RustPBX

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A high-performance, software-defined PBX built in Rust — the AI-native communication platform for next-gen contact centers.

RustPBX externalizes all call control via HTTP/WebSocket/Webhook. Route decisions, media control, and event streams are fully programmable in any language.

Voice Agent functionality has moved to Active Call. This repo focuses on SIP Proxy & PBX.

GitHub | Website

Quick Start

Run RustPBX with minimal config in 2 commands:

Minimal config.toml:

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"
allow_registration = false

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

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

Access	URL
Web Console	`http://localhost:8080/console/`
SIP Proxy	`udp://localhost:5060`
Register SIP phone as	`1001` / `password`

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

Why RustPBX?

Software-Defined	AI-Native	High Performance
Every INVITE calls your HTTP webhook. Return JSON routing decisions. No recompilation needed.	AI agents are native participants — listen, speak, barge, transfer via WebSocket.	800 concurrent calls with RTP proxy: 6ms latency, 0% loss, 280MB memory.

Core Capabilities

SIP & Media — Full SIP stack (UDP/TCP/WS/TLS/WebRTC), RTP relay, NAT traversal, TLS/SRTP with auto ACME certs.

Routing & Control — HTTP Router (dynamic routing decisions), RWI WebSocket Interface (real-time call control), Queue/ACD (sequential or parallel agent ringing).

Recording & Analytics — SipFlow unified SIP+RTP capture, post-call transcript via local SenseVoice (offline), CDR webhooks.

Operations — Built-in Web Console, WebRTC Phone, RBAC, Prometheus metrics + OpenTelemetry.

Programmable Interfaces

RustPBX exposes all call logic through standard protocols — no C modules, no recompilation.

HTTP Router

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

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

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

Actions: forward · reject · abort · spam

RWI (Real-time WebSocket Interface)

JSON-over-WebSocket for in-call control:

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

See API Integration Guide and RWI Protocol.

Editions

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

Benchmark

Tested on 2026-04-03 · RustPBX 0.4.0 · Linux x86_64 · 16 cores / 32 GB · G.711 PCMU

Level	Scenario	Completion	Peak	Loss	Latency	CPU	Mem
500	signaling only	100%	500	0%	4.40ms	32.4%	137 MB
500	+ RTP proxy	100%	500	0%	3.73ms	98.4%	183 MB
500	+ sipflow	100%	500	0%	5.96ms	101%	198 MB
800	signaling only	100%	800	0%	8.32ms	47.9%	192 MB
800	+ RTP proxy	100%	800	0%	6.38ms	155%	265 MB
800	+ sipflow	100%	800	0%	6.08ms	156%	280 MB

Per-channel overhead: ~0.06% CPU / 0.24 MB (signaling); ~0.19% CPU / 0.33 MB (with RTP proxy).

See Benchmark Details for methodology and full results.

Use Cases

Scenario	Description
AI Contact Center	AI agents handle calls 24/7, escalate to humans
Cloud Call Center	Multi-tenant SaaS, remote agents, WebRTC + SIP
Enterprise UC	Internal comms, conferencing, CRM integration
VoIP Wholesale	Multi-carrier routing, flexible billing (Commerce)
Compliance Recording	PCI/healthcare recording, AI quality inspection

Architecture

RustPBX Architecture

App Service (AI Agents, HTTP DialPlan, CRM) → RustPBX Core (B2BUA, IVR, Media, Queue, CDR) → Access (PSTN, WebRTC, SIP, Mobile)

Build from Source

# Linux: apt-get install 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-compile via cross: cargo install cross && cross build --release --target aarch64-unknown-linux-gnu

Screenshots

Extensions	Call Records	Route Editor

Transcript	SIP Flow	WebRTC Phone

Documentation

Guide	Description
Configuration Guide	All config options
API Integration Guide	HTTP Router, Webhooks, Call Control
RWI Protocol	WebSocket Interface

Troubleshooting

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

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

License

Community: MIT · Commercial: hi@miuda.ai

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