Rust-grade concurrency
Rust's async ecosystem keeps sub-millisecond scheduling even in high-concurrency media scenarios, ensuring consistent call quality.
Rust-native PBX
RustPBX: A high-performance, secure, fully open voice core
RustPBX is maintained by the miuda.ai team and released fully open source. Built in Rust, it pairs realtime performance with memory safety, delivering carrier-grade reliability alongside a modern developer experience for AI-native voice applications.
Why RustPBX?
RustPBX embraces an observable, extensible PBX architecture: the core modules run efficiently on the Rust runtime, while downstream systems integrate through APIs, webhooks, or WebSocket event streams to sync with AI assistants, CRMs, ticketing, and more.
- Open source under MIT—customize and extend without restriction
- Security-first design avoids memory risks common in legacy PBX stacks
- Cloud-ready deployment model that fits CI/CD and infrastructure-as-code pipelines
Get started fast
Launch with docker compose or Kubernetes, then connect with the RustPBX SDK to complete your first call flow in minutes.
A voice platform engineered for builders
A scalable foundation that spans media, orchestration, and compliance so teams can trust their voice core.
Full-stack open ecosystem
Core code, SDKs, and ops tooling are open sourced for community-driven plugins and bespoke voice + AI workflows.
Event-driven AI integration
WebSocket / gRPC event buses stream call state in realtime for seamless AI agent, QA, and knowledge base integrations.
Core calling capabilities
A carrier-grade voice core that covers access, media, and call control with deterministic latency.
- Unlimited extensions with per-device concurrent login policies
- Codec support: G.711 / G.722 / G.729 / Opus
- Built-in WebRTC and SRTP for encrypted, cross-platform sessions
- SIP over WebSocket for seamless browser endpoints
- DTMF passthrough via RFC2833 and SIP INFO
- Native recording, attended and blind transfer, and three-way calling
- Adaptive call queues with hold music, retries, overflow routing, and more
Configurable operations
Visual tooling turns complex routing and billing logic into intuitive workflows.
- Visual routing rules with an advanced drag-and-drop editor
- SIP Trunk and DID onboarding with capacity management
- Custom billing templates, including department-level chargeback
- Rich call detail records with transcription and export
Engineered for operations teams
Full-stack tooling for DevOps and voice engineers to deploy, debug, and scale with confidence.
- In-browser routing and SIP trunk tests powered by WebRTC + SIP over WebSocket
- Live lookup for calls, registered devices, and endpoint status
- Advanced admin console with observability and audit trails
- Multi-account, multi-department, role-based access control
- Support for MySQL, PostgreSQL, and SQLite backends
- Guided IVR design and publishing pipeline
Production-grade security posture
Defend high-concurrency, real-time voice workloads from access to data lifecycle.
- Embedded firewall with IP and User-Agent allow/deny policies
- Configurable call concurrency caps and rate limits to absorb spikes
- Automated archival with scheduled backups to S3-compatible storage
- Standardized Docker-based deployment for consistent environments
Ready to build the next-generation voice platform?
Talk with us about enterprise deployment, migration planning, or bespoke development. We welcome community contributors who want to shape the future of PBX with our engineering team.