RustPBX

A high-performance, software-defined PBX built in Rust.

All call control is externalized via HTTP/WebSocket/Webhook — routing, media control, and events are programmable in any language. No C modules, no recompilation.

GitHub | Website

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Quick Start

Option 1: Docker (recommended)

mkdir -p rustpbx && cd rustpbx
curl -O https://raw.githubusercontent.com/restsend/rustpbx/main/config.toml.example

docker run -d --name rustpbx --net host \
  -v $(pwd)/config.toml.example:/app/config.toml \
  ghcr.io/restsend/rustpbx:latest --conf /app/config.toml

# Create the admin account (first time only)
docker exec rustpbx /app/rustpbx --conf /app/config.toml \
  --super-username admin --super-password changeme

Option 2: Build from source

git clone --recurse-submodules https://github.com/restsend/rustpbx
cd rustpbx
cargo run --release -- --conf config.toml.example

Cross-compile for Windows 64-bit (from Linux/macOS):

rustup target add x86_64-pc-windows-gnu
cargo build --release --target x86_64-pc-windows-gnu
# output: target/x86_64-pc-windows-gnu/release/rustpbx.exe

That’s it — the example config boots a full PBX out of the box:

Access	Value
Web Console	http://localhost:8080/console/
SIP Proxy	udp://localhost:15060
Test extensions	bob / alice, password 123456

No config at all? cargo run --release also works — built-in defaults: HTTP 0.0.0.0:8080, SQLite, SIP UDP 5060.

Minimal config

Want your own config.toml? This is all you need:

http_addr = "0.0.0.0:8080"
database_url = "sqlite://rustpbx.sqlite3"

[proxy]
addr = "0.0.0.0"
udp_port = 5060
modules = ["auth", "registrar", "call"]

[[proxy.user_backends]]
type = "memory"
users = [{ username = "1001", password = "password" }]

[console]
base_path = "/console"

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Features

SIP & Media	Full SIP stack (UDP/TCP/WS/TLS/WebRTC), RTP relay & transcoding, NAT traversal, TLS/SRTP with auto ACME certs
Routing & Control	HTTP Router (webhook routing decisions), RWI WebSocket (real-time call control), Queue/ACD
AI-Native	AI agents as native call participants — listen, speak, barge, transfer via WebSocket
Recording & Analytics	SipFlow unified SIP+RTP capture, offline transcription (SenseVoice), CDR webhooks
Operations	Built-in Web Console & WebRTC phone, RBAC, Prometheus metrics, OpenTelemetry

Performance: 5000 concurrent calls with RTP relay on ~3.8 cores, 0% packet loss — linear scaling. See benchmark details.

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Programmable Interfaces

HTTP Router

Every incoming INVITE calls your webhook. Return JSON to decide routing:

// POST to your webhook: { "call_id": "abc-123", "from": "sip:+861390000@trunk", "to": "sip:400800" }
// Your response:       { "action": "forward", "targets": ["sip:1001@internal"], "record": true }

Actions: forward · reject · abort · spam

RWI (Real-time WebSocket Interface)

JSON-over-WebSocket for in-call control: originate, answer, hangup, bridge, transfer, play, record.start, queue/ACD commands, supervisor listen/whisper/barge, conference commands, and more.

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Documentation

Guide	Description
Configuration Guide	All config options
API Integration Guide	HTTP Router, Webhooks, Call Control
RWI Protocol	WebSocket Interface
Outbound Dial SSE API	Predictive outbound dialing
Live Transcript SSE API	Real-time transcription

Troubleshooting

SIP 401 behind NAT/Docker — set the realm explicitly:

[proxy]
realms = ["your-public-ip:5060"]

Contributors

@shenjinti · @yeoleobun · @dengxiayehu · @zen2do · @autopeasant · @rgon · @themooer1 · @v0l · @tgeorge06 · @umran · @davidcforbes · @deweydb · @zhouhailin

License

Community: MIT · Commercial: hi@miuda.ai · https://miuda.ai