RustPBX

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A high-performance Software-Defined PBX built in Rust — An AI-native communication platform for next-generation contact centers.

Different from Asterisk/FreeSWITCH (C modules), RustPBX exposes all call control via HTTP/WebSocket/Webhook, making it fully programmable in any language. Route decisions, media control, and event streams are all externalized — AI becomes a native participant in every call.

Note: The Voice Agent functionality has been moved to a separate repository: Active Call. This repository now focuses on the SIP Proxy and PBX features.

GitHub | Website

Software-Defined Communication

RustPBX breaks the closed architecture of traditional PBX systems by exposing three powerful integration channels:

| Channel | Protocol | Purpose | |---------|----------|---------| | Policy Decision | HTTP Router | Real-time routing decisions: AI first, agent queue, IVR, or direct transfer | | Real-time Control | RWI (WebSocket) | In-call control: listen, whisper, barge, transfer, hold, media injection | | Event Stream | Webhook | Push CDR, queue status, and events to your CRM/ticketing system |

Editions

| | Community | Commerce | |---|---|---| | License | MIT | Commercial | | SIP Proxy + Media | ✅ | ✅ | | HTTP Router (dynamic routing) | ✅ | ✅ | | Queue / ACD | ✅ | ✅ | | Call Recording + SipFlow | ✅ | ✅ | | Transcript (SenseVoice offline) | ✅ | ✅ | | Web Console | ✅ | ✅ | | RWI (WebSocket Interface) | ✅ | ✅ | | VoIP Wholesale (VOS3000 alternative) | ❌ | ✅ | | IVR Visual Editor | ❌ | ✅ | | Voicemail Pro | ❌ | ✅ | | Enterprise Auth (LDAP/SAML/MFA) | ❌ | ✅ | | Endpoint Manager (phone auto-provisioning) | ❌ | ✅ |

AI-Native UCaaS Architecture

RustPBX Architecture

Architecture Layers:

App Service Layer: AI Voice Agent, Human Agents, HTTP DialPlan Handler, RWI Call Control, Webhook Consumer, CRM/Ticketing
RustPBX Core Layer: B2BUA, IVR, Media Fabric, Queue/ACD, Recording, CDR, SIP Trunk
Access Layer: PSTN (SIP Trunk), WebRTC Browser, SIP Client, Mobile App

Core Capabilities

SIP & Media

SIP Proxy — Full SIP stack (UDP/TCP/WS/TLS/WebRTC), registration, auth, B2BUA
Media Proxy — RTP relay, NAT traversal, WebRTC ↔ SIP bridging
TLS/SRTP — End-to-end encryption with automatic ACME certificate management

Routing & Control

HTTP Router — Every INVITE hits your webhook; you return routing decision in JSON
RWI (WebSocket Interface) — Real-time call control: originate, answer, hold, transfer, record, queue management, supervisor whisper/barge, and media stream injection (PCM)
Queue / ACD — Sequential or parallel agent ringing, hold music, priority scheduling

Recording & Analytics

SipFlow Recording — Unified SIP+RTP capture; hourly files with on-demand playback (no file-handle exhaustion)
Transcript — Post-call transcription via local SenseVoice (offline, no cloud dependency)
CDR Webhooks — Push call detail records + recordings to your system on hangup

Operations

Web Console — Built-in management UI with visual configuration
WebRTC Phone — Built-in browser softphone for testing
RBAC — Role-Based Access Control with fine-grained permissions
Observability — Built-in Prometheus metrics + OpenTelemetry tracing

Typical Use Cases

| Scenario | Description | |----------|-------------| | AI Contact Center | AI Voice Agent handles incoming calls, transfers to human agents for complex issues, 24/7 availability | | Cloud Call Center | Multi-tenant SaaS architecture, remote agents, WebRTC + SIP endpoints | | Enterprise UC | Internal communication, conferencing, voicemail, CRM/OA integration | | VoIP Wholesale | Multi-carrier routing, flexible billing, profit optimization (Commercial) | | Compliance Recording | Financial/healthcare compliance recording, AI quality inspection, PCI masking | | Outbound Marketing | Predictive dialing, call analytics, lead scoring |

Quick Start (Docker)

# Commerce image (includes Wholesale + all commercial plugins)
docker pull docker.cnb.cool/miuda.ai/rustpbx:latest

# Community image
docker pull ghcr.io/restsend/rustpbx:latest

Minimal config.toml:

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

[console]
base_path = "/console"
allow_registration = false

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

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

[sipflow]
type = "local"
root = "./config/cdr"
subdirs = "hourly"

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

# Create first admin
docker exec rustpbx /app/rustpbx --conf /app/config.toml \
  --super-username admin --super-password changeme

Web console: http://localhost:8080/console/ SIP proxy: udp://localhost:5060

Build from Source

Dependencies (Linux):

apt-get install -y cmake pkg-config libasound2-dev libssl-dev libopus-dev

macOS:

brew install cmake openssl pkg-config

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

Cross-compilation for aarch64 / x86_64 via cross:

cargo install cross
cross build --release --target aarch64-unknown-linux-gnu

HTTP Router — The Key Extension Point

RustPBX calls your API on every incoming INVITE. You decide what happens:

[proxy.http_router]
url = "https://your-api.com/route"
timeout_ms = 3000

// POST to your endpoint:
{ "call_id": "abc-123", "from": "sip:+861390000@trunk", "to": "sip:400800", "direction": "inbound" }

// Your response:
{ "action": "forward", "targets": ["sip:ai-agent@internal"], "record": true }

Actions: forward · reject · abort · spam See API Integration Guide for the full webhook and active call control reference.

RWI — Real-time WebSocket Interface

RWI provides JSON-over-WebSocket for real-time call control:

| Category | Commands | |----------|----------| | Call Control | originate, answer, hangup, bridge, transfer, hold, reject | | Media Control | play, stop, stream_start, inject_start (PCM) | | Recording | record.start, pause, resume, stop, mask_segment | | Queue Management | enqueue, dequeue, set_priority, assign_agent, requeue | | Supervisor | listen, whisper, barge, takeover | | Conference | create, add, remove, mute, destroy |

See RWI Protocol for details.

Screenshots

| Extensions | Call Records | Route Editor | |---|---|---| | | | |

| Transcript | SIP Flow | WebRTC Phone | |---|---|---| | | | |

Documentation

| | | |---|---| | Configuration Guide | All config options | | API Integration Guide | HTTP Router, Webhooks, Active Call Control | | RWI Protocol | WebSocket Interface for real-time call control |

Troubleshooting

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

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

Tech Stack

Built with: Rust · Tokio · Axum · SeaORM · SQLite/PostgreSQL/MySQL · Prometheus · OpenTelemetry · WebRTC · SIP (RFC 3261)

License

Community edition: MIT Commercial edition : hi@miuda.ai

https://miuda.ai - Maintenance & commercial support

Rustpbx

Install / Use

README