RustPBX

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

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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

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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.	5000 concurrent calls with RTP proxy: ~3.8 cores, 0% loss. Linear scaling — 16 cores → ~16000+.

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Core Capabilities

SIP & Media — Full SIP stack (UDP/TCP/WS/TLS/WebRTC), RTP relay, NAT traversal, TLS/SRTP with auto ACME certs. Fast registration via JWT or HTTP token (skip 401/407).

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.

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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.

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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)	❌	✅

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Benchmark

Tested on 2026-07-09 · RustPBX 0.4.10 · rustrtc 0.3.89 · Linux x86_64 · AMD Ryzen 7 5700X (8C/16T) / 32 GB · G.711 PCMU · CPS=100

Concurrent	Scenario	Completion	Peak Conc	Packet Loss	CPU(Peak/Avg)	Memory(Peak)
500	signaling only	100%	600	0%	18.4% / 9.9%	382 MB
500	+ RTP proxy (bridge)	100%	595	0%	77.3% / 59.7%	524 MB
500	+ sipflow	100%	597	0%	75.7% / 33.4%	569 MB
2000	signaling only	100%	2090	0%	39.8% / 24.5%	704 MB
2000	+ RTP proxy (bridge)	100%	2094	0%	255% / 186%	1209 MB
2000	+ sipflow	100%	2086	0%	254% / 186%	1209 MB
4000	signaling only	100%	4078	0%	54.0% / 34.8%	1131 MB
4000	+ RTP proxy (bridge)	100%	4083	0%	363% / 277%	2101 MB
4000	+ sipflow	100%	4079	0%	363% / 277%	2104 MB
5000	signaling only	100%	5066	0%	58.2% / 39.1%	1343 MB
5000	+ RTP proxy (bridge)	100%	4670	0%	378% / 298%	2389 MB
5000	+ sipflow	100%	4693	0%	378% / 297%	2416 MB

Per-channel overhead: 0.008% CPU (signaling) / ~0.076% CPU (RTP proxy). Scaling is linear — 5000 concurrent uses ~3.8 cores, leaving ~12 cores for additional load (16000+ theoretical).

See Benchmark Details for methodology and full results.

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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

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Architecture

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

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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 --recurse-submodules 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

Submodules

Commerce addons are managed as git submodules under src/addons/:

Submodule	Repository
src/addons/cc	https://cnb.cool/miuda.ai/cc
src/addons/wholesale	https://cnb.cool/miuda.ai/wholesale
src/addons/endpoint_manager	https://cnb.cool/miuda.ai/endpoint_manager
src/addons/enterprise_auth	https://cnb.cool/miuda.ai/enterprise_auth
src/addons/ivr_editor	https://cnb.cool/miuda.ai/ivr_editor
src/addons/sbc	https://cnb.cool/miuda.ai/sbc
src/addons/telemetry	https://cnb.cool/miuda.ai/telemetry
src/addons/voicemail	https://cnb.cool/miuda.ai/voicemail

# Initialize submodules after clone
git submodule update --init --recursive

# Pull latest changes for all submodules
git submodule update --remote

# Pull latest for a specific submodule
git submodule update --remote src/addons/cc

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Screenshots

Extensions	Call Records	Route Editor

Transcript	SIP Flow	WebRTC Phone

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Documentation

| Guide | Description | |—|—|—| | Configuration Guide | All config options | | Authentication | User backends, JWT & HTTP token fast registration | | API Integration Guide | HTTP Router, Webhooks, Call Control, Recording | | RWI Protocol | WebSocket Interface | | RWI Events Reference | Event types, fields, JSON examples (中文) | | RWI Events Reference (EN) | Event types, fields, JSON examples (English) |

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Troubleshooting

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

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

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License

Community: MIT · Commercial: hi@miuda.ai

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