OpenSIPS

OpenSIPS is a GPL‑licensed, high‑performance SIP proxy/server written in C that serves as the backbone for a wide spectrum of real‑time communication services. It is architected to process tens of thousands of calls per second while remaining highly configurable through a modular framework. The core daemon exposes a powerful routing script language (based on the SER “route” syntax) that allows developers to define custom call‑flow logic, load‑balancing, and policy enforcement at the SIP level. OpenSIPS can be extended via a rich module ecosystem—over 120 modules covering everything from authentication, presence, and media handling to database integration and external API hooks.

Architecture & Technical Stack

• Core Engine: C‑based, event‑driven, non‑blocking I/O model that scales on multi‑core CPUs. The codebase is actively maintained with CI pipelines (GitHub Actions) and fuzz testing, ensuring stability for production deployments.
• Module System: Each module implements a well‑defined interface, allowing developers to plug in custom logic (e.g., auth, acc, db_mysql) or write new modules in C. Modules can be loaded/unloaded at runtime, facilitating rapid iteration.
• Database Layer: Supports a variety of relational databases (MySQL/MariaDB, PostgreSQL) and NoSQL backends via the db_mysql, db_postgres, or db_sqlite modules. The SQL abstraction layer allows developers to write database‑agnostic routing scripts.
• External APIs: OpenSIPS exposes a RESTful JSON API (via the json module) and a gRPC interface for real‑time control, making it straightforward to integrate with web services or microservice architectures.
• Transport & Media: Native support for SIP over UDP/TCP/TLS, WebSocket (WSS), and DTLS. The rtp_proxy module can act as a media relay, while the new RTP.io engine provides efficient media handling.

Core Capabilities for Developers

• Routing Scripts: Fine‑grained control over SIP message flow using the route{} blocks, with access to session variables ($ru, $rr, etc.) and built‑in functions for SIP header manipulation.
• Dynamic Sockets & Structured SDP: The 3.6 release introduces dynamic socket handling and structured SDP parsing, enabling advanced media negotiation patterns.
• Unified Branches & Call‑Control: Features like branch routing and call‑control APIs allow developers to build complex call scenarios (e.g., conference bridges, IVR systems) with minimal boilerplate.
• Extensible Module API: The module development kit (MDK) provides macros and helper functions to expose new C APIs, making it possible to integrate proprietary protocols or custom authentication schemes.

Deployment & Infrastructure

OpenSIPS is designed for self‑hosting in both on‑premise data centers and cloud environments. It runs as a single binary, making it lightweight for containerization (Docker images are available on Docker Hub). The CI pipeline builds multi‑arch binaries (amd64, arm64), and the project offers official Dockerfiles for quick spin‑ups. For high‑availability, developers can deploy multiple OpenSIPS instances behind a load balancer or use the cluster module to share state via Redis or PostgreSQL. The daemon’s low memory footprint (≈ 50 MB) and CPU efficiency enable scaling to millions of concurrent calls on commodity hardware.

Integration & Extensibility

• Plugin Ecosystem: Hundreds of community and commercial modules cover presence, WebRTC signaling, SIP over TLS, and more. Developers can contribute new modules or fork existing ones.
• Webhooks & Callbacks: The websocket and json modules expose real‑time event streams, allowing external services to react to call state changes or authentication events.
• Custom Modules: The MDK supports exporting C functions that can be called from routing scripts, enabling tight coupling with legacy systems or custom billing engines.
• API Hooks: The rest and grpc modules provide programmatic control over registration, subscription, and call routing, facilitating integration with modern microservice stacks.

Developer Experience

OpenSIPS boasts comprehensive documentation on the official website, with module‑level READMEs and a detailed manual for routing scripts. The active mailing lists (users, devel) provide rapid support, and the community actively contributes patches. The project’s open‑source nature (GPL) removes licensing friction, while the modular architecture ensures that developers can ship feature‑rich services without reinventing core SIP logic.

Use Cases

• Carrier‑grade PBX: High‑throughput trunking, call routing, and billing integration.
• WebRTC Gateways: SIP‑to‑WebSocket translation for browser‑based VoIP.
• Session Border Controllers: Media encryption, NAT traversal, and QoS enforcement.
• Call Centers & IVR: Dynamic call routing with real‑time analytics via the acc module.
• IoT Telephony: Lightweight SIP endpoints for machine‑to‑machine communication.

Advantages Over Alternatives