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RustPBXOpen source · MIT licensed · v0.5

Intelligent voice infrastructure
built with Rust.

SIP trunking, intelligent routing, real-time billing, CDR analytics, and observability in one open-source softswitch.

sipflow · call-id 7f3a9c21@pbx-1live
ext 1001RustPBXcarrier-bINVITE sip:+4420…0.00100 Trying0.31INVITE (LCR → carrier-b)1.12180 Ringing48.6180 Ringing48.9200 OK (opus/48000)2310200 OK2311ACK2342RTP · SRTP ⟷ two-way media
PDD 48 ms
MOS 4.41
Loss 0.02 %
Concurrent calls on a single server
5,000+
Maximum supported call-center agents
8,000+
Call cost reduction after LCR and circuit breakers
38%
Availability with drain-based rolling upgrades
99.9%

Who we serve

For global enterprises, VoIP wholesalers, cloud communications platforms, and CPaaS providers.

Products

One core. Three voice services.

PBX, wholesale softswitch, and call center share the same SIP proxy, media engine, CDRs, and control plane. Enable the commercial plugins you need.

RustPBX

Capabilities

Everything you need for voice operations

Every call takes the same path through one core: checked at the edge, routed to the best carrier, rated in real time, traced end to end and rolled into reports.

Inbound call · INVITE +44 20 7946 0958

  1. 01At the edge

    Security and reliability

    Carrier-grade reliability built on the Rust async runtime, with layered protection.

    • High-concurrency async architecture designed for 99.9%+ availability
    • IP ACLs, scanner User-Agent blacklists, firewall defenses, and rate limiting help defend against DDoS attacks and unauthorized access
    • Real-time monitoring and alerts for system health, active calls, and resource usage
    EDGE PROTECTIONINTERNETIP ACLdeny default−1Scanner UASipVicious−2Rate limitREG floods−2Authdigest / JWT−1PBXcore+2SYSTEM HEALTHUptime42d 06hCPU23%Memory412 MBActive calls1,284 / 5,500Trunk probescarrier-acarrier-bcarrier-cALERTSTrunk carrier-b not answeringOPTIONS probe timeout → fallback trunkauth_ban 203.0.113.75 failed REGISTERs → webhook · mail
  2. 02Choosing a route

    Routing and traffic management

    Precise control over call paths, quality, and carrier selection.

    • LCR, priority routing, and quality-based adaptive routing
    • Sliding-window ASR/ACD circuit breakers and automatic failover protect answer rates and margins
    • SIP trunk management, number rewriting, and header manipulation for seamless carrier interoperability
    INVITE+44 20 7946 0958tenant acme-voipROUTING PROFILEpremium-routesprefix 44 → 3 trunksorder: cost → priorityskip: window ASR < 30%CANDIDATES · PREFIX 441carrier-a$0.0061/minASR 56%ASR 18%closed✕ OPEN · skip2carrier-b$0.0068/minASR 61%closed3carrier-c$0.0074/minASR 58%closedEGRESS TO CARRIER-B · REWRITE✓ 200 OKR-URI+442079460958 → 9901#442079460958Caller1001 → +44 20 3808 0000 (pool)Headers− P-Asserted-Identity + X-Route: premium→ Failover in one hop; the carrier gets a clean, rewritten INVITE
  3. 03Rating the call

    Billing and financial management

    Real-time billing, credit control, and invoicing for prepaid and postpaid services.

    • Real-time billing for prepaid and postpaid services, with automatic suspension and multiple currencies
    • Complete CDR management with real-time search across individual and aggregated records
    • Rate tables, tenant balances, CPS / concurrency limits, and automated invoicing
    Billing and financial management
  4. 04During the call

    Observability and end-to-end tracing

    Trace every call from SIP signaling to RTP media quality.

    • SipFlow captures SIP and RTP traffic for per-call signaling diagnostics and recording playback
    • End-to-end call tracing with call-leg timelines, error references, and per-call media quality (packet loss, jitter, MOS)
    • Millisecond-level real-time data helps teams find and resolve issues quickly
    Observability and end-to-end tracing
  5. 05After the call

    Reporting and analytics

    Detailed insights by route, tenant, and agent team.

    • Trend charts for call volume, duration, and ASR
    • Revenue, cost, and margin analysis by route and destination
    Reporting and analytics

BYE · CDR written

Highlights

Optimized for the Call-Processing Hot Path

Performance

High-concurrency performance

Rust’s async ecosystem keeps sub-millisecond scheduling and stable performance in high-concurrency media scenarios.

5,000+ concurrent calls on a single server

Security

Secure and modern protocols

WebRTC and SRTP (DTLS / SDES), SIP over TLS, and WebSocket, with support for G.711, G.722, G.729, Opus, and iLBC.

SRTP · SIP over TLS · WebRTC · Opus · G.722

Extensibility

Open & extensible

HTTP dynamic routing, Step IVR, RWI over WebSocket and REST, webhooks, and a plugin system.

REST · RWI WebSocket · Webhooks · Addons

Voice agents

Put your voice agent on the phone.

  • 01

    Bring your own agent with RWI

    Your app runs the conversation. It exchanges audio over WebSocket and uses RWI to answer calls or transfer callers to your team.

  • 02

    Connect a realtime voice model

    Choose a supported endpoint. The built-in RustPBX adapter streams the caller’s speech to the model and plays its replies back.

Read the RWI guide
CustomerSIP / RTPHuman agentSupport · ext. 204TransferRustPBXExecutes call actionsSIP / RTP ↔ PCMTwo-way audioWebSocketRWIcall.hangupcall.transferRWI CLIENTVoice agentChooses the next actionConversation contextModel responseLLM

Compare

RustPBX vs traditional voice stacks

Asterisk and FreeSWITCH are proven systems. RustPBX targets a different operating point: one memory-safe core with a modern control plane, built for clusters and automation.

RustPBXFreeSWITCHAsterisk
Language / safetyRust (memory-safe)CC
Signaling + media in one runtimeYes, async tasks on a shared runtimeYes, a thread per session (pooled)Yes, a thread per channel
Cluster session registryBuilt inExternal designExternal design
Control planeREST + RWI WebSocketESLAMI / ARI
Wholesale billing / BSSIntegrated addonExternalExternal
Call Center / ACDIntegrated addonVia FusionPBX etc.Third-party
LicenseMIT core + commercial addonsMPLGPL / commercial

Customers

Deployments in production

Supporting SMS and voice communications providers and ride-hailing platforms. These anonymous case studies show how customers use Miuda in production.

SMS & voice communications · 200 employees

A communications provider brings wholesale operations together

Complex routing, fragmented tenant management, lengthy settlements and growing compliance requirements made wholesale VoIP operations difficult to scale. The provider needed more precise tenant management, automated billing and better use of carrier resources.

  • Voice wholesale for multiple customers with high concurrent call volumes.
  • Trunk management with traffic distribution, allowlists, blocklists and anomalous traffic blocking.
  • Multi-tenant billing operations with automated calculations and account reconciliation.

Routing by tenant and business rules improves carrier resource utilization. Bulk management, automated billing and policy distribution streamline operations, while route optimization, audit trails and risk controls help reduce operational risk.

CUSTOMER INDUSTRYCommunications service providerSMS messaging and wholesale VoIP services

Ride-hailing · 3,000+ call-center agents

A ride-hailing platform connects drivers and passengers at scale

Peak call volumes risked delays and service interruptions. Long queues and slow responses affected drivers and passengers, increasing churn risk, while a call center with more than 3,000 agents carried substantial staffing and management costs.

  • Smart notifications for trip changes, service exceptions, marketing campaigns and emergency safety announcements.
  • Automated follow-up on cancellations, initial reviews of fare disputes, and lost-item enquiries and tracking.
  • Driver and passenger coordination for pickup locations and real-time trip changes, with handoffs between voice agents and human support.

Cluster deployment and elastic scaling support peak demand and reliable communication. Voice bots handle frequent, routine tasks around the clock to reduce operating costs. Fast responses and collaboration between AI and human agents improve the driver and passenger experience.

CUSTOMER INDUSTRYRide-hailing platformRide-hailing and mobility services

FAQ

Frequently asked questions

Is RustPBX free and open source?

Yes. The RustPBX core is open source under the MIT license, including the SIP proxy, media engine, call records and web console. Clustering is part of the open-source core; commercial editions add licensed addons such as one-step cluster management, wholesale billing and the call center.

Which protocols and codecs does RustPBX support?

SIP over UDP, TCP and TLS, SIP over WebSocket for WebRTC, and SRTP including DTLS-SRTP and SDES-SRTP. Codecs include G.711 a/µ, G.729, G.722, Opus and iLBC, selectable per trunk or extension, with transcoding in the media bridge.

Can I run a wholesale voice business on it?

Yes. The wholesale addon adds tenant management, rate decks, routing profiles with least-cost routing, circuit breakers with sliding-window ASR/ACD monitoring, real-time billing and per-tenant CDRs — a complete class 4 softswitch and BSS stack.

What call center features are included?

Skill-based ACD queues, agent states with wrap-up, overflow strategies, live supervisor monitoring with listen / whisper / barge-in, screen-pop with Call-Info / User-to-User headers, and post-call CSAT surveys.

Does it integrate with AI voice services?

Yes. ASR/TTS drivers, per-call transcription, a realtime AI-voice bridge for LLM agents and an IVR engine with dynamic prompts let you build voice agents that hand off to humans on the same platform.

How do I deploy RustPBX?

With Docker or as a single static binary on Linux. The built-in web console handles extensions, trunks, routing and diagnostics. SQLite works out of the box; MySQL or PostgreSQL is recommended for clusters.

Ready to modernise your voice stack?

Start a 7-day trial, or tell us your scale and we’ll walk through routing, queues and billing with you.