v1.19.1

changelog.md

@@ -1,5 +1,12 @@
 # Changelog
 
+## 2026-04-10 - 1.19.1 - fix(readme)
+refresh documentation for jitter buffering, voicemail, and WebSocket signaling details
+
+- Add adaptive jitter buffer and packet loss concealment details to the audio pipeline documentation
+- Document voicemail unheard count and heard-state API endpoints
+- Update WebSocket event and browser signaling examples to reflect current message types
+
 ## 2026-04-10 - 1.19.0 - feat(proxy-engine,codec-lib)
 add adaptive RTP jitter buffering with Opus packet loss concealment and stable 20ms resampling
 

package.json

@@ -1,6 +1,6 @@
 {
   "name": "siprouter",
-  "version": "1.19.0",
+  "version": "1.19.1",
   "private": true,
   "type": "module",
   "scripts": {

readme.md

@@ -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, voicemail, IVR menus, 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, adaptive jitter buffering, ML noise suppression, neural TTS, voicemail, IVR menus, and a slick web dashboard.
 
 ## Issue Reporting and Security
 
@@ -17,6 +17,7 @@ siprouter sits between your SIP trunk providers and your endpoints — hardware
 - 🎛️ **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
+- 🎯 **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
@@ -49,7 +50,7 @@ siprouter sits between your SIP trunk providers and your endpoints — hardware
 │  │                                │  │
 │  │ SIP Stack · Dialog SM · Auth   │  │
 │  │ Call Manager · N-Leg Mixer     │  │
-│  │ 48kHz f32 Bus · RNNoise       │  │
+│  │ 48kHz f32 Bus · Jitter Buffer  │  │
 │  │ Codec Engine · RTP Port Pool   │  │
 │  │ WebRTC Engine · Kokoro TTS     │  │
 │  │ Voicemail · IVR · Recording    │  │
@@ -246,14 +247,15 @@ The `proxy-engine` binary handles all real-time audio processing with a **48kHz
 ### Audio Pipeline
 
 ```
-Inbound:   Wire RTP → Decode → Resample to 48kHz → Denoise (RNNoise) → Mix Bus
+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
 ```
 
-- **FFT-based resampling** via `rubato` — high-quality sinc interpolation with cached resampler state for seamless inter-frame continuity
+- **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.
+- **Packet loss concealment (PLC)** — on missing packets, Opus legs invoke the decoder's built-in PLC (`decode(None)`) to synthesize a smooth fill frame. Non-Opus legs (G.722, PCMU) apply exponential fade (0.85×) toward silence to avoid hard discontinuities.
+- **FFT-based resampling** via `rubato` — high-quality sinc interpolation with canonical 20ms chunk sizes to ensure consistent resampler state across frames, preventing filter discontinuities
 - **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
 
 ---
@@ -272,9 +274,10 @@ Announcements and voicemail greetings are synthesized using [Kokoro TTS](https:/
 
 ## 📧 Voicemail
 
-- Configurable voicemail boxes with custom TTS greetings
-- Automatic routing on no-answer timeout
-- Recording with configurable max duration and message count
+- Configurable voicemail boxes with custom TTS greetings (text + voice) or uploaded WAV
+- Automatic routing on no-answer timeout (configurable, default 25s)
+- Recording with configurable max duration (default 120s) and message count limit (default 50)
+- Unheard message tracking for MWI (message waiting indication)
 - Web dashboard playback and management
 - WAV storage in `.nogit/voicemail/`
 
@@ -319,8 +322,10 @@ Announcements and voicemail greetings are synthesized using [Kokoro TTS](https:/
 | `/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/unheard` | GET | Get unheard message count |
 | `/api/voicemail/:box/:id/audio` | GET | Stream voicemail audio |
+| `/api/voicemail/:box/:id/heard` | POST | Mark a voicemail message as heard |
+| `/api/voicemail/:box/:id` | DELETE | Delete a voicemail message |
 
 ### WebSocket Events
 
@@ -329,8 +334,18 @@ Connect to `/ws` for real-time push:
 ```jsonc
 { "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
+{ "type": "call-update", "data": { ... } }      // Call state change notification
+{ "type": "webrtc-answer", "data": { ... } }    // WebRTC SDP answer for browser calls
+{ "type": "webrtc-error", "data": { ... } }     // WebRTC signaling error
+```
+
+Browser → server signaling:
+
+```jsonc
+{ "type": "webrtc-offer", "data": { ... } }     // Browser sends SDP offer
+{ "type": "webrtc-accept", "data": { ... } }    // Browser accepts incoming call
+{ "type": "webrtc-ice", "data": { ... } }       // ICE candidate exchange
+{ "type": "webrtc-hangup", "data": { ... } }    // Browser hangs up
 ```
 
 ---
@@ -365,7 +380,7 @@ pnpm run restartBackground
 
 ## License and Legal Information
 
-This repository contains open-source code licensed under the MIT License. A copy of the license can be found in the [license](./license) file.
+This repository contains open-source code licensed under the MIT License. A copy of the license can be found in the [LICENSE](./LICENSE) file.
 
 **Please note:** The MIT License does not grant permission to use the trade names, trademarks, service marks, or product names of the project, except as required for reasonable and customary use in describing the origin of the work and reproducing the content of the NOTICE file.
 

ts/00_commitinfo_data.ts

@@ -3,6 +3,6 @@
  */
 export const commitinfo = {
   name: 'siprouter',
-  version: '1.19.0',
+  version: '1.19.1',
   description: 'undefined'
 }

ts_web/00_commitinfo_data.ts

@@ -3,6 +3,6 @@
  */
 export const commitinfo = {
   name: 'siprouter',
-  version: '1.19.0',
+  version: '1.19.1',
   description: 'undefined'
 }