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@@ -1,5 +1,19 @@
# 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)
use negotiated SDP payload types when wiring SIP legs and enable default nnnoiseless features for telephony denoising
- Select the negotiated codec payload type from SDP answers instead of always using the first offered codec
- Preserve the device leg's preferred payload type from its own INVITE SDP when attaching it to the mixer
- Enable default nnnoiseless features in codec-lib and proxy-engine dependencies
## 2026-04-10 - 1.17.1 - fix(proxy-engine,codec-lib,sip-proto,ts)
preserve negotiated media details and improve RTP audio handling across call legs

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@@ -1,6 +1,6 @@
{
"name": "siprouter",
"version": "1.17.1",
"version": "1.18.0",
"private": true,
"type": "module",
"scripts": {

309
readme.md
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@@ -1,6 +1,6 @@
# @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
@@ -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:
- 📞 **SIP B2BUA** — Terminates and re-originates calls with full RFC 3261 dialog state management
- 🌐 **WebRTC Bridge** — Browser-based softphone with bidirectional audio to the SIP network
- 🎛️ **Multi-Provider Trunking** — Register with multiple SIP providers simultaneously (sipgate, easybell, o2, etc.)
- 🔊 **Rust Codec Engine** — Real-time Opus ↔ G.722 ↔ PCMU ↔ PCMA transcoding in native Rust
- 🤖 **ML Noise Suppression** — RNNoise denoiser with per-direction state (to SIP / to browser)
- 🗣️ **Neural TTS** — Kokoro-powered "connecting your call" announcements, pre-encoded for instant playback
- 🔀 **Hub Model Calls** — N-leg calls with dynamic add/remove, transfer, and RTP fan-out
- 🖥 **Web Dashboard** — Real-time SPA with live call monitoring, browser phone, contact management, provider config
- 📞 **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 Opus audio to the SIP network
- 🎛️ **Multi-Provider Trunking** — Register with multiple SIP providers simultaneously (sipgate, easybell, etc.) with automatic failover
- 🎧 **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
- 🔀 **N-Leg Mix-Minus Mixer** — Conference-grade mixing with dynamic leg add/remove, transfer, and per-source audio separation
- 📧 **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
- 🎙️ **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 │
│ │
┌──────────┐ ┌──────────────────┐
│ Call Hub │ │ Rust Transcoder │
│ │ N legs │──│ Opus/G.722/PCM │ │
│ │ fan-out │ │ + RNNoise │ │
└────┬─────┘ └──────────────────┘
┌────┴─────┐ ┌──────────────────┐
│ SIP Stack│ │ Kokoro TTS │
│ │ Dialog SM│ │ (ONNX Runtime) │ │
└────┬─────┘ └──────────────────┘
┌────┴──────────────────────────┐
│ │ Local Registrar + Provider
│ │ Registration Engine
└───────────────────────────────┘
└──────────┬──────────────┬────────────┘
────────────┐ ┌───────────
│ SIP Devices │ │ SIP Trunk
│ (HT801, etc)│ │ Providers │
└─────────────┘ └────────────┘
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 · RNNoise │ │
│ │ Codec Engine · RTP Port Pool │ │
│ WebRTC Engine · Kokoro TTS │
│ │ Voicemail · IVR · Recording │ │
│ └────┬──────────────────┬────────┘
───────┤──────────────────┤───────────┘
┌──────┴──────┐ ┌──────┴──────┐
│ 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
- **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
```bash
# Clone and install
# Clone and install dependencies
pnpm install
# Build the Rust binaries (opus-codec + tts-engine)
# Build the Rust proxy-engine binary
pnpm run buildRust
# Bundle the web frontend
@@ -87,57 +96,92 @@ pnpm run bundle
### Configuration
Create `.nogit/config.json` with your setup:
Create `.nogit/config.json`:
```jsonc
{
"proxy": {
"lanIp": "192.168.1.100", // Your server's LAN IP
"lanPort": 5070, // SIP signaling port
"rtpPortRange": [20000, 20200],// RTP relay port pool (even ports)
"webUiPort": 3060 // Dashboard port
"lanIp": "192.168.1.100", // Your server's LAN IP
"lanPort": 5070, // SIP signaling port
"publicIpSeed": "stun.example.com", // STUN server for public IP discovery
"rtpPortRange": { "min": 20000, "max": 20200 }, // RTP port pool (even ports)
"webUiPort": 3060 // Dashboard + REST API port
},
"providers": [
{
"id": "my-trunk",
"name": "My SIP Provider",
"host": "sip.provider.com",
"port": 5060,
"displayName": "My SIP Provider",
"domain": "sip.provider.com",
"outboundProxy": { "address": "sip.provider.com", "port": 5060 },
"username": "user",
"password": "pass",
"codecs": ["G.722", "PCMA", "PCMU"],
"registerExpiry": 3600
"codecs": [9, 0, 8, 101], // G.722, PCMU, PCMA, telephone-event
"registerIntervalSec": 300
}
],
"devices": [
{
"id": "desk-phone",
"name": "Desk Phone",
"type": "sip"
"displayName": "Desk Phone",
"expectedAddress": "192.168.1.50",
"extension": "100"
}
],
"routing": {
"inbound": {
"default": { "target": "all-devices", "ringBrowser": true }
"routes": [
{
"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)
For neural "connecting your call" announcements, download the Kokoro TTS model:
For neural announcements and voicemail greetings, download the Kokoro TTS model:
```bash
mkdir -p .nogit/tts
# Download the full-quality model (310MB) + voices (27MB)
curl -L -o .nogit/tts/kokoro-v1.0.onnx \
https://github.com/mzdk100/kokoro/releases/download/V1.0/kokoro-v1.0.onnx
curl -L -o .nogit/tts/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
@@ -145,7 +189,7 @@ If the model files aren't present, the announcement feature is simply disabled
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)
@@ -157,68 +201,91 @@ Place `cert.pem` and `key.pem` in `.nogit/` for TLS on the dashboard.
```
siprouter/
├── ts/ # TypeScript source
│ ├── sipproxy.ts # Main entry — bootstraps everything
│ ├── config.ts # Config loader & validation
│ ├── registrar.ts # Local SIP registrar for devices
│ ├── providerstate.ts # Per-provider upstream registration engine
│ ├── frontend.ts # Web dashboard HTTP/WS server + REST API
│ ├── webrtcbridge.ts # WebRTC signaling layer
│ ├── opusbridge.ts # Rust IPC bridge (smartrust)
│ ├── codec.ts # High-level RTP transcoding interface
── announcement.ts # Neural TTS announcement generator
├── sip/ # Zero-dependency SIP protocol library
│ ├── message.ts # SIP message parser/builder/mutator
├── dialog.ts # RFC 3261 dialog state machine
│ ├── helpers.ts # SDP builder, digest auth, codec registry
│ └── rewrite.ts # SIP URI + SDP body rewriting
└── call/ # Hub-model call management
│ ├── call-manager.ts # Central registry, factory, routing
│ ├── 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
├── ts/ # TypeScript control plane
│ ├── sipproxy.ts # Main entry — bootstraps everything
│ ├── config.ts # Config loader & validation
│ ├── proxybridge.ts # Rust proxy-engine IPC bridge (smartrust)
│ ├── 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
├── ts_web/ # Web frontend (Lit-based SPA)
│ ├── elements/ # Web components (9 dashboard views)
│ └── state/ # App state, WebRTC client, notifications
├── rust/ # Rust workspace (the data plane)
│ └── crates/
│ ├── opus-codec/ # Real-time audio transcoder (Opus/G.722/PCM)
── tts-engine/ # Kokoro neural TTS CLI
├── html/ # Static HTML shell
├── .nogit/ # Secrets, config, models (gitignored)
── dist_rust/ # Compiled Rust binaries (gitignored)
│ ├── codec-lib/ # Audio codec library (Opus/G.722/PCMU/PCMA)
── sip-proto/ # Zero-dependency SIP protocol library
│ └── proxy-engine/ # Main binary — SIP engine + mixer + RTP
── 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 |
|-------|-------------|-------------|----------|
| **Opus** | 111 | 48 kHz | WebRTC browsers |
| **G.722** | 9 | 16 kHz | HD SIP devices |
### Supported Codecs
| Codec | PT | Native Rate | Use Case |
|-------|:--:|:-----------:|----------|
| **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 |
| **PCMA** (G.711 A-law) | 8 | 8 kHz | Legacy SIP |
**Features:**
- 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
- Raw PCM encoding for TTS frame processing
### Audio Pipeline
```
Inbound: Wire RTP → Decode → Resample to 48kHz → Denoise (RNNoise) → Mix Bus
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
- **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
- Pre-encoded to G.722 + Opus for zero-latency RTP playback during call setup
- **~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
---
## 📧 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 |
|------|-------------|
| **Overview** | Stats tiles — uptime, providers, devices, active calls |
| **Calls** | Active calls with leg details, codec info, packet counters. Add/remove legs, transfer, hangup |
| **Phone** | Browser softphone — mic/speaker selection, audio meters, dial pad, incoming call popup |
| **Contacts** | Contact management with click-to-call |
| **Providers** | SIP trunk config with registration status |
| **Log** | Live streaming log viewer |
| 📊 **Overview** | Stats tiles — uptime, providers, devices, active calls |
| 📞 **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 |
| 🔀 **Routes** | Routing rule management — match/action model with priority |
| 📧 **Voicemail** | Voicemail box management + message playback |
| 🔢 **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
| 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/hangup` | POST | Hang up a call |
| `/api/call/:id/addleg` | POST | Add a leg to an active call |
| `/api/call/:id/addexternal` | POST | Add an external participant |
| `/api/call/:id/addleg` | POST | Add a device leg to an active call |
| `/api/call/:id/addexternal` | POST | Add an external participant via provider |
| `/api/call/:id/removeleg` | POST | Remove a leg from 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
Connect to `/ws` for real-time push:
```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": "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 |
|------|----------|---------|
| 5070 (configurable) | UDP | SIP signaling |
| 2000020200 (configurable) | UDP | RTP relay (even ports, per-call allocation) |
| 2000020200 (configurable) | UDP | RTP media (even ports, per-call allocation) |
| 3060 (configurable) | TCP | Web dashboard + WebSocket + REST API |
---
@@ -275,23 +351,16 @@ Connect to `/ws` for real-time push:
# Start in dev mode
pnpm start
# Build Rust crates
# Build Rust proxy-engine
pnpm run buildRust
# Bundle web frontend
pnpm run bundle
# Restart background server (build + bundle + restart)
# Build + bundle + restart background server
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

226
rust/Cargo.lock generated
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@@ -237,6 +237,17 @@ version = "1.1.2"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "1505bd5d3d116872e7271a6d4e16d81d0c8570876c8de68093a09ac269d8aac0"
[[package]]
name = "atty"
version = "0.2.14"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "d9b39be18770d11421cdb1b9947a45dd3f37e93092cbf377614828a319d5fee8"
dependencies = [
"hermit-abi",
"libc",
"winapi",
]
[[package]]
name = "audiopus"
version = "0.3.0-rc.0"
@@ -487,6 +498,31 @@ dependencies = [
"inout",
]
[[package]]
name = "clap"
version = "3.2.25"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "4ea181bf566f71cb9a5d17a59e1871af638180a18fb0035c92ae62b705207123"
dependencies = [
"atty",
"bitflags 1.3.2",
"clap_lex",
"indexmap 1.9.3",
"once_cell",
"strsim",
"termcolor",
"textwrap",
]
[[package]]
name = "clap_lex"
version = "0.2.4"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "2850f2f5a82cbf437dd5af4d49848fbdfc27c157c3d010345776f952765261c5"
dependencies = [
"os_str_bytes",
]
[[package]]
name = "cmake"
version = "0.1.58"
@@ -700,6 +736,125 @@ version = "0.3.8"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "06d2e3287df1c007e74221c49ca10a95d557349e54b3a75dc2fb14712c751f04"
[[package]]
name = "dasp"
version = "0.11.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "7381b67da416b639690ac77c73b86a7b5e64a29e31d1f75fb3b1102301ef355a"
dependencies = [
"dasp_envelope",
"dasp_frame",
"dasp_interpolate",
"dasp_peak",
"dasp_ring_buffer",
"dasp_rms",
"dasp_sample",
"dasp_signal",
"dasp_slice",
"dasp_window",
]
[[package]]
name = "dasp_envelope"
version = "0.11.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "8ec617ce7016f101a87fe85ed44180839744265fae73bb4aa43e7ece1b7668b6"
dependencies = [
"dasp_frame",
"dasp_peak",
"dasp_ring_buffer",
"dasp_rms",
"dasp_sample",
]
[[package]]
name = "dasp_frame"
version = "0.11.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "b2a3937f5fe2135702897535c8d4a5553f8b116f76c1529088797f2eee7c5cd6"
dependencies = [
"dasp_sample",
]
[[package]]
name = "dasp_interpolate"
version = "0.11.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "7fc975a6563bb7ca7ec0a6c784ead49983a21c24835b0bc96eea11ee407c7486"
dependencies = [
"dasp_frame",
"dasp_ring_buffer",
"dasp_sample",
]
[[package]]
name = "dasp_peak"
version = "0.11.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "5cf88559d79c21f3d8523d91250c397f9a15b5fc72fbb3f87fdb0a37b79915bf"
dependencies = [
"dasp_frame",
"dasp_sample",
]
[[package]]
name = "dasp_ring_buffer"
version = "0.11.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "07d79e19b89618a543c4adec9c5a347fe378a19041699b3278e616e387511ea1"
[[package]]
name = "dasp_rms"
version = "0.11.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "a6c5dcb30b7e5014486e2822537ea2beae50b19722ffe2ed7549ab03774575aa"
dependencies = [
"dasp_frame",
"dasp_ring_buffer",
"dasp_sample",
]
[[package]]
name = "dasp_sample"
version = "0.11.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "0c87e182de0887fd5361989c677c4e8f5000cd9491d6d563161a8f3a5519fc7f"
[[package]]
name = "dasp_signal"
version = "0.11.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "aa1ab7d01689c6ed4eae3d38fe1cea08cba761573fbd2d592528d55b421077e7"
dependencies = [
"dasp_envelope",
"dasp_frame",
"dasp_interpolate",
"dasp_peak",
"dasp_ring_buffer",
"dasp_rms",
"dasp_sample",
"dasp_window",
]
[[package]]
name = "dasp_slice"
version = "0.11.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "4e1c7335d58e7baedafa516cb361360ff38d6f4d3f9d9d5ee2a2fc8e27178fa1"
dependencies = [
"dasp_frame",
"dasp_sample",
]
[[package]]
name = "dasp_window"
version = "0.11.1"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "99ded7b88821d2ce4e8b842c9f1c86ac911891ab89443cc1de750cae764c5076"
dependencies = [
"dasp_sample",
]
[[package]]
name = "data-encoding"
version = "2.10.0"
@@ -1214,6 +1369,12 @@ dependencies = [
"subtle",
]
[[package]]
name = "hashbrown"
version = "0.12.3"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "8a9ee70c43aaf417c914396645a0fa852624801b24ebb7ae78fe8272889ac888"
[[package]]
name = "hashbrown"
version = "0.15.5"
@@ -1246,6 +1407,15 @@ version = "0.5.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "2304e00983f87ffb38b55b444b5e3b60a884b5d30c0fca7d82fe33449bbe55ea"
[[package]]
name = "hermit-abi"
version = "0.1.19"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "62b467343b94ba476dcb2500d242dadbb39557df889310ac77c5d99100aaac33"
dependencies = [
"libc",
]
[[package]]
name = "hex"
version = "0.4.3"
@@ -1446,6 +1616,16 @@ dependencies = [
"zstd",
]
[[package]]
name = "indexmap"
version = "1.9.3"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "bd070e393353796e801d209ad339e89596eb4c8d430d18ede6a1cced8fafbd99"
dependencies = [
"autocfg",
"hashbrown 0.12.3",
]
[[package]]
name = "indexmap"
version = "2.14.0"
@@ -1739,7 +1919,13 @@ version = "0.5.2"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "805d5964d1e7a0006a7fdced7dae75084d66d18b35f1dfe81bd76929b1f8da0c"
dependencies = [
"anyhow",
"clap",
"dasp",
"dasp_interpolate",
"dasp_ring_buffer",
"easyfft",
"hound",
"once_cell",
]
@@ -1905,6 +2091,12 @@ dependencies = [
"ureq",
]
[[package]]
name = "os_str_bytes"
version = "6.6.1"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "e2355d85b9a3786f481747ced0e0ff2ba35213a1f9bd406ed906554d7af805a1"
[[package]]
name = "p256"
version = "0.11.1"
@@ -2883,6 +3075,21 @@ dependencies = [
"windows-sys",
]
[[package]]
name = "termcolor"
version = "1.4.1"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "06794f8f6c5c898b3275aebefa6b8a1cb24cd2c6c79397ab15774837a0bc5755"
dependencies = [
"winapi-util",
]
[[package]]
name = "textwrap"
version = "0.16.2"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "c13547615a44dc9c452a8a534638acdf07120d4b6847c8178705da06306a3057"
[[package]]
name = "thiserror"
version = "1.0.69"
@@ -3244,7 +3451,7 @@ source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "bb0e353e6a2fbdc176932bbaab493762eb1255a7900fe0fea1a2f96c296cc909"
dependencies = [
"anyhow",
"indexmap",
"indexmap 2.14.0",
"wasm-encoder",
"wasmparser",
]
@@ -3257,7 +3464,7 @@ checksum = "47b807c72e1bac69382b3a6fb3dbe8ea4c0ed87ff5629b8685ae6b9a611028fe"
dependencies = [
"bitflags 2.11.0",
"hashbrown 0.15.5",
"indexmap",
"indexmap 2.14.0",
"semver",
]
@@ -3515,6 +3722,15 @@ version = "0.4.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "ac3b87c63620426dd9b991e5ce0329eff545bccbbb34f3be09ff6fb6ab51b7b6"
[[package]]
name = "winapi-util"
version = "0.1.11"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "c2a7b1c03c876122aa43f3020e6c3c3ee5c05081c9a00739faf7503aeba10d22"
dependencies = [
"windows-sys",
]
[[package]]
name = "winapi-x86_64-pc-windows-gnu"
version = "0.4.0"
@@ -3564,7 +3780,7 @@ checksum = "b7c566e0f4b284dd6561c786d9cb0142da491f46a9fbed79ea69cdad5db17f21"
dependencies = [
"anyhow",
"heck",
"indexmap",
"indexmap 2.14.0",
"prettyplease",
"syn 2.0.117",
"wasm-metadata",
@@ -3595,7 +3811,7 @@ checksum = "9d66ea20e9553b30172b5e831994e35fbde2d165325bec84fc43dbf6f4eb9cb2"
dependencies = [
"anyhow",
"bitflags 2.11.0",
"indexmap",
"indexmap 2.14.0",
"log",
"serde",
"serde_derive",
@@ -3614,7 +3830,7 @@ checksum = "ecc8ac4bc1dc3381b7f59c34f00b67e18f910c2c0f50015669dde7def656a736"
dependencies = [
"anyhow",
"id-arena",
"indexmap",
"indexmap 2.14.0",
"log",
"semver",
"serde",

View File

@@ -7,4 +7,4 @@ edition = "2021"
audiopus = "0.3.0-rc.0"
ezk-g722 = "0.1"
rubato = "0.14"
nnnoiseless = { version = "0.5", default-features = false }
nnnoiseless = "0.5"

View File

@@ -10,7 +10,7 @@ path = "src/main.rs"
[dependencies]
codec-lib = { path = "../codec-lib" }
sip-proto = { path = "../sip-proto" }
nnnoiseless = { version = "0.5", default-features = false }
nnnoiseless = "0.5"
tokio = { version = "1", features = ["full"] }
serde = { version = "1", features = ["derive"] }
serde_json = "1"

View File

@@ -196,7 +196,17 @@ impl CallManager {
};
// Mutable borrow on call/leg is now released.
let sip_pt = codecs.first().copied().unwrap_or(9);
let mut sip_pt = codecs.first().copied().unwrap_or(9);
// If the message has SDP (e.g., 200 OK answer), use the negotiated codec
// instead of the offered one.
if msg.has_sdp_body() {
if let Some(ep) = parse_sdp_endpoint(&msg.body) {
if let Some(pt) = ep.codec_pt {
sip_pt = pt;
}
}
}
match action {
SipLegAction::None => {}
@@ -301,12 +311,20 @@ impl CallManager {
);
// Wire device leg to mixer.
// Use the device's preferred codec from its INVITE SDP,
// not the provider's negotiated codec.
let dev_pt = device_invite
.has_sdp_body()
.then(|| parse_sdp_endpoint(&device_invite.body))
.flatten()
.and_then(|ep| ep.codec_pt)
.unwrap_or(sip_pt);
if let Some(dev_remote_addr) = dev_remote {
let dev_channels = create_leg_channels();
spawn_sip_inbound(dev_rtp_socket.clone(), dev_channels.inbound_tx);
spawn_sip_outbound(dev_rtp_socket, dev_remote_addr, dev_channels.outbound_rx);
if let Some(call) = self.calls.get(call_id) {
call.add_leg_to_mixer(&dev_leg_id, sip_pt, dev_channels.inbound_rx, dev_channels.outbound_tx)
call.add_leg_to_mixer(&dev_leg_id, dev_pt, dev_channels.inbound_rx, dev_channels.outbound_tx)
.await;
}
}

View File

@@ -368,8 +368,8 @@ async fn mixer_loop(
.unwrap_or_else(|_| vec![0.0f32; MIX_FRAME_SIZE])
};
// Per-leg inbound denoising at 48kHz.
// Skip for Opus/WebRTC legs — browsers already apply
// their own noise suppression via getUserMedia.
// Only for SIP telephony legs — WebRTC browsers
// already apply noise suppression via getUserMedia.
let processed = if slot.codec_pt != codec_lib::PT_OPUS {
TranscodeState::denoise_f32(&mut slot.denoiser, &pcm_48k)
} else {

View File

@@ -3,6 +3,6 @@
*/
export const commitinfo = {
name: 'siprouter',
version: '1.17.1',
version: '1.18.0',
description: 'undefined'
}

View File

@@ -3,6 +3,6 @@
*/
export const commitinfo = {
name: 'siprouter',
version: '1.17.1',
version: '1.18.0',
description: 'undefined'
}