Map<string, IProviderStatus>();
+  private deviceStatuses = new Map<string, IDeviceStatus>();
+  private activeCalls = new Map<string, IActiveCall>();
+  private callHistory: ICallHistoryEntry[] = [];
+
+  constructor(appConfig: IAppConfig) {
+    this.appConfig = appConfig;
+    this.rebuildConfigState();
+  }
+
+  updateConfig(appConfig: IAppConfig): void {
+    this.appConfig = appConfig;
+    this.rebuildConfigState();
+  }
+
+  buildStatusSnapshot(
+    instanceId: string,
+    startTime: number,
+    browserDeviceIds: string[],
+    voicemailCounts: Record<string, number>,
+  ): IStatusSnapshot {
+    const devices = [...this.deviceStatuses.values()];
+    for (const deviceId of browserDeviceIds) {
+      devices.push({
+        id: deviceId,
+        displayName: 'Browser',
+        address: null,
+        port: 0,
+        aor: null,
+        connected: true,
+        isBrowser: true,
+      });
+    }
+
+    return {
+      instanceId,
+      uptime: Math.floor((Date.now() - startTime) / 1000),
+      lanIp: this.appConfig.proxy.lanIp,
+      providers: [...this.providerStatuses.values()],
+      devices,
+      calls: [...this.activeCalls.values()].map((call) => ({
+        ...call,
+        duration: Math.floor((Date.now() - call.startedAt) / 1000),
+        legs: [...call.legs.values()].map((leg) => ({
+          ...leg,
+          pktSent: 0,
+          pktReceived: 0,
+          transcoding: false,
+        })),
+      })),
+      callHistory: this.callHistory,
+      contacts: this.appConfig.contacts || [],
+      voicemailCounts,
+    };
+  }
+
+  noteDashboardCallStarted(callId: string, number: string, providerId?: string): void {
+    this.activeCalls.set(callId, {
+      id: callId,
+      direction: 'outbound',
+      callerNumber: null,
+      calleeNumber: number,
+      providerUsed: providerId || null,
+      state: 'setting-up',
+      startedAt: Date.now(),
+      legs: new Map(),
+    });
+  }
+
+  noteProviderRegistered(data: IProviderRegisteredEvent): { wasRegistered: boolean } | null {
+    const provider = this.providerStatuses.get(data.provider_id);
+    if (!provider) {
+      return null;
+    }
+
+    const wasRegistered = provider.registered;
+    provider.registered = data.registered;
+    provider.publicIp = data.public_ip;
+    return { wasRegistered };
+  }
+
+  noteDeviceRegistered(data: IDeviceRegisteredEvent): boolean {
+    const device = this.deviceStatuses.get(data.device_id);
+    if (!device) {
+      return false;
+    }
+
+    device.address = data.address;
+    device.port = data.port;
+    device.aor = data.aor;
+    device.connected = true;
+    return true;
+  }
+
+  noteIncomingCall(data: IIncomingCallEvent): void {
+    this.activeCalls.set(data.call_id, {
+      id: data.call_id,
+      direction: 'inbound',
+      callerNumber: data.from_uri,
+      calleeNumber: data.to_number,
+      providerUsed: data.provider_id,
+      state: 'ringing',
+      startedAt: Date.now(),
+      legs: new Map(),
+    });
+  }
+
+  noteOutboundDeviceCall(data: IOutboundCallEvent): void {
+    this.activeCalls.set(data.call_id, {
+      id: data.call_id,
+      direction: 'outbound',
+      callerNumber: data.from_device,
+      calleeNumber: data.to_number,
+      providerUsed: null,
+      state: 'setting-up',
+      startedAt: Date.now(),
+      legs: new Map(),
+    });
+  }
+
+  noteOutboundCallStarted(data: IOutboundCallStartedEvent): void {
+    this.activeCalls.set(data.call_id, {
+      id: data.call_id,
+      direction: 'outbound',
+      callerNumber: null,
+      calleeNumber: data.number,
+      providerUsed: data.provider_id,
+      state: 'setting-up',
+      startedAt: Date.now(),
+      legs: new Map(),
+    });
+  }
+
+  noteCallRinging(data: ICallRingingEvent): void {
+    const call = this.activeCalls.get(data.call_id);
+    if (call) {
+      call.state = 'ringing';
+    }
+  }
+
+  noteCallAnswered(data: ICallAnsweredEvent): boolean {
+    const call = this.activeCalls.get(data.call_id);
+    if (!call) {
+      return false;
+    }
+
+    call.state = 'connected';
+
+    if (data.provider_media_addr && data.provider_media_port) {
+      for (const leg of call.legs.values()) {
+        if (leg.type !== 'sip-provider') {
+          continue;
+        }
+
+        leg.remoteMedia = `${data.provider_media_addr}:${data.provider_media_port}`;
+        if (data.sip_pt !== undefined) {
+          leg.codec = CODEC_NAMES[data.sip_pt] || `PT${data.sip_pt}`;
+        }
+        break;
+      }
+    }
+
+    return true;
+  }
+
+  noteCallEnded(data: ICallEndedEvent): boolean {
+    const call = this.activeCalls.get(data.call_id);
+    if (!call) {
+      return false;
+    }
+
+    this.callHistory.unshift({
+      id: call.id,
+      direction: call.direction,
+      callerNumber: call.callerNumber,
+      calleeNumber: call.calleeNumber,
+      providerUsed: call.providerUsed,
+      startedAt: call.startedAt,
+      duration: data.duration,
+      legs: [...call.legs.values()].map((leg) => ({
+        id: leg.id,
+        type: leg.type,
+        metadata: leg.metadata || {},
+      })),
+    });
+
+    if (this.callHistory.length > MAX_HISTORY) {
+      this.callHistory.pop();
+    }
+
+    this.activeCalls.delete(data.call_id);
+    return true;
+  }
+
+  noteLegAdded(data: ILegAddedEvent): void {
+    const call = this.activeCalls.get(data.call_id);
+    if (!call) {
+      return;
+    }
+
+    call.legs.set(data.leg_id, {
+      id: data.leg_id,
+      type: data.kind,
+      state: data.state,
+      codec: data.codec ?? null,
+      rtpPort: data.rtpPort ?? null,
+      remoteMedia: data.remoteMedia ?? null,
+      metadata: data.metadata || {},
+    });
+  }
+
+  noteLegRemoved(data: ILegRemovedEvent): void {
+    this.activeCalls.get(data.call_id)?.legs.delete(data.leg_id);
+  }
+
+  noteLegStateChanged(data: ILegStateChangedEvent): void {
+    const call = this.activeCalls.get(data.call_id);
+    if (!call) {
+      return;
+    }
+
+    const existingLeg = call.legs.get(data.leg_id);
+    if (existingLeg) {
+      existingLeg.state = data.state;
+      if (data.metadata) {
+        existingLeg.metadata = data.metadata;
+      }
+      return;
+    }
+
+    call.legs.set(data.leg_id, {
+      id: data.leg_id,
+      type: this.inferLegType(data.leg_id),
+      state: data.state,
+      codec: null,
+      rtpPort: null,
+      remoteMedia: null,
+      metadata: data.metadata || {},
+    });
+  }
+
+  private rebuildConfigState(): void {
+    const nextProviderStatuses = new Map<string, IProviderStatus>();
+    for (const provider of this.appConfig.providers) {
+      const previous = this.providerStatuses.get(provider.id);
+      nextProviderStatuses.set(provider.id, {
+        id: provider.id,
+        displayName: provider.displayName,
+        registered: previous?.registered ?? false,
+        publicIp: previous?.publicIp ?? null,
+      });
+    }
+    this.providerStatuses = nextProviderStatuses;
+
+    const nextDeviceStatuses = new Map<string, IDeviceStatus>();
+    for (const device of this.appConfig.devices) {
+      const previous = this.deviceStatuses.get(device.id);
+      nextDeviceStatuses.set(device.id, {
+        id: device.id,
+        displayName: device.displayName,
+        address: previous?.address ?? null,
+        port: previous?.port ?? 0,
+        aor: previous?.aor ?? null,
+        connected: previous?.connected ?? false,
+        isBrowser: false,
+      });
+    }
+    this.deviceStatuses = nextDeviceStatuses;
+  }
+
+  private inferLegType(legId: string): TLegType {
+    if (legId.includes('-prov')) {
+      return 'sip-provider';
+    }
+    if (legId.includes('-dev')) {
+      return 'sip-device';
+    }
+    return 'webrtc';
+  }
+}

ts/runtime/webrtc-linking.ts

@@ -0,0 +1,66 @@
+export interface IProviderMediaInfo {
+  addr: string;
+  port: number;
+  sipPt: number;
+}
+
+export interface IWebRtcLinkTarget {
+  sessionId: string;
+  media: IProviderMediaInfo;
+}
+
+export class WebRtcLinkManager {
+  private sessionToCall = new Map<string, string>();
+  private callToSession = new Map<string, string>();
+  private pendingCallMedia = new Map<string, IProviderMediaInfo>();
+
+  acceptCall(callId: string, sessionId: string): IProviderMediaInfo | null {
+    const previousCallId = this.sessionToCall.get(sessionId);
+    if (previousCallId && previousCallId !== callId) {
+      this.callToSession.delete(previousCallId);
+    }
+
+    const previousSessionId = this.callToSession.get(callId);
+    if (previousSessionId && previousSessionId !== sessionId) {
+      this.sessionToCall.delete(previousSessionId);
+    }
+
+    this.sessionToCall.set(sessionId, callId);
+    this.callToSession.set(callId, sessionId);
+
+    const pendingMedia = this.pendingCallMedia.get(callId) ?? null;
+    if (pendingMedia) {
+      this.pendingCallMedia.delete(callId);
+    }
+    return pendingMedia;
+  }
+
+  noteCallAnswered(callId: string, media: IProviderMediaInfo): IWebRtcLinkTarget | null {
+    const sessionId = this.callToSession.get(callId);
+    if (!sessionId) {
+      this.pendingCallMedia.set(callId, media);
+      return null;
+    }
+
+    return { sessionId, media };
+  }
+
+  removeSession(sessionId: string): string | null {
+    const callId = this.sessionToCall.get(sessionId) ?? null;
+    this.sessionToCall.delete(sessionId);
+    if (callId) {
+      this.callToSession.delete(callId);
+    }
+    return callId;
+  }
+
+  cleanupCall(callId: string): string | null {
+    const sessionId = this.callToSession.get(callId) ?? null;
+    this.callToSession.delete(callId);
+    this.pendingCallMedia.delete(callId);
+    if (sessionId) {
+      this.sessionToCall.delete(sessionId);
+    }
+    return sessionId;
+  }
+}

ts/shared/proxy-events.ts

@@ -0,0 +1,145 @@
+import type { TLegType } from './status.ts';
+
+export interface IIncomingCallEvent {
+  call_id: string;
+  from_uri: string;
+  to_number: string;
+  provider_id: string;
+  ring_browsers?: boolean;
+}
+
+export interface IOutboundCallEvent {
+  call_id: string;
+  from_device: string | null;
+  to_number: string;
+}
+
+export interface IOutboundCallStartedEvent {
+  call_id: string;
+  number: string;
+  provider_id: string;
+}
+
+export interface ICallRingingEvent {
+  call_id: string;
+}
+
+export interface ICallAnsweredEvent {
+  call_id: string;
+  provider_media_addr?: string;
+  provider_media_port?: number;
+  sip_pt?: number;
+}
+
+export interface ICallEndedEvent {
+  call_id: string;
+  reason: string;
+  duration: number;
+  from_side?: string;
+}
+
+export interface IProviderRegisteredEvent {
+  provider_id: string;
+  registered: boolean;
+  public_ip: string | null;
+}
+
+export interface IDeviceRegisteredEvent {
+  device_id: string;
+  display_name: string;
+  address: string;
+  port: number;
+  aor: string;
+  expires: number;
+}
+
+export interface ISipUnhandledEvent {
+  method_or_status: string;
+  call_id?: string;
+  from_addr: string;
+  from_port: number;
+}
+
+export interface ILegAddedEvent {
+  call_id: string;
+  leg_id: string;
+  kind: TLegType;
+  state: string;
+  codec?: string | null;
+  rtpPort?: number | null;
+  remoteMedia?: string | null;
+  metadata?: Record<string, unknown>;
+}
+
+export interface ILegRemovedEvent {
+  call_id: string;
+  leg_id: string;
+}
+
+export interface ILegStateChangedEvent {
+  call_id: string;
+  leg_id: string;
+  state: string;
+  metadata?: Record<string, unknown>;
+}
+
+export interface IWebRtcIceCandidateEvent {
+  session_id: string;
+  candidate: string;
+  sdp_mid?: string;
+  sdp_mline_index?: number;
+}
+
+export interface IWebRtcStateEvent {
+  session_id?: string;
+  state: string;
+}
+
+export interface IWebRtcTrackEvent {
+  session_id?: string;
+  kind: string;
+  codec: string;
+}
+
+export interface IWebRtcAudioRxEvent {
+  session_id?: string;
+  packet_count: number;
+}
+
+export interface IVoicemailStartedEvent {
+  call_id: string;
+  caller_number?: string;
+}
+
+export interface IRecordingDoneEvent {
+  file_path: string;
+  duration_ms: number;
+  caller_number?: string;
+}
+
+export interface IVoicemailErrorEvent {
+  call_id: string;
+  error: string;
+}
+
+export type TProxyEventMap = {
+  provider_registered: IProviderRegisteredEvent;
+  device_registered: IDeviceRegisteredEvent;
+  incoming_call: IIncomingCallEvent;
+  outbound_device_call: IOutboundCallEvent;
+  outbound_call_started: IOutboundCallStartedEvent;
+  call_ringing: ICallRingingEvent;
+  call_answered: ICallAnsweredEvent;
+  call_ended: ICallEndedEvent;
+  sip_unhandled: ISipUnhandledEvent;
+  leg_added: ILegAddedEvent;
+  leg_removed: ILegRemovedEvent;
+  leg_state_changed: ILegStateChangedEvent;
+  webrtc_ice_candidate: IWebRtcIceCandidateEvent;
+  webrtc_state: IWebRtcStateEvent;
+  webrtc_track: IWebRtcTrackEvent;
+  webrtc_audio_rx: IWebRtcAudioRxEvent;
+  voicemail_started: IVoicemailStartedEvent;
+  recording_done: IRecordingDoneEvent;
+  voicemail_error: IVoicemailErrorEvent;
+};

ts/shared/status.ts

@@ -0,0 +1,89 @@
+import type { IContact } from '../config.ts';
+
+export type TLegType = 'sip-device' | 'sip-provider' | 'webrtc' | 'tool';
+export type TCallDirection = 'inbound' | 'outbound';
+
+export interface IProviderStatus {
+  id: string;
+  displayName: string;
+  registered: boolean;
+  publicIp: string | null;
+}
+
+export interface IDeviceStatus {
+  id: string;
+  displayName: string;
+  address: string | null;
+  port: number;
+  aor: string | null;
+  connected: boolean;
+  isBrowser: boolean;
+}
+
+export interface IActiveLeg {
+  id: string;
+  type: TLegType;
+  state: string;
+  codec: string | null;
+  rtpPort: number | null;
+  remoteMedia: string | null;
+  metadata: Record<string, unknown>;
+}
+
+export interface IActiveCall {
+  id: string;
+  direction: TCallDirection;
+  callerNumber: string | null;
+  calleeNumber: string | null;
+  providerUsed: string | null;
+  state: string;
+  startedAt: number;
+  legs: Map<string, IActiveLeg>;
+}
+
+export interface IHistoryLeg {
+  id: string;
+  type: string;
+  metadata: Record<string, unknown>;
+}
+
+export interface ICallHistoryEntry {
+  id: string;
+  direction: TCallDirection;
+  callerNumber: string | null;
+  calleeNumber: string | null;
+  providerUsed: string | null;
+  startedAt: number;
+  duration: number;
+  legs: IHistoryLeg[];
+}
+
+export interface ILegStatus extends IActiveLeg {
+  pktSent: number;
+  pktReceived: number;
+  transcoding: boolean;
+}
+
+export interface ICallStatus {
+  id: string;
+  direction: TCallDirection;
+  callerNumber: string | null;
+  calleeNumber: string | null;
+  providerUsed: string | null;
+  state: string;
+  startedAt: number;
+  duration: number;
+  legs: ILegStatus[];
+}
+
+export interface IStatusSnapshot {
+  instanceId: string;
+  uptime: number;
+  lanIp: string;
+  providers: IProviderStatus[];
+  devices: IDeviceStatus[];
+  calls: ICallStatus[];
+  callHistory: ICallHistoryEntry[];
+  contacts: IContact[];
+  voicemailCounts: Record<string, number>;
+}

ts/sipproxy.ts

@@ -1,36 +1,20 @@
 /**
- * SIP proxy — entry point.
+ * SIP proxy bootstrap.
  *
- * Spawns the Rust proxy-engine which handles ALL SIP protocol mechanics.
- * TypeScript is the control plane:
- * - Loads config and pushes it to Rust
- * - Receives high-level events (incoming calls, registration, etc.)
- * - Drives the web dashboard
- * - Manages IVR, voicemail, announcements
- * - Handles WebRTC browser signaling (forwarded to Rust in Phase 2)
- *
- * No raw SIP ever touches TypeScript.
+ * Spawns the Rust proxy-engine, wires runtime state/event handling,
+ * and starts the web dashboard plus browser signaling layer.
  */
 
 import fs from 'node:fs';
 import path from 'node:path';
 
-import { loadConfig } from './config.ts';
-import type { IAppConfig } from './config.ts';
+import { loadConfig, type IAppConfig } from './config.ts';
 import { broadcastWs, initWebUi } from './frontend.ts';
-import {
-  initWebRtcSignaling,
-  sendToBrowserDevice,
-  getAllBrowserDeviceIds,
-  getBrowserDeviceWs,
-} from './webrtcbridge.ts';
-import { initAnnouncement } from './announcement.ts';
-import { PromptCache } from './call/prompt-cache.ts';
+import { initWebRtcSignaling, getAllBrowserDeviceIds, sendToBrowserDevice } from './webrtcbridge.ts';
 import { VoiceboxManager } from './voicebox.ts';
 import {
   initProxyEngine,
   configureProxyEngine,
-  onProxyEvent,
   hangupCall,
   makeCall,
   shutdownProxyEngine,
@@ -38,640 +22,200 @@ import {
   webrtcIce,
   webrtcLink,
   webrtcClose,
-  addLeg,
-  removeLeg,
 } from './proxybridge.ts';
-import type {
-  IIncomingCallEvent,
-  IOutboundCallEvent,
-  ICallEndedEvent,
-  IProviderRegisteredEvent,
-  IDeviceRegisteredEvent,
-} from './proxybridge.ts';
-
-// ---------------------------------------------------------------------------
-// Config
-// ---------------------------------------------------------------------------
+import { registerProxyEventHandlers } from './runtime/proxy-events.ts';
+import { StatusStore } from './runtime/status-store.ts';
+import { WebRtcLinkManager, type IProviderMediaInfo } from './runtime/webrtc-linking.ts';
 
 let appConfig: IAppConfig = loadConfig();
 
 const LOG_PATH = path.join(process.cwd(), 'sip_trace.log');
-
-// ---------------------------------------------------------------------------
-// Logging
-// ---------------------------------------------------------------------------
-
 const startTime = Date.now();
 const instanceId = `${Date.now()}-${Math.random().toString(36).slice(2, 8)}`;
 
+const statusStore = new StatusStore(appConfig);
+const webRtcLinks = new WebRtcLinkManager();
+const voiceboxManager = new VoiceboxManager(log);
+
+voiceboxManager.init(appConfig.voiceboxes ?? []);
+initWebRtcSignaling({ log });
+
 function now(): string {
   return new Date().toISOString().replace('T', ' ').slice(0, 19);
 }
 
-function log(msg: string): void {
-  const line = `${now()} ${msg}\n`;
+function log(message: string): void {
+  const line = `${now()} ${message}\n`;
   fs.appendFileSync(LOG_PATH, line);
   process.stdout.write(line);
-  broadcastWs('log', { message: msg });
+  broadcastWs('log', { message });
 }
 
-// ---------------------------------------------------------------------------
-// Shadow state — maintained from Rust events for the dashboard
-// ---------------------------------------------------------------------------
-
-interface IProviderStatus {
-  id: string;
-  displayName: string;
-  registered: boolean;
-  publicIp: string | null;
+function errorMessage(error: unknown): string {
+  return error instanceof Error ? error.message : String(error);
 }
 
-interface IDeviceStatus {
-  id: string;
-  displayName: string;
-  address: string | null;
-  port: number;
-  connected: boolean;
-  isBrowser: boolean;
-}
-
-interface IActiveLeg {
-  id: string;
-  type: 'sip-device' | 'sip-provider' | 'webrtc' | 'tool';
-  state: string;
-  codec: string | null;
-  rtpPort: number | null;
-  remoteMedia: string | null;
-  metadata: Record<string, unknown>;
-}
-
-interface IActiveCall {
-  id: string;
-  direction: string;
-  callerNumber: string | null;
-  calleeNumber: string | null;
-  providerUsed: string | null;
-  state: string;
-  startedAt: number;
-  legs: Map<string, IActiveLeg>;
-}
-
-interface IHistoryLeg {
-  id: string;
-  type: string;
-  metadata: Record<string, unknown>;
-}
-
-interface ICallHistoryEntry {
-  id: string;
-  direction: string;
-  callerNumber: string | null;
-  calleeNumber: string | null;
-  startedAt: number;
-  duration: number;
-  legs: IHistoryLeg[];
-}
-
-const providerStatuses = new Map<string, IProviderStatus>();
-const deviceStatuses = new Map<string, IDeviceStatus>();
-const activeCalls = new Map<string, IActiveCall>();
-const callHistory: ICallHistoryEntry[] = [];
-const MAX_HISTORY = 100;
-
-// WebRTC session ↔ call linking state.
-// Both pieces (session accept + call media info) can arrive in any order.
-const webrtcSessionToCall = new Map<string, string>(); // sessionId → callId
-const webrtcCallToSession = new Map<string, string>(); // callId → sessionId
-const pendingCallMedia = new Map<string, { addr: string; port: number; sipPt: number }>(); // callId → provider media info
-
-// Initialize provider statuses from config (all start as unregistered).
-for (const p of appConfig.providers) {
-  providerStatuses.set(p.id, {
-    id: p.id,
-    displayName: p.displayName,
-    registered: false,
-    publicIp: null,
-  });
-}
-
-// Initialize device statuses from config.
-for (const d of appConfig.devices) {
-  deviceStatuses.set(d.id, {
-    id: d.id,
-    displayName: d.displayName,
-    address: null,
-    port: 0,
-    connected: false,
-    isBrowser: false,
-  });
-}
-
-// ---------------------------------------------------------------------------
-// Initialize subsystems
-// ---------------------------------------------------------------------------
-
-const promptCache = new PromptCache(log);
-const voiceboxManager = new VoiceboxManager(log);
-voiceboxManager.init(appConfig.voiceboxes ?? []);
-
-// WebRTC signaling (browser device registration).
-initWebRtcSignaling({ log });
-
-// ---------------------------------------------------------------------------
-// Status snapshot (fed to web dashboard)
-// ---------------------------------------------------------------------------
-
-function getStatus() {
-  // Merge SIP devices (from Rust) + browser devices (from TS WebSocket).
-  const devices = [...deviceStatuses.values()];
-  for (const bid of getAllBrowserDeviceIds()) {
-    devices.push({
-      id: bid,
-      displayName: 'Browser',
-      address: null,
-      port: 0,
-      connected: true,
-      isBrowser: true,
-    });
-  }
-
+function buildProxyConfig(config: IAppConfig): Record<string, unknown> {
   return {
-    instanceId,
-    uptime: Math.floor((Date.now() - startTime) / 1000),
-    lanIp: appConfig.proxy.lanIp,
-    providers: [...providerStatuses.values()],
-    devices,
-    calls: [...activeCalls.values()].map((c) => ({
-      ...c,
-      duration: Math.floor((Date.now() - c.startedAt) / 1000),
-      legs: [...c.legs.values()].map((l) => ({
-        id: l.id,
-        type: l.type,
-        state: l.state,
-        codec: l.codec,
-        rtpPort: l.rtpPort,
-        remoteMedia: l.remoteMedia,
-        metadata: l.metadata || {},
-        pktSent: 0,
-        pktReceived: 0,
-        transcoding: false,
-      })),
-    })),
-    callHistory,
-    contacts: appConfig.contacts || [],
-    voicemailCounts: voiceboxManager.getAllUnheardCounts(),
+    proxy: config.proxy,
+    providers: config.providers,
+    devices: config.devices,
+    routing: config.routing,
+    voiceboxes: config.voiceboxes ?? [],
+    ivr: config.ivr,
   };
 }
 
-// ---------------------------------------------------------------------------
-// Start Rust proxy engine
-// ---------------------------------------------------------------------------
+function getStatus() {
+  return statusStore.buildStatusSnapshot(
+    instanceId,
+    startTime,
+    getAllBrowserDeviceIds(),
+    voiceboxManager.getAllUnheardCounts(),
+  );
+}
+
+function requestWebRtcLink(callId: string, sessionId: string, media: IProviderMediaInfo): void {
+  log(`[webrtc] linking session=${sessionId.slice(0, 8)} to call=${callId} media=${media.addr}:${media.port} pt=${media.sipPt}`);
+  void webrtcLink(sessionId, callId, media.addr, media.port, media.sipPt).then((ok) => {
+    log(`[webrtc] link result: ${ok}`);
+  });
+}
+
+async function configureRuntime(config: IAppConfig): Promise<boolean> {
+  return configureProxyEngine(buildProxyConfig(config));
+}
+
+async function reloadConfig(): Promise<void> {
+  try {
+    const previousConfig = appConfig;
+    const nextConfig = loadConfig();
+
+    appConfig = nextConfig;
+    statusStore.updateConfig(nextConfig);
+    voiceboxManager.init(nextConfig.voiceboxes ?? []);
+
+    if (nextConfig.proxy.lanPort !== previousConfig.proxy.lanPort) {
+      log('[config] proxy.lanPort changed; restart required for SIP socket rebinding');
+    }
+    if (nextConfig.proxy.webUiPort !== previousConfig.proxy.webUiPort) {
+      log('[config] proxy.webUiPort changed; restart required for web UI rebinding');
+    }
+
+    const configured = await configureRuntime(nextConfig);
+    if (configured) {
+      log('[config] reloaded - proxy engine reconfigured');
+    } else {
+      log('[config] reload failed - proxy engine rejected config');
+    }
+  } catch (error: unknown) {
+    log(`[config] reload failed: ${errorMessage(error)}`);
+  }
+}
 
 async function startProxyEngine(): Promise<void> {
-  const ok = await initProxyEngine(log);
-  if (!ok) {
+  const started = await initProxyEngine(log);
+  if (!started) {
     log('[FATAL] failed to start proxy engine');
     process.exit(1);
   }
 
-  // Subscribe to events from Rust BEFORE sending configure.
-  onProxyEvent('provider_registered', (data: IProviderRegisteredEvent) => {
-    const ps = providerStatuses.get(data.provider_id);
-    if (ps) {
-      const wasRegistered = ps.registered;
-      ps.registered = data.registered;
-      ps.publicIp = data.public_ip;
-      if (data.registered && !wasRegistered) {
-        log(`[provider:${data.provider_id}] registered (publicIp=${data.public_ip})`);
-      } else if (!data.registered && wasRegistered) {
-        log(`[provider:${data.provider_id}] registration lost`);
-      }
-      broadcastWs('registration', { providerId: data.provider_id, registered: data.registered });
-    }
-  });
-
-  onProxyEvent('device_registered', (data: IDeviceRegisteredEvent) => {
-    const ds = deviceStatuses.get(data.device_id);
-    if (ds) {
-      ds.address = data.address;
-      ds.port = data.port;
-      ds.connected = true;
-      log(`[registrar] ${data.display_name} registered from ${data.address}:${data.port}`);
-    }
-  });
-
-  onProxyEvent('incoming_call', (data: IIncomingCallEvent) => {
-    log(`[call] incoming: ${data.from_uri} → ${data.to_number} via ${data.provider_id} (${data.call_id})`);
-    activeCalls.set(data.call_id, {
-      id: data.call_id,
-      direction: 'inbound',
-      callerNumber: data.from_uri,
-      calleeNumber: data.to_number,
-      providerUsed: data.provider_id,
-      state: 'ringing',
-      startedAt: Date.now(),
-      legs: new Map(),
-    });
-
-    // Notify browsers of incoming call.
-    const browserIds = getAllBrowserDeviceIds();
-    for (const bid of browserIds) {
-      sendToBrowserDevice(bid, {
-        type: 'webrtc-incoming',
-        callId: data.call_id,
-        from: data.from_uri,
-        deviceId: bid,
-      });
-    }
-  });
-
-  onProxyEvent('outbound_device_call', (data: IOutboundCallEvent) => {
-    log(`[call] outbound: device ${data.from_device} → ${data.to_number} (${data.call_id})`);
-    activeCalls.set(data.call_id, {
-      id: data.call_id,
-      direction: 'outbound',
-      callerNumber: data.from_device,
-      calleeNumber: data.to_number,
-      providerUsed: null,
-      state: 'setting-up',
-      startedAt: Date.now(),
-      legs: new Map(),
-    });
-  });
-
-  onProxyEvent('outbound_call_started', (data: any) => {
-    log(`[call] outbound started: ${data.call_id} → ${data.number} via ${data.provider_id}`);
-    activeCalls.set(data.call_id, {
-      id: data.call_id,
-      direction: 'outbound',
-      callerNumber: null,
-      calleeNumber: data.number,
-      providerUsed: data.provider_id,
-      state: 'setting-up',
-      startedAt: Date.now(),
-      legs: new Map(),
-    });
-
-    // Notify all browser devices — they can connect via WebRTC to listen/talk.
-    const browserIds = getAllBrowserDeviceIds();
-    for (const bid of browserIds) {
-      sendToBrowserDevice(bid, {
-        type: 'webrtc-incoming',
-        callId: data.call_id,
-        from: data.number,
-        deviceId: bid,
-      });
-    }
-  });
-
-  onProxyEvent('call_ringing', (data: { call_id: string }) => {
-    const call = activeCalls.get(data.call_id);
-    if (call) call.state = 'ringing';
-  });
-
-  onProxyEvent('call_answered', (data: { call_id: string; provider_media_addr?: string; provider_media_port?: number; sip_pt?: number }) => {
-    const call = activeCalls.get(data.call_id);
-    if (call) {
-      call.state = 'connected';
-      log(`[call] ${data.call_id} connected`);
-
-      // Enrich provider leg with media info from the answered event.
-      if (data.provider_media_addr && data.provider_media_port) {
-        for (const leg of call.legs.values()) {
-          if (leg.type === 'sip-provider') {
-            leg.remoteMedia = `${data.provider_media_addr}:${data.provider_media_port}`;
-            if (data.sip_pt !== undefined) {
-              const codecNames: Record<number, string> = { 0: 'PCMU', 8: 'PCMA', 9: 'G.722', 111: 'Opus' };
-              leg.codec = codecNames[data.sip_pt] || `PT${data.sip_pt}`;
-            }
-            break;
-          }
-        }
-      }
-    }
-
-    // Try to link WebRTC session to this call for audio bridging.
-    if (data.provider_media_addr && data.provider_media_port) {
-      const sessionId = webrtcCallToSession.get(data.call_id);
-      if (sessionId) {
-        // Both session and media info available — link now.
-        const sipPt = data.sip_pt ?? 9;
-        log(`[webrtc] linking session=${sessionId.slice(0, 8)} to call=${data.call_id} media=${data.provider_media_addr}:${data.provider_media_port} pt=${sipPt}`);
-        webrtcLink(sessionId, data.call_id, data.provider_media_addr, data.provider_media_port, sipPt).then((ok) => {
-          log(`[webrtc] link result: ${ok}`);
-        });
-      } else {
-        // Session not yet accepted — store media info for when it arrives.
-        pendingCallMedia.set(data.call_id, {
-          addr: data.provider_media_addr,
-          port: data.provider_media_port,
-          sipPt: data.sip_pt ?? 9,
-        });
-        log(`[webrtc] media info cached for call=${data.call_id}, waiting for session accept`);
-      }
-    }
-  });
-
-  onProxyEvent('call_ended', (data: ICallEndedEvent) => {
-    const call = activeCalls.get(data.call_id);
-    if (call) {
-      log(`[call] ${data.call_id} ended: ${data.reason} (${data.duration}s)`);
-      // Snapshot legs with metadata for history.
-      const historyLegs: IHistoryLeg[] = [];
-      for (const [, leg] of call.legs) {
-        historyLegs.push({
-          id: leg.id,
-          type: leg.type,
-          metadata: leg.metadata || {},
-        });
-      }
-      // Move to history.
-      callHistory.unshift({
-        id: call.id,
-        direction: call.direction,
-        callerNumber: call.callerNumber,
-        calleeNumber: call.calleeNumber,
-        startedAt: call.startedAt,
-        duration: data.duration,
-        legs: historyLegs,
-      });
-      if (callHistory.length > MAX_HISTORY) callHistory.pop();
-      activeCalls.delete(data.call_id);
-
-      // Notify browser(s) that the call ended.
-      broadcastWs('webrtc-call-ended', { callId: data.call_id });
-
-      // Clean up WebRTC session mappings.
-      const sessionId = webrtcCallToSession.get(data.call_id);
-      if (sessionId) {
-        webrtcCallToSession.delete(data.call_id);
-        webrtcSessionToCall.delete(sessionId);
-        webrtcClose(sessionId).catch(() => {});
-      }
-      pendingCallMedia.delete(data.call_id);
-    }
-  });
-
-  onProxyEvent('sip_unhandled', (data: any) => {
-    log(`[sip] unhandled ${data.method_or_status} Call-ID=${data.call_id?.slice(0, 20)} from=${data.from_addr}:${data.from_port}`);
-  });
-
-  // Leg events (multiparty) — update shadow state so the dashboard shows legs.
-  onProxyEvent('leg_added', (data: any) => {
-    log(`[leg] added: call=${data.call_id} leg=${data.leg_id} kind=${data.kind} state=${data.state}`);
-    const call = activeCalls.get(data.call_id);
-    if (call) {
-      call.legs.set(data.leg_id, {
-        id: data.leg_id,
-        type: data.kind,
-        state: data.state,
-        codec: data.codec ?? null,
-        rtpPort: data.rtpPort ?? null,
-        remoteMedia: data.remoteMedia ?? null,
-        metadata: data.metadata || {},
-      });
-    }
-  });
-
-  onProxyEvent('leg_removed', (data: any) => {
-    log(`[leg] removed: call=${data.call_id} leg=${data.leg_id}`);
-    activeCalls.get(data.call_id)?.legs.delete(data.leg_id);
-  });
-
-  onProxyEvent('leg_state_changed', (data: any) => {
-    log(`[leg] state: call=${data.call_id} leg=${data.leg_id} → ${data.state}`);
-    const call = activeCalls.get(data.call_id);
-    if (!call) return;
-    const leg = call.legs.get(data.leg_id);
-    if (leg) {
-      leg.state = data.state;
-      if (data.metadata) leg.metadata = data.metadata;
-    } else {
-      // Initial legs (provider/device) don't emit leg_added — create on first state change.
-      const legId: string = data.leg_id;
-      const type = legId.includes('-prov') ? 'sip-provider' : legId.includes('-dev') ? 'sip-device' : 'webrtc';
-      call.legs.set(data.leg_id, {
-        id: data.leg_id,
-        type,
-        state: data.state,
-        codec: null,
-        rtpPort: null,
-        remoteMedia: null,
-        metadata: data.metadata || {},
-      });
-    }
-  });
-
-  // WebRTC events from Rust — forward ICE candidates to browser via WebSocket.
-  onProxyEvent('webrtc_ice_candidate', (data: any) => {
-    // Find the browser's WebSocket by session ID and send the ICE candidate.
-    broadcastWs('webrtc-ice', {
-      sessionId: data.session_id,
-      candidate: { candidate: data.candidate, sdpMid: data.sdp_mid, sdpMLineIndex: data.sdp_mline_index },
-    });
-  });
-
-  onProxyEvent('webrtc_state', (data: any) => {
-    log(`[webrtc] session=${data.session_id?.slice(0, 8)} state=${data.state}`);
-  });
-
-  onProxyEvent('webrtc_track', (data: any) => {
-    log(`[webrtc] session=${data.session_id?.slice(0, 8)} track=${data.kind} codec=${data.codec}`);
-  });
-
-  onProxyEvent('webrtc_audio_rx', (data: any) => {
-    if (data.packet_count === 1 || data.packet_count === 50) {
-      log(`[webrtc] session=${data.session_id?.slice(0, 8)} browser audio rx #${data.packet_count}`);
-    }
-  });
-
-  // Voicemail events.
-  onProxyEvent('voicemail_started', (data: any) => {
-    log(`[voicemail] started for call ${data.call_id} caller=${data.caller_number}`);
-  });
-
-  onProxyEvent('recording_done', (data: any) => {
-    log(`[voicemail] recording done: ${data.file_path} (${data.duration_ms}ms) caller=${data.caller_number}`);
-    // Save voicemail metadata via VoiceboxManager.
-    voiceboxManager.addMessage?.('default', {
-      callerNumber: data.caller_number || 'Unknown',
-      callerName: null,
-      fileName: data.file_path,
-      durationMs: data.duration_ms,
-    });
-  });
-
-  onProxyEvent('voicemail_error', (data: any) => {
-    log(`[voicemail] error: ${data.error} call=${data.call_id}`);
-  });
-
-  // Send full config to Rust — this binds the SIP socket and starts registrations.
-  const configured = await configureProxyEngine({
-    proxy: appConfig.proxy,
-    providers: appConfig.providers,
-    devices: appConfig.devices,
-    routing: appConfig.routing,
+  registerProxyEventHandlers({
+    log,
+    statusStore,
+    voiceboxManager,
+    webRtcLinks,
+    getBrowserDeviceIds: getAllBrowserDeviceIds,
+    sendToBrowserDevice,
+    broadcast: broadcastWs,
+    onLinkWebRtcSession: requestWebRtcLink,
+    onCloseWebRtcSession: (sessionId) => {
+      void webrtcClose(sessionId);
+    },
   });
 
+  const configured = await configureRuntime(appConfig);
   if (!configured) {
     log('[FATAL] failed to configure proxy engine');
     process.exit(1);
   }
 
-  const providerList = appConfig.providers.map((p) => p.displayName).join(', ');
-  const deviceList = appConfig.devices.map((d) => d.displayName).join(', ');
+  const providerList = appConfig.providers.map((provider) => provider.displayName).join(', ');
+  const deviceList = appConfig.devices.map((device) => device.displayName).join(', ');
   log(`proxy engine started | LAN ${appConfig.proxy.lanIp}:${appConfig.proxy.lanPort} | providers: ${providerList} | devices: ${deviceList}`);
-
-  // Generate TTS audio (WAV files on disk, played by Rust audio_player).
-  try {
-    await initAnnouncement(log);
-
-    // Pre-generate prompts.
-    await promptCache.generateBeep('voicemail-beep', 1000, 500, 8000);
-    for (const vb of appConfig.voiceboxes ?? []) {
-      if (!vb.enabled) continue;
-      const promptId = `voicemail-greeting-${vb.id}`;
-      if (vb.greetingWavPath) {
-        await promptCache.loadWavPrompt(promptId, vb.greetingWavPath);
-      } else {
-        const text = vb.greetingText || 'The person you are trying to reach is not available. Please leave a message after the tone.';
-        await promptCache.generatePrompt(promptId, text, vb.greetingVoice || 'af_bella');
-      }
-    }
-    if (appConfig.ivr?.enabled) {
-      for (const menu of appConfig.ivr.menus) {
-        await promptCache.generatePrompt(`ivr-menu-${menu.id}`, menu.promptText, menu.promptVoice || 'af_bella');
-      }
-    }
-    log(`[startup] prompts cached: ${promptCache.listIds().join(', ') || 'none'}`);
-  } catch (e) {
-    log(`[tts] init failed: ${e}`);
-  }
 }
 
-// ---------------------------------------------------------------------------
-// Web UI
-// ---------------------------------------------------------------------------
-
-initWebUi(
+initWebUi({
+  port: appConfig.proxy.webUiPort,
   getStatus,
   log,
-  (number, deviceId, providerId) => {
-    // Outbound calls from dashboard — send make_call command to Rust.
+  onStartCall: (number, deviceId, providerId) => {
     log(`[dashboard] start call: ${number} device=${deviceId || 'any'} provider=${providerId || 'auto'}`);
-    // Fire-and-forget — the async result comes via events.
-    makeCall(number, deviceId, providerId).then((callId) => {
+    void makeCall(number, deviceId, providerId).then((callId) => {
       if (callId) {
         log(`[dashboard] call started: ${callId}`);
-        activeCalls.set(callId, {
-          id: callId,
-          direction: 'outbound',
-          callerNumber: null,
-          calleeNumber: number,
-          providerUsed: providerId || null,
-          state: 'setting-up',
-          startedAt: Date.now(),
-          legs: new Map(),
-        });
+        statusStore.noteDashboardCallStarted(callId, number, providerId);
       } else {
         log(`[dashboard] call failed for ${number}`);
       }
     });
-    // Return a temporary ID so the frontend doesn't show "failed" immediately.
+
     return { id: `pending-${Date.now()}` };
   },
-  (callId) => {
-    hangupCall(callId);
+  onHangupCall: (callId) => {
+    void hangupCall(callId);
     return true;
   },
-  () => {
-    // Config saved — reconfigure Rust engine.
-    try {
-      const fresh = loadConfig();
-      Object.assign(appConfig, fresh);
-
-      // Update shadow state.
-      for (const p of fresh.providers) {
-        if (!providerStatuses.has(p.id)) {
-          providerStatuses.set(p.id, {
-            id: p.id, displayName: p.displayName, registered: false, publicIp: null,
-          });
-        }
-      }
-      for (const d of fresh.devices) {
-        if (!deviceStatuses.has(d.id)) {
-          deviceStatuses.set(d.id, {
-            id: d.id, displayName: d.displayName, address: null, port: 0, connected: false, isBrowser: false,
-          });
-        }
-      }
-
-      // Re-send config to Rust.
-      configureProxyEngine({
-        proxy: fresh.proxy,
-        providers: fresh.providers,
-        devices: fresh.devices,
-        routing: fresh.routing,
-      }).then((ok) => {
-        if (ok) log('[config] reloaded — proxy engine reconfigured');
-        else log('[config] reload failed — proxy engine rejected config');
-      });
-    } catch (e: any) {
-      log(`[config] reload failed: ${e.message}`);
-    }
-  },
-  undefined, // callManager — legacy, replaced by Rust proxy-engine
-  voiceboxManager, // voiceboxManager
-  // WebRTC signaling → forwarded to Rust proxy-engine.
-  async (sessionId, sdp, ws) => {
+  onConfigSaved: reloadConfig,
+  voiceboxManager,
+  onWebRtcOffer: async (sessionId, sdp, ws) => {
     log(`[webrtc] offer from browser session=${sessionId.slice(0, 8)} sdp_type=${typeof sdp} sdp_len=${sdp?.length || 0}`);
     if (!sdp || typeof sdp !== 'string' || sdp.length < 10) {
       log(`[webrtc] WARNING: invalid SDP (type=${typeof sdp}), skipping offer`);
       return;
     }
+
     log(`[webrtc] sending offer to Rust (${sdp.length}b)...`);
     const result = await webrtcOffer(sessionId, sdp);
     log(`[webrtc] Rust result: ${JSON.stringify(result)?.slice(0, 200)}`);
     if (result?.sdp) {
       ws.send(JSON.stringify({ type: 'webrtc-answer', sessionId, sdp: result.sdp }));
       log(`[webrtc] answer sent to browser session=${sessionId.slice(0, 8)}`);
-    } else {
-      log(`[webrtc] ERROR: no answer SDP from Rust`);
+      return;
     }
+
+    log('[webrtc] ERROR: no answer SDP from Rust');
   },
-  async (sessionId, candidate) => {
+  onWebRtcIce: async (sessionId, candidate) => {
     await webrtcIce(sessionId, candidate);
   },
-  async (sessionId) => {
+  onWebRtcClose: async (sessionId) => {
+    webRtcLinks.removeSession(sessionId);
     await webrtcClose(sessionId);
   },
-  // onWebRtcAccept — browser has accepted a call, linking session to call.
-  (callId: string, sessionId: string) => {
+  onWebRtcAccept: (callId, sessionId) => {
     log(`[webrtc] accept: callId=${callId} sessionId=${sessionId.slice(0, 8)}`);
 
-    // Store bidirectional mapping.
-    webrtcSessionToCall.set(sessionId, callId);
-    webrtcCallToSession.set(callId, sessionId);
-
-    // Check if we already have media info for this call (provider answered first).
-    const media = pendingCallMedia.get(callId);
-    if (media) {
-      pendingCallMedia.delete(callId);
-      log(`[webrtc] linking session=${sessionId.slice(0, 8)} to call=${callId} media=${media.addr}:${media.port} pt=${media.sipPt}`);
-      webrtcLink(sessionId, callId, media.addr, media.port, media.sipPt).then((ok) => {
-        log(`[webrtc] link result: ${ok}`);
-      });
-    } else {
-      log(`[webrtc] session ${sessionId.slice(0, 8)} accepted, waiting for call_answered media info`);
+    const pendingMedia = webRtcLinks.acceptCall(callId, sessionId);
+    if (pendingMedia) {
+      requestWebRtcLink(callId, sessionId, pendingMedia);
+      return;
     }
+
+    log(`[webrtc] session ${sessionId.slice(0, 8)} accepted, waiting for call_answered media info`);
   },
-);
+});
 
-// ---------------------------------------------------------------------------
-// Start
-// ---------------------------------------------------------------------------
+void startProxyEngine();
 
-startProxyEngine();
+process.on('SIGINT', () => {
+  log('SIGINT, exiting');
+  shutdownProxyEngine();
+  process.exit(0);
+});
 
-process.on('SIGINT', () => { log('SIGINT, exiting'); shutdownProxyEngine(); process.exit(0); });
-process.on('SIGTERM', () => { log('SIGTERM, exiting'); shutdownProxyEngine(); process.exit(0); });
+process.on('SIGTERM', () => {
+  log('SIGTERM, exiting');
+  shutdownProxyEngine();
+  process.exit(0);
+});

ts/voicebox.ts

@@ -29,12 +29,14 @@ export interface IVoiceboxConfig {
   greetingVoice?: string;
   /** Path to uploaded WAV greeting (overrides TTS). */
   greetingWavPath?: string;
-  /** Seconds to wait before routing to voicemail (default 25). */
-  noAnswerTimeoutSec: number;
-  /** Maximum recording duration in seconds (default 120). */
-  maxRecordingSec: number;
-  /** Maximum stored messages per box (default 50). */
-  maxMessages: number;
+  /** Seconds to wait before routing to voicemail. Defaults to 25 when
+   *  absent — both the config loader and `VoiceboxManager.init` apply
+   *  the default via `??=`. */
+  noAnswerTimeoutSec?: number;
+  /** Maximum recording duration in seconds. Defaults to 120. */
+  maxRecordingSec?: number;
+  /** Maximum stored messages per box. Defaults to 50. */
+  maxMessages?: number;
 }
 
 export interface IVoicemailMessage {
@@ -148,6 +150,35 @@ export class VoiceboxManager {
   // Message CRUD
   // -------------------------------------------------------------------------
 
+  /**
+   * Convenience wrapper around `saveMessage` — used by the `recording_done`
+   * event handler, which has a raw recording path + caller info and needs
+   * to persist metadata. Generates `id`, sets `timestamp = now`, defaults
+   * `heard = false`, and normalizes `fileName` to a basename (the WAV is
+   * expected to already live in the box's directory).
+   */
+  addMessage(
+    boxId: string,
+    info: {
+      callerNumber: string;
+      callerName?: string | null;
+      fileName: string;
+      durationMs: number;
+    },
+  ): void {
+    const msg: IVoicemailMessage = {
+      id: crypto.randomUUID(),
+      boxId,
+      callerNumber: info.callerNumber,
+      callerName: info.callerName ?? undefined,
+      timestamp: Date.now(),
+      durationMs: info.durationMs,
+      fileName: path.basename(info.fileName),
+      heard: false,
+    };
+    this.saveMessage(msg);
+  }
+
   /**
    * Save a new voicemail message.
    * The WAV file should already exist at the expected path.

ts/webrtcbridge.ts

@@ -5,8 +5,8 @@
  * - Browser device registration/unregistration via WebSocket
  * - WS → deviceId mapping
  *
- * All WebRTC media logic (PeerConnection, RTP, transcoding) lives in
- * ts/call/webrtc-leg.ts and is managed by the CallManager.
+ * All WebRTC media logic (PeerConnection, RTP, transcoding, mixer wiring)
+ * lives in the Rust proxy-engine. This module only tracks browser sessions.
  */
 
 import { WebSocket } from 'ws';
@@ -39,7 +39,7 @@ export function initWebRtcSignaling(cfg: IWebRtcSignalingConfig): void {
 
 /**
  * Handle a WebRTC signaling message from a browser client.
- * Only handles registration; offer/ice/hangup are routed through CallManager.
+ * Only handles registration; offer/ice/hangup are routed through frontend.ts.
  */
 export function handleWebRtcSignaling(
   ws: WebSocket,
@@ -51,7 +51,7 @@ export function handleWebRtcSignaling(
     handleRegister(ws, message.sessionId!, message.userAgent, message._remoteIp);
   }
   // Other webrtc-* types (offer, ice, hangup, accept) are handled
-  // by the CallManager via frontend.ts WebSocket handler.
+  // by the frontend.ts WebSocket handler and forwarded to Rust.
 }
 
 /**
@@ -64,13 +64,6 @@ export function sendToBrowserDevice(deviceId: string, data: unknown): boolean {
   return true;
 }
 
-/**
- * Get the WebSocket for a browser device (used by CallManager to create WebRtcLegs).
- */
-export function getBrowserDeviceWs(deviceId: string): WebSocket | null {
-  return deviceIdToWs.get(deviceId) ?? null;
-}
-
 /**
  * Get all registered browser device IDs.
  */

ts_web/00_commitinfo_data.ts

@@ -3,6 +3,6 @@
  */
 export const commitinfo = {
   name: 'siprouter',
-  version: '1.19.2',
+  version: '1.25.0',
   description: 'undefined'
 }

ts_web/elements/sipproxy-devices.ts

@@ -41,11 +41,10 @@ export class SipproxyDevices extends DeesElement {
             },
           },
           {
-            key: 'contact',
+            key: 'address',
             header: 'Contact',
             renderer: (_val: any, row: any) => {
-              const c = row.contact;
-              const text = c ? (c.port ? `${c.address}:${c.port}` : c.address) : '--';
+              const text = row.address ? (row.port ? `${row.address}:${row.port}` : row.address) : '--';
               return html`<span style="font-family:'JetBrains Mono',monospace;font-size:.75rem">${text}</span>`;
             },
           },

ts_web/elements/sipproxy-view-overview.ts

@@ -186,11 +186,10 @@ export class SipproxyViewOverview extends DeesElement {
             },
           },
           {
-            key: 'contact',
+            key: 'address',
             header: 'Contact',
             renderer: (_val: any, row: any) => {
-              const c = row.contact;
-              const text = c ? (c.port ? `${c.address}:${c.port}` : c.address) : '--';
+              const text = row.address ? (row.port ? `${row.address}:${row.port}` : row.address) : '--';
               return html`<span style="font-family:'JetBrains Mono',monospace;font-size:.75rem">${text}</span>`;
             },
           },

ts_web/elements/sipproxy-view-providers.ts

@@ -164,173 +164,346 @@ export class SipproxyViewProviders extends DeesElement {
         iconName: 'lucide:plus',
         type: ['header'] as any,
         actionFunc: async () => {
-          await this.openAddModal();
-        },
-      },
-      {
-        name: 'Add Sipgate',
-        iconName: 'lucide:phone',
-        type: ['header'] as any,
-        actionFunc: async () => {
-          await this.openAddModal(PROVIDER_TEMPLATES.sipgate, 'Sipgate');
-        },
-      },
-      {
-        name: 'Add O2/Alice',
-        iconName: 'lucide:phone',
-        type: ['header'] as any,
-        actionFunc: async () => {
-          await this.openAddModal(PROVIDER_TEMPLATES.o2, 'O2/Alice');
+          await this.openAddStepper();
         },
       },
     ];
   }
 
-  // ---- add provider modal --------------------------------------------------
+  // ---- add provider stepper ------------------------------------------------
 
-  private async openAddModal(
-    template?: typeof PROVIDER_TEMPLATES.sipgate,
-    templateName?: string,
-  ) {
-    const { DeesModal } = await import('@design.estate/dees-catalog');
+  private async openAddStepper() {
+    const { DeesStepper } = await import('@design.estate/dees-catalog');
+    type TDeesStepper = InstanceType<typeof DeesStepper>;
+    // IStep / menuOptions types: we keep content typing loose (`any[]`) to
+    // avoid having to import tsclass IMenuItem just for one parameter annotation.
 
-    const formData = {
-      displayName: templateName || '',
-      domain: template?.domain || '',
-      outboundProxyAddress: template?.outboundProxy?.address || '',
-      outboundProxyPort: String(template?.outboundProxy?.port ?? 5060),
+    type TProviderType = 'Custom' | 'Sipgate' | 'O2/Alice';
+    interface IAccumulator {
+      providerType: TProviderType;
+      displayName: string;
+      domain: string;
+      outboundProxyAddress: string;
+      outboundProxyPort: string;
+      username: string;
+      password: string;
+      // Advanced — exposed in step 4
+      registerIntervalSec: string;
+      codecs: string;
+      earlyMediaSilence: boolean;
+    }
+
+    const accumulator: IAccumulator = {
+      providerType: 'Custom',
+      displayName: '',
+      domain: '',
+      outboundProxyAddress: '',
+      outboundProxyPort: '5060',
       username: '',
       password: '',
-      registerIntervalSec: String(template?.registerIntervalSec ?? 300),
-      codecs: template?.codecs ? template.codecs.join(', ') : '9, 0, 8, 101',
-      earlyMediaSilence: template?.quirks?.earlyMediaSilence ?? false,
+      registerIntervalSec: '300',
+      codecs: '9, 0, 8, 101',
+      earlyMediaSilence: false,
     };
 
-    const heading = template
-      ? `Add ${templateName} Provider`
-      : 'Add Provider';
+    // Snapshot the currently-selected step's form (if any) into accumulator.
+    const snapshotActiveForm = async (stepper: TDeesStepper) => {
+      const form = stepper.activeForm;
+      if (!form) return;
+      const data: Record<string, any> = await form.collectFormData();
+      Object.assign(accumulator, data);
+    };
 
-    await DeesModal.createAndShow({
-      heading,
-      width: 'small',
-      showCloseButton: true,
+    // Overwrite template-owned fields. Keep user-owned fields (username,
+    // password) untouched. displayName is replaced only when empty or still
+    // holds a branded auto-fill.
+    const applyTemplate = (type: TProviderType) => {
+      const tpl =
+        type === 'Sipgate' ? PROVIDER_TEMPLATES.sipgate
+        : type === 'O2/Alice' ? PROVIDER_TEMPLATES.o2
+        : null;
+      if (!tpl) return;
+      accumulator.domain = tpl.domain;
+      accumulator.outboundProxyAddress = tpl.outboundProxy.address;
+      accumulator.outboundProxyPort = String(tpl.outboundProxy.port);
+      accumulator.registerIntervalSec = String(tpl.registerIntervalSec);
+      accumulator.codecs = tpl.codecs.join(', ');
+      accumulator.earlyMediaSilence = tpl.quirks.earlyMediaSilence;
+      if (
+        !accumulator.displayName ||
+        accumulator.displayName === 'Sipgate' ||
+        accumulator.displayName === 'O2/Alice'
+      ) {
+        accumulator.displayName = type;
+      }
+    };
+
+    // --- Step builders (called after step 1 so accumulator is populated) ---
+
+    const buildConnectionStep = (): any => ({
+      title: 'Connection',
       content: html`
-        <div style="display:flex;flex-direction:column;gap:12px;padding:4px 0;">
+        <dees-form>
           <dees-input-text
             .key=${'displayName'}
             .label=${'Display Name'}
-            .value=${formData.displayName}
-            @input=${(e: Event) => { formData.displayName = (e.target as any).value; }}
+            .value=${accumulator.displayName}
+            .required=${true}
           ></dees-input-text>
           <dees-input-text
             .key=${'domain'}
             .label=${'Domain'}
-            .value=${formData.domain}
-            @input=${(e: Event) => { formData.domain = (e.target as any).value; }}
+            .value=${accumulator.domain}
+            .required=${true}
           ></dees-input-text>
           <dees-input-text
             .key=${'outboundProxyAddress'}
             .label=${'Outbound Proxy Address'}
-            .value=${formData.outboundProxyAddress}
-            @input=${(e: Event) => { formData.outboundProxyAddress = (e.target as any).value; }}
+            .value=${accumulator.outboundProxyAddress}
           ></dees-input-text>
           <dees-input-text
             .key=${'outboundProxyPort'}
             .label=${'Outbound Proxy Port'}
-            .value=${formData.outboundProxyPort}
-            @input=${(e: Event) => { formData.outboundProxyPort = (e.target as any).value; }}
+            .value=${accumulator.outboundProxyPort}
           ></dees-input-text>
+        </dees-form>
+      `,
+      menuOptions: [
+        {
+          name: 'Continue',
+          iconName: 'lucide:arrow-right',
+          action: async (stepper: TDeesStepper) => {
+            await snapshotActiveForm(stepper);
+            stepper.goNext();
+          },
+        },
+      ],
+    });
+
+    const buildCredentialsStep = (): any => ({
+      title: 'Credentials',
+      content: html`
+        <dees-form>
           <dees-input-text
             .key=${'username'}
             .label=${'Username / Auth ID'}
-            .value=${formData.username}
-            @input=${(e: Event) => { formData.username = (e.target as any).value; }}
+            .value=${accumulator.username}
+            .required=${true}
           ></dees-input-text>
           <dees-input-text
             .key=${'password'}
             .label=${'Password'}
             .isPasswordBool=${true}
-            .value=${formData.password}
-            @input=${(e: Event) => { formData.password = (e.target as any).value; }}
+            .value=${accumulator.password}
+            .required=${true}
           ></dees-input-text>
+        </dees-form>
+      `,
+      menuOptions: [
+        {
+          name: 'Continue',
+          iconName: 'lucide:arrow-right',
+          action: async (stepper: TDeesStepper) => {
+            await snapshotActiveForm(stepper);
+            stepper.goNext();
+          },
+        },
+      ],
+    });
+
+    const buildAdvancedStep = (): any => ({
+      title: 'Advanced',
+      content: html`
+        <dees-form>
           <dees-input-text
             .key=${'registerIntervalSec'}
             .label=${'Register Interval (sec)'}
-            .value=${formData.registerIntervalSec}
-            @input=${(e: Event) => { formData.registerIntervalSec = (e.target as any).value; }}
+            .value=${accumulator.registerIntervalSec}
           ></dees-input-text>
           <dees-input-text
             .key=${'codecs'}
             .label=${'Codecs (comma-separated payload types)'}
-            .value=${formData.codecs}
-            @input=${(e: Event) => { formData.codecs = (e.target as any).value; }}
+            .value=${accumulator.codecs}
           ></dees-input-text>
           <dees-input-checkbox
             .key=${'earlyMediaSilence'}
             .label=${'Early Media Silence (quirk)'}
-            .value=${formData.earlyMediaSilence}
-            @newValue=${(e: CustomEvent) => { formData.earlyMediaSilence = e.detail; }}
+            .value=${accumulator.earlyMediaSilence}
           ></dees-input-checkbox>
-        </div>
+        </dees-form>
       `,
       menuOptions: [
         {
-          name: 'Cancel',
-          iconName: 'lucide:x',
-          action: async (modalRef: any) => {
-            modalRef.destroy();
+          name: 'Continue',
+          iconName: 'lucide:arrow-right',
+          action: async (stepper: TDeesStepper) => {
+            await snapshotActiveForm(stepper);
+            // Rebuild the review step so its rendering reflects the latest
+            // accumulator values (the review step captures values at build time).
+            stepper.steps = [...stepper.steps.slice(0, 4), buildReviewStep()];
+            await (stepper as any).updateComplete;
+            stepper.goNext();
           },
         },
-        {
-          name: 'Create',
-          iconName: 'lucide:check',
-          action: async (modalRef: any) => {
-            if (!formData.displayName.trim() || !formData.domain.trim()) {
-              deesCatalog.DeesToast.error('Display name and domain are required');
-              return;
-            }
-            try {
-              const providerId = slugify(formData.displayName);
-              const codecs = formData.codecs
+      ],
+    });
+
+    const buildReviewStep = (): any => {
+      const resolvedId = slugify(accumulator.displayName);
+      return {
+        title: 'Review & Create',
+        content: html`
+          <dees-panel>
+            <div
+              style="display:grid;grid-template-columns:auto 1fr;gap:6px 16px;font-size:.85rem;padding:8px 4px;"
+            >
+              <div style="color:#94a3b8;">Type</div>
+              <div>${accumulator.providerType}</div>
+              <div style="color:#94a3b8;">Display Name</div>
+              <div>${accumulator.displayName}</div>
+              <div style="color:#94a3b8;">ID</div>
+              <div style="font-family:'JetBrains Mono',monospace;">${resolvedId}</div>
+              <div style="color:#94a3b8;">Domain</div>
+              <div>${accumulator.domain}</div>
+              <div style="color:#94a3b8;">Outbound Proxy</div>
+              <div>
+                ${accumulator.outboundProxyAddress || accumulator.domain}:${accumulator.outboundProxyPort}
+              </div>
+              <div style="color:#94a3b8;">Username</div>
+              <div>${accumulator.username}</div>
+              <div style="color:#94a3b8;">Password</div>
+              <div>${'*'.repeat(Math.min(accumulator.password.length, 12))}</div>
+              <div style="color:#94a3b8;">Register Interval</div>
+              <div>${accumulator.registerIntervalSec}s</div>
+              <div style="color:#94a3b8;">Codecs</div>
+              <div>${accumulator.codecs}</div>
+              <div style="color:#94a3b8;">Early-Media Silence</div>
+              <div>${accumulator.earlyMediaSilence ? 'yes' : 'no'}</div>
+            </div>
+          </dees-panel>
+        `,
+        menuOptions: [
+          {
+            name: 'Create Provider',
+            iconName: 'lucide:check',
+            action: async (stepper: TDeesStepper) => {
+              // Collision-resolve id against current state.
+              const existing = (this.appData.providers || []).map((p) => p.id);
+              let uniqueId = resolvedId;
+              let suffix = 2;
+              while (existing.includes(uniqueId)) {
+                uniqueId = `${resolvedId}-${suffix++}`;
+              }
+
+              const parsedCodecs = accumulator.codecs
                 .split(',')
                 .map((s: string) => parseInt(s.trim(), 10))
                 .filter((n: number) => !isNaN(n));
 
               const newProvider: any = {
-                id: providerId,
-                displayName: formData.displayName.trim(),
-                domain: formData.domain.trim(),
+                id: uniqueId,
+                displayName: accumulator.displayName.trim(),
+                domain: accumulator.domain.trim(),
                 outboundProxy: {
-                  address: formData.outboundProxyAddress.trim() || formData.domain.trim(),
-                  port: parseInt(formData.outboundProxyPort, 10) || 5060,
+                  address:
+                    accumulator.outboundProxyAddress.trim() || accumulator.domain.trim(),
+                  port: parseInt(accumulator.outboundProxyPort, 10) || 5060,
                 },
-                username: formData.username.trim(),
-                password: formData.password,
-                registerIntervalSec: parseInt(formData.registerIntervalSec, 10) || 300,
-                codecs,
+                username: accumulator.username.trim(),
+                password: accumulator.password,
+                registerIntervalSec: parseInt(accumulator.registerIntervalSec, 10) || 300,
+                codecs: parsedCodecs.length ? parsedCodecs : [9, 0, 8, 101],
                 quirks: {
-                  earlyMediaSilence: formData.earlyMediaSilence,
+                  earlyMediaSilence: accumulator.earlyMediaSilence,
                 },
               };
 
-              const result = await appState.apiSaveConfig({
-                addProvider: newProvider,
-              });
-
-              if (result.ok) {
-                modalRef.destroy();
-                deesCatalog.DeesToast.success(`Provider "${formData.displayName}" created`);
-              } else {
-                deesCatalog.DeesToast.error('Failed to save provider');
+              try {
+                const result = await appState.apiSaveConfig({
+                  addProvider: newProvider,
+                });
+                if (result.ok) {
+                  await stepper.destroy();
+                  deesCatalog.DeesToast.success(
+                    `Provider "${newProvider.displayName}" created`,
+                  );
+                } else {
+                  deesCatalog.DeesToast.error('Failed to save provider');
+                }
+              } catch (err: any) {
+                console.error('Failed to create provider:', err);
+                deesCatalog.DeesToast.error('Failed to create provider');
+              }
+            },
+          },
+        ],
+      };
+    };
+
+    // --- Step 1: Provider Type ------------------------------------------------
+    //
+    // Note: `DeesStepper.createAndShow()` dismisses on backdrop click; a user
+    // mid-form could lose work. Acceptable for v1 — revisit if users complain.
+
+    const typeOptions: { option: string; key: TProviderType }[] = [
+      { option: 'Custom', key: 'Custom' },
+      { option: 'Sipgate', key: 'Sipgate' },
+      { option: 'O2 / Alice', key: 'O2/Alice' },
+    ];
+    const currentTypeOption =
+      typeOptions.find((o) => o.key === accumulator.providerType) || null;
+
+    const typeStep: any = {
+      title: 'Choose provider type',
+      content: html`
+        <dees-form>
+          <dees-input-dropdown
+            .key=${'providerType'}
+            .label=${'Provider Type'}
+            .options=${typeOptions}
+            .selectedOption=${currentTypeOption}
+            .enableSearch=${false}
+            .required=${true}
+          ></dees-input-dropdown>
+        </dees-form>
+      `,
+      menuOptions: [
+        {
+          name: 'Continue',
+          iconName: 'lucide:arrow-right',
+          action: async (stepper: TDeesStepper) => {
+            // `dees-input-dropdown.value` is an object `{option, key, payload?}`,
+            // not a plain string — extract the `key` directly instead of using
+            // the generic `snapshotActiveForm` helper (which would clobber
+            // `accumulator.providerType`'s string type via Object.assign).
+            const form = stepper.activeForm;
+            if (form) {
+              const data: Record<string, any> = await form.collectFormData();
+              const selected = data.providerType;
+              if (selected && typeof selected === 'object' && selected.key) {
+                accumulator.providerType = selected.key as TProviderType;
               }
-            } catch (err: any) {
-              console.error('Failed to create provider:', err);
-              deesCatalog.DeesToast.error('Failed to create provider');
             }
+            if (!accumulator.providerType) {
+              accumulator.providerType = 'Custom';
+            }
+            applyTemplate(accumulator.providerType);
+            // (Re)build steps 2-5 with current accumulator values.
+            stepper.steps = [
+              typeStep,
+              buildConnectionStep(),
+              buildCredentialsStep(),
+              buildAdvancedStep(),
+              buildReviewStep(),
+            ];
+            await (stepper as any).updateComplete;
+            stepper.goNext();
           },
         },
       ],
-    });
+    };
+
+    await DeesStepper.createAndShow({ steps: [typeStep] });
   }
 
   // ---- edit provider modal -------------------------------------------------

ts_web/state/appstate.ts

@@ -2,72 +2,12 @@
  * Application state — receives live updates from the proxy via WebSocket.
  */
 
-export interface IProviderStatus {
-  id: string;
-  displayName: string;
-  registered: boolean;
-  publicIp: string | null;
-}
+import type { IContact } from '../../ts/config.ts';
+import type { ICallHistoryEntry, ICallStatus, IDeviceStatus, IProviderStatus } from '../../ts/shared/status.ts';
 
-export interface IDeviceStatus {
-  id: string;
-  displayName: string;
-  contact: { address: string; port: number } | null;
-  aor: string;
-  connected: boolean;
-  isBrowser: boolean;
-}
-
-export interface ILegStatus {
-  id: string;
-  type: 'sip-device' | 'sip-provider' | 'webrtc' | 'tool';
-  state: string;
-  remoteMedia: { address: string; port: number } | null;
-  rtpPort: number | null;
-  pktSent: number;
-  pktReceived: number;
-  codec: string | null;
-  transcoding: boolean;
-  metadata?: Record<string, unknown>;
-}
-
-export interface ICallStatus {
-  id: string;
-  state: string;
-  direction: 'inbound' | 'outbound' | 'internal';
-  callerNumber: string | null;
-  calleeNumber: string | null;
-  providerUsed: string | null;
-  createdAt: number;
-  duration: number;
-  legs: ILegStatus[];
-}
-
-export interface IHistoryLeg {
-  id: string;
-  type: string;
-  metadata: Record<string, unknown>;
-}
-
-export interface ICallHistoryEntry {
-  id: string;
-  direction: 'inbound' | 'outbound' | 'internal';
-  callerNumber: string | null;
-  calleeNumber: string | null;
-  providerUsed: string | null;
-  startedAt: number;
-  duration: number;
-  legs?: IHistoryLeg[];
-}
-
-export interface IContact {
-  id: string;
-  name: string;
-  number: string;
-  company?: string;
-  notes?: string;
-  starred?: boolean;
-}
+export type { IContact };
+export type { ICallHistoryEntry, ICallStatus, IDeviceStatus, IProviderStatus };
+export type { ILegStatus } from '../../ts/shared/status.ts';
 
 export interface IAppState {
   connected: boolean;