v1.20.5

changelog.md

@@ -1,5 +1,11 @@
 # Changelog
 
+## 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
 

package.json

@@ -1,6 +1,6 @@
 {
   "name": "siprouter",
-  "version": "1.20.4",
+  "version": "1.20.5",
   "private": true,
   "type": "module",
   "scripts": {

readme.md

@@ -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
@@ -246,9 +264,17 @@ 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.

ts/00_commitinfo_data.ts

@@ -3,6 +3,6 @@
  */
 export const commitinfo = {
   name: 'siprouter',
-  version: '1.20.4',
+  version: '1.20.5',
   description: 'undefined'
 }

ts_web/00_commitinfo_data.ts

@@ -3,6 +3,6 @@
  */
 export const commitinfo = {
   name: 'siprouter',
-  version: '1.20.4',
+  version: '1.20.5',
   description: 'undefined'
 }