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feat(readme): expand documentation for voicemail, IVR, audio engine, and API capabilities

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Jürgen Kunz
2026-04-10 17:25:34 +00:00
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changelog.md
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# Changelog # Changelog
## 2026-04-10 - 1.18.0 - feat(readme)
expand documentation for voicemail, IVR, audio engine, and API capabilities
- Updates the feature overview to document voicemail, IVR menus, call recording, enhanced TTS, and the 48kHz float audio engine
- Refreshes the architecture section to describe the TypeScript control plane, Rust proxy-engine data plane, and JSON-over-stdio IPC
- Clarifies REST API and WebSocket coverage with voicemail endpoints, incoming call events, and refined endpoint descriptions
## 2026-04-10 - 1.17.2 - fix(proxy-engine) ## 2026-04-10 - 1.17.2 - fix(proxy-engine)
use negotiated SDP payload types when wiring SIP legs and enable default nnnoiseless features for telephony denoising use negotiated SDP payload types when wiring SIP legs and enable default nnnoiseless features for telephony denoising

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readme.md
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# @serve.zone/siprouter # @serve.zone/siprouter
A production-grade **SIP B2BUA + WebRTC bridge** built with TypeScript and Rust. Routes calls between SIP providers, SIP hardware devices, and browser softphones — with real-time codec transcoding, ML noise suppression, neural TTS announcements, and a slick web dashboard. A production-grade **SIP B2BUA + WebRTC bridge** built with TypeScript and Rust. Routes calls between SIP providers, SIP hardware devices, and browser softphones — with real-time codec transcoding, ML noise suppression, neural TTS, voicemail, IVR menus, and a slick web dashboard.
## Issue Reporting and Security ## Issue Reporting and Security
@@ -12,14 +12,16 @@ For reporting bugs, issues, or security vulnerabilities, please visit [community
siprouter sits between your SIP trunk providers and your endpoints — hardware phones, ATAs, browser softphones — and handles **everything** in between: siprouter sits between your SIP trunk providers and your endpoints — hardware phones, ATAs, browser softphones — and handles **everything** in between:
- 📞 **SIP B2BUA** — Terminates and re-originates calls with full RFC 3261 dialog state management - 📞 **SIP B2BUA** — Terminates and re-originates calls with full RFC 3261 dialog state management, digest auth, and SDP negotiation
- 🌐 **WebRTC Bridge** — Browser-based softphone with bidirectional audio to the SIP network - 🌐 **WebRTC Bridge** — Browser-based softphone with bidirectional Opus audio to the SIP network
- 🎛️ **Multi-Provider Trunking** — Register with multiple SIP providers simultaneously (sipgate, easybell, o2, etc.) - 🎛️ **Multi-Provider Trunking** — Register with multiple SIP providers simultaneously (sipgate, easybell, etc.) with automatic failover
- 🔊 **Rust Codec Engine** — Real-time Opus ↔ G.722 ↔ PCMU ↔ PCMA transcoding in native Rust - 🎧 **48kHz f32 Audio Engine** — High-fidelity internal audio bus at 48kHz/32-bit float with native Opus float encode/decode, FFT-based resampling, and per-leg ML noise suppression
- 🤖 **ML Noise Suppression** — RNNoise denoiser with per-direction state (to SIP / to browser) - 🔀 **N-Leg Mix-Minus Mixer** — Conference-grade mixing with dynamic leg add/remove, transfer, and per-source audio separation
- 🗣️ **Neural TTS** — Kokoro-powered "connecting your call" announcements, pre-encoded for instant playback - 📧 **Voicemail** — Configurable voicemail boxes with TTS greetings, recording, and web playback
- 🔀 **Hub Model Calls** — N-leg calls with dynamic add/remove, transfer, and RTP fan-out - 🔢 **IVR Menus** — DTMF-navigable interactive voice response with nested menus, routing actions, and custom prompts
- 🖥️ **Web Dashboard** — Real-time SPA with live call monitoring, browser phone, contact management, provider config - 🗣️ **Neural TTS** — Kokoro-powered announcements and greetings with 25+ voice presets, backed by espeak-ng fallback
- 🎙️ **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
--- ---
@@ -35,32 +37,38 @@ siprouter sits between your SIP trunk providers and your endpoints — hardware
┌──────────────────────────────────────┐ ┌──────────────────────────────────────┐
│ siprouter │ │ siprouter │
│ │ │ │
│ ┌──────────┐ ┌──────────────────┐ │ │ TypeScript Control Plane │
│ │ Call Hub │ │ Rust Transcoder │ │ │ ┌────────────────────────────────┐ │
│ │ N legs │──│ Opus/G.722/PCM │ │ │ │ Config · WebRTC Signaling │ │
│ │ fan-out │ │ + RNNoise │ │ │ │ REST API · Web Dashboard │ │
│ └────┬─────┘ └──────────────────┘ │ │ │ Voicebox Manager · TTS Cache │ │
│ │ │ │ └────────────┬───────────────────┘ │
│ ┌────┴─────┐ ┌──────────────────┐ │ │ JSON-over-stdio IPC │
│ │ SIP Stack│ │ Kokoro TTS │ │ │ ┌────────────┴───────────────────┐ │
│ │ Dialog SM│ │ (ONNX Runtime) │ │ │ │ Rust proxy-engine (data plane) │ │
│ └────┬─────┘ └──────────────────┘ │ │ │ │ │
│ │ │ │ │ SIP Stack · Dialog SM · Auth │ │
│ ┌────┴──────────────────────────┐ │ │ │ Call Manager · N-Leg Mixer │ │
│ │ Local Registrar + Provider │ │ │ │ 48kHz f32 Bus · RNNoise │ │
│ │ Registration Engine │ │ │ │ Codec Engine · RTP Port Pool │ │
│ └───────────────────────────────┘ │ │ │ WebRTC Engine · Kokoro TTS │ │
└──────────┬──────────────┬────────────┘ │ │ Voicemail · IVR · Recording │ │
│ │ │ └────┬──────────────────┬────────┘ │
┌──────┴──────┐ ┌─────┴──────┐ └───────┤──────────────────┤───────────┘
│ SIP Devices │ │ SIP Trunk │ │ │
│ (HT801, etc)│ │ Providers │ ┌──────┴──────┐ ┌──────┴──────┐
└─────────────┘ └────────────┘ │ SIP Devices │ │ SIP Trunk │
│ (HT801 etc) │ │ Providers │
└─────────────┘ └─────────────┘
``` ```
### The Hub Model ### 🧠 Key Design Decisions
Every call is a **hub** with N legs. Each leg is either a `SipLeg` (hardware device or provider) or a `WebRtcLeg` (browser). RTP flows through the hub — each leg's received audio is forwarded to all other legs, with codec transcoding handled transparently by the Rust engine. - **Hub Model** — Every call is a hub with N legs. Each leg is a `SipLeg` (device/provider) or `WebRtcLeg` (browser). Legs can be dynamically added, removed, or transferred without tearing down the call.
- **Rust Data Plane** — All SIP protocol handling, codec transcoding, mixing, and RTP I/O runs in native Rust for real-time performance. TypeScript handles config, signaling, REST API, and dashboard.
- **48kHz f32 Internal Bus** — Audio is processed at maximum quality internally. Encoding/decoding to wire format (G.722, PCMU, Opus) happens solely at the leg boundary.
- **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.
--- ---
@@ -70,15 +78,16 @@ Every call is a **hub** with N legs. Each leg is either a `SipLeg` (hardware dev
- **Node.js** ≥ 20 with `tsx` globally available - **Node.js** ≥ 20 with `tsx` globally available
- **pnpm** for package management - **pnpm** for package management
- **Rust** toolchain (for building the codec engine and TTS) - **Rust** toolchain (for building the proxy engine)
- **espeak-ng** (optional, for TTS fallback)
### Install & Build ### Install & Build
```bash ```bash
# Clone and install # Clone and install dependencies
pnpm install pnpm install
# Build the Rust binaries (opus-codec + tts-engine) # Build the Rust proxy-engine binary
pnpm run buildRust pnpm run buildRust
# Bundle the web frontend # Bundle the web frontend
@@ -87,57 +96,92 @@ pnpm run bundle
### Configuration ### Configuration
Create `.nogit/config.json` with your setup: Create `.nogit/config.json`:
```jsonc ```jsonc
{ {
"proxy": { "proxy": {
"lanIp": "192.168.1.100", // Your server's LAN IP "lanIp": "192.168.1.100", // Your server's LAN IP
"lanPort": 5070, // SIP signaling port "lanPort": 5070, // SIP signaling port
"rtpPortRange": [20000, 20200],// RTP relay port pool (even ports) "publicIpSeed": "stun.example.com", // STUN server for public IP discovery
"webUiPort": 3060 // Dashboard port "rtpPortRange": { "min": 20000, "max": 20200 }, // RTP port pool (even ports)
"webUiPort": 3060 // Dashboard + REST API port
}, },
"providers": [ "providers": [
{ {
"id": "my-trunk", "id": "my-trunk",
"name": "My SIP Provider", "displayName": "My SIP Provider",
"host": "sip.provider.com", "domain": "sip.provider.com",
"port": 5060, "outboundProxy": { "address": "sip.provider.com", "port": 5060 },
"username": "user", "username": "user",
"password": "pass", "password": "pass",
"codecs": ["G.722", "PCMA", "PCMU"], "codecs": [9, 0, 8, 101], // G.722, PCMU, PCMA, telephone-event
"registerExpiry": 3600 "registerIntervalSec": 300
} }
], ],
"devices": [ "devices": [
{ {
"id": "desk-phone", "id": "desk-phone",
"name": "Desk Phone", "displayName": "Desk Phone",
"type": "sip" "expectedAddress": "192.168.1.50",
"extension": "100"
} }
], ],
"routing": { "routing": {
"inbound": { "routes": [
"default": { "target": "all-devices", "ringBrowser": true } {
"id": "inbound-default",
"name": "Ring all devices",
"priority": 100,
"direction": "inbound",
"match": {},
"action": {
"targets": ["desk-phone"],
"ringBrowsers": true,
"voicemailBox": "main",
"noAnswerTimeout": 25
}
},
{
"id": "outbound-default",
"name": "Route via trunk",
"priority": 100,
"direction": "outbound",
"match": {},
"action": { "provider": "my-trunk" }
}
]
},
"voiceboxes": [
{
"id": "main",
"enabled": true,
"greetingText": "Please leave a message after the beep.",
"greetingVoice": "af_bella",
"noAnswerTimeoutSec": 25,
"maxRecordingSec": 120,
"maxMessages": 50
} }
} ],
"contacts": [
{ "id": "1", "name": "Alice", "number": "+491234567890", "starred": true }
]
} }
``` ```
### TTS Setup (Optional) ### TTS Setup (Optional)
For neural "connecting your call" announcements, download the Kokoro TTS model: For neural announcements and voicemail greetings, download the Kokoro TTS model:
```bash ```bash
mkdir -p .nogit/tts mkdir -p .nogit/tts
# Download the full-quality model (310MB) + voices (27MB)
curl -L -o .nogit/tts/kokoro-v1.0.onnx \ curl -L -o .nogit/tts/kokoro-v1.0.onnx \
https://github.com/mzdk100/kokoro/releases/download/V1.0/kokoro-v1.0.onnx https://github.com/mzdk100/kokoro/releases/download/V1.0/kokoro-v1.0.onnx
curl -L -o .nogit/tts/voices.bin \ curl -L -o .nogit/tts/voices.bin \
https://github.com/mzdk100/kokoro/releases/download/V1.0/voices.bin https://github.com/mzdk100/kokoro/releases/download/V1.0/voices.bin
``` ```
If the model files aren't present, the announcement feature is simply disabled — everything else works fine. Without the model files, TTS falls back to `espeak-ng`. Without either, announcements are skipped — everything else works fine.
### Run ### Run
@@ -145,7 +189,7 @@ If the model files aren't present, the announcement feature is simply disabled
pnpm start pnpm start
``` ```
The SIP proxy starts on the configured port and the web dashboard is available at `http://<your-ip>:3060`. The SIP proxy starts on the configured port and the web dashboard is available at `https://<your-ip>:3060`.
### HTTPS (Optional) ### HTTPS (Optional)
@@ -157,68 +201,91 @@ Place `cert.pem` and `key.pem` in `.nogit/` for TLS on the dashboard.
``` ```
siprouter/ siprouter/
├── ts/ # TypeScript source ├── ts/ # TypeScript control plane
│ ├── sipproxy.ts # Main entry — bootstraps everything │ ├── sipproxy.ts # Main entry — bootstraps everything
│ ├── config.ts # Config loader & validation │ ├── config.ts # Config loader & validation
│ ├── registrar.ts # Local SIP registrar for devices │ ├── proxybridge.ts # Rust proxy-engine IPC bridge (smartrust)
│ ├── providerstate.ts # Per-provider upstream registration engine │ ├── frontend.ts # Web dashboard HTTP/WS server + REST API
│ ├── frontend.ts # Web dashboard HTTP/WS server + REST API │ ├── webrtcbridge.ts # WebRTC signaling layer
│ ├── webrtcbridge.ts # WebRTC signaling layer │ ├── registrar.ts # Browser softphone registration
│ ├── opusbridge.ts # Rust IPC bridge (smartrust) │ ├── announcement.ts # TTS announcement generator (espeak-ng / Kokoro)
│ ├── codec.ts # High-level RTP transcoding interface │ ├── voicebox.ts # Voicemail box management
│ ├── announcement.ts # Neural TTS announcement generator │ └── call/
│ ├── sip/ # Zero-dependency SIP protocol library │ └── prompt-cache.ts # Named audio prompt WAV management
│ │ ├── message.ts # SIP message parser/builder/mutator │
│ │ ├── dialog.ts # RFC 3261 dialog state machine ├── ts_web/ # Web frontend (Lit-based SPA)
│ │ ├── helpers.ts # SDP builder, digest auth, codec registry │ ├── elements/ # Web components (9 dashboard views)
│ │ └── rewrite.ts # SIP URI + SDP body rewriting │ └── state/ # App state, WebRTC client, notifications
│ └── call/ # Hub-model call management │
│ ├── call-manager.ts # Central registry, factory, routing ├── rust/ # Rust workspace (the data plane)
│ ├── call.ts # Call hub — owns N legs, media fan-out
│ ├── sip-leg.ts # SIP device/provider connection
│ ├── webrtc-leg.ts # Browser WebRTC connection
│ └── rtp-port-pool.ts # UDP port allocation
├── ts_web/ # Web frontend (Lit-based SPA)
│ ├── elements/ # Web components (dashboard, phone, etc.)
│ └── state/ # App state, WebRTC client, notifications
├── rust/ # Rust workspace
│ └── crates/ │ └── crates/
│ ├── opus-codec/ # Real-time audio transcoder (Opus/G.722/PCM) │ ├── codec-lib/ # Audio codec library (Opus/G.722/PCMU/PCMA)
│ └── tts-engine/ # Kokoro neural TTS CLI │ ├── sip-proto/ # Zero-dependency SIP protocol library
├── html/ # Static HTML shell │ └── proxy-engine/ # Main binary — SIP engine + mixer + RTP
├── .nogit/ # Secrets, config, models (gitignored) │
└── dist_rust/ # Compiled Rust binaries (gitignored) ├── html/ # Static HTML shell
├── .nogit/ # Secrets, config, TTS models (gitignored)
└── dist_rust/ # Compiled Rust binary (gitignored)
``` ```
--- ---
## 🎧 Codec Engine (Rust) ## 🎧 Audio Engine (Rust)
The `opus-codec` binary handles all real-time audio processing via a JSON-over-stdio IPC protocol: The `proxy-engine` binary handles all real-time audio processing with a **48kHz f32 internal bus** — encoding and decoding happens only at leg boundaries.
| Codec | Payload Type | Sample Rate | Use Case | ### Supported Codecs
|-------|-------------|-------------|----------|
| **Opus** | 111 | 48 kHz | WebRTC browsers | | Codec | PT | Native Rate | Use Case |
| **G.722** | 9 | 16 kHz | HD SIP devices | |-------|:--:|:-----------:|----------|
| **Opus** | 111 | 48 kHz | WebRTC browsers (native float encode/decode — zero i16 quantization) |
| **G.722** | 9 | 16 kHz | HD SIP devices & providers |
| **PCMU** (G.711 µ-law) | 0 | 8 kHz | Legacy SIP | | **PCMU** (G.711 µ-law) | 0 | 8 kHz | Legacy SIP |
| **PCMA** (G.711 A-law) | 8 | 8 kHz | Legacy SIP | | **PCMA** (G.711 A-law) | 8 | 8 kHz | Legacy SIP |
**Features:** ### Audio Pipeline
- Per-call isolated codec sessions (no cross-call state corruption)
- FFT-based sample rate conversion via `rubato` ```
- **RNNoise ML noise suppression** with per-direction state — denoises audio flowing to SIP separately from audio flowing to the browser Inbound: Wire RTP → Decode → Resample to 48kHz → Denoise (RNNoise) → Mix Bus
- Raw PCM encoding for TTS frame processing Outbound: Mix Bus → Mix-Minus → Resample to codec rate → Encode → Wire RTP
```
- **FFT-based resampling** via `rubato` — high-quality sinc interpolation with cached resampler state for seamless inter-frame continuity
- **ML noise suppression** via `nnnoiseless` (RNNoise) — per-leg inbound denoising with SIMD acceleration (AVX/SSE). Skipped for WebRTC legs (browsers already denoise via getUserMedia)
- **Mix-minus mixing** — each participant hears everyone except themselves, accumulated in f64 precision
- **In-tick packet reorder** — inbound RTP packets are sorted by sequence number before decoding, protecting G.722 ADPCM state from out-of-order delivery
- **RFC 3550 compliant header parsing** — properly handles CSRC lists and header extensions
--- ---
## 🗣️ Neural TTS (Rust) ## 🗣️ Neural TTS
The `tts-engine` binary uses [Kokoro TTS](https://github.com/mzdk100/kokoro) (82M parameter neural model) to synthesize announcements at startup: 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:
- **24 kHz, 16-bit mono** output - **24 kHz, 16-bit mono** output
- **25+ voice presets** — American/British, male/female (e.g., `af_bella`, `am_adam`, `bf_emma`, `bm_george`) - **25+ voice presets** — American/British, male/female (e.g., `af_bella`, `am_adam`, `bf_emma`, `bm_george`)
- **~800ms** synthesis time for a 3-second announcement - **~800ms** synthesis time for a 3-second phrase
- Pre-encoded to G.722 + Opus for zero-latency RTP playback during call setup - Lazy-loaded on first use — no startup cost if TTS is unused
- Falls back to `espeak-ng` if the ONNX model is not available
---
## 📧 Voicemail
- Configurable voicemail boxes with custom TTS greetings
- Automatic routing on no-answer timeout
- Recording with configurable max duration and message count
- Web dashboard playback and management
- WAV storage in `.nogit/voicemail/`
---
## 🔢 IVR (Interactive Voice Response)
- DTMF-navigable menus with configurable entries
- Actions: route to extension, route to voicemail, transfer, submenu, hangup, repeat prompt
- Custom TTS prompts per menu
- Nested menu support
--- ---
@@ -228,33 +295,42 @@ The `tts-engine` binary uses [Kokoro TTS](https://github.com/mzdk100/kokoro) (82
| View | Description | | View | Description |
|------|-------------| |------|-------------|
| **Overview** | Stats tiles — uptime, providers, devices, active calls | | 📊 **Overview** | Stats tiles — uptime, providers, devices, active calls |
| **Calls** | Active calls with leg details, codec info, packet counters. Add/remove legs, transfer, hangup | | 📞 **Calls** | Active calls with leg details, codec info, add/remove legs, transfer, hangup |
| **Phone** | Browser softphone — mic/speaker selection, audio meters, dial pad, incoming call popup | | ☎️ **Phone** | Browser softphone — mic/speaker selection, audio meters, dial pad, incoming call popup |
| **Contacts** | Contact management with click-to-call | | 🔀 **Routes** | Routing rule management — match/action model with priority |
| **Providers** | SIP trunk config with registration status | | 📧 **Voicemail** | Voicemail box management + message playback |
| **Log** | Live streaming log viewer | | 🔢 **IVR** | IVR menu builder — DTMF entries, TTS prompts, nested menus |
| 👤 **Contacts** | Contact management with click-to-call |
| 🔌 **Providers** | SIP trunk configuration and registration status |
| 📋 **Log** | Live streaming log viewer |
### REST API ### REST API
| Endpoint | Method | Description | | Endpoint | Method | Description |
|----------|--------|-------------| |----------|--------|-------------|
| `/api/status` | GET | Full system status (providers, devices, calls) | | `/api/status` | GET | Full system status (providers, devices, calls, history) |
| `/api/call` | POST | Originate a call | | `/api/call` | POST | Originate a call |
| `/api/hangup` | POST | Hang up a call | | `/api/hangup` | POST | Hang up a call |
| `/api/call/:id/addleg` | POST | Add a leg to an active call | | `/api/call/:id/addleg` | POST | Add a device leg to an active call |
| `/api/call/:id/addexternal` | POST | Add an external participant | | `/api/call/:id/addexternal` | POST | Add an external participant via provider |
| `/api/call/:id/removeleg` | POST | Remove a leg from a call | | `/api/call/:id/removeleg` | POST | Remove a leg from a call |
| `/api/transfer` | POST | Transfer a call | | `/api/transfer` | POST | Transfer a call |
| `/api/config` | GET/POST | Read or update configuration (hot-reload) | | `/api/config` | GET | Read current configuration |
| `/api/config` | POST | Update configuration (hot-reload) |
| `/api/voicemail/:box` | GET | List voicemail messages |
| `/api/voicemail/:box/:id` | DELETE | Delete a voicemail message |
| `/api/voicemail/:box/:id/audio` | GET | Stream voicemail audio |
### WebSocket Events ### WebSocket Events
Connect to `/ws` for real-time push: Connect to `/ws` for real-time push:
```jsonc ```jsonc
{ "type": "status", "data": { ... } } // Full status snapshot (1s interval) { "type": "status", "data": { ... } } // Full status snapshot (1s interval)
{ "type": "log", "data": { "message": "..." } } // Log lines in real-time { "type": "log", "data": { "message": "..." } } // Log lines in real-time
{ "type": "incoming_call", "data": { ... } } // Incoming call notification
{ "type": "call_ended", "data": { ... } } // Call ended notification
``` ```
--- ---
@@ -264,7 +340,7 @@ Connect to `/ws` for real-time push:
| Port | Protocol | Purpose | | Port | Protocol | Purpose |
|------|----------|---------| |------|----------|---------|
| 5070 (configurable) | UDP | SIP signaling | | 5070 (configurable) | UDP | SIP signaling |
| 20000–20200 (configurable) | UDP | RTP relay (even ports, per-call allocation) | | 20000–20200 (configurable) | UDP | RTP media (even ports, per-call allocation) |
| 3060 (configurable) | TCP | Web dashboard + WebSocket + REST API | | 3060 (configurable) | TCP | Web dashboard + WebSocket + REST API |
--- ---
@@ -275,23 +351,16 @@ Connect to `/ws` for real-time push:
# Start in dev mode # Start in dev mode
pnpm start pnpm start
# Build Rust crates # Build Rust proxy-engine
pnpm run buildRust pnpm run buildRust
# Bundle web frontend # Bundle web frontend
pnpm run bundle pnpm run bundle
# Restart background server (build + bundle + restart) # Build + bundle + restart background server
pnpm run restartBackground pnpm run restartBackground
``` ```
### Key Design Decisions
- **Hub Model** — Calls are N-leg hubs, not point-to-point. This enables multi-party, dynamic leg manipulation, and transfer without tearing down the call.
- **Zero-dependency SIP library** — `ts/sip/` is a pure data-level SIP stack (parse/build/mutate/serialize). No transport or timer logic — those live in the application layer.
- **Rust for the hot path** — Codec transcoding and noise suppression run in native Rust for real-time performance. TypeScript handles signaling and orchestration.
- **Per-session codec isolation** — Each call gets its own Opus/G.722 encoder/decoder state in the Rust process, preventing stateful codec prediction from leaking between concurrent calls.
--- ---
## License and Legal Information ## License and Legal Information

2
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@@ -3,6 +3,6 @@
*/ */
export const commitinfo = { export const commitinfo = {
name: 'siprouter', name: 'siprouter',
version: '1.17.2', version: '1.18.0',
description: 'undefined' description: 'undefined'
} }

2
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@@ -3,6 +3,6 @@
*/ */
export const commitinfo = { export const commitinfo = {
name: 'siprouter', name: 'siprouter',
version: '1.17.2', version: '1.18.0',
description: 'undefined' description: 'undefined'
} }