Kamailio

Kamailio is a high‑performance, open‑source SIP signaling server designed to route and proxy millions of call setups per second. It implements the full RFC3261 SIP stack while extending it with modern features such as WebSocket support for WebRTC, TLS encryption, and asynchronous processing. The core is written in C and compiled with a modular architecture that allows developers to enable or disable functionality at compile time, giving fine‑grained control over the binary footprint and runtime overhead.

• Scalable routing engine – supports thousands of concurrent dialogs, load‑balancing across back‑ends (e.g., Asterisk, FreeSWITCH), and least‑cost routing with ENUM/DID.
• Asynchronous I/O – non‑blocking TCP/UDP/SCTP and WebSocket handling, reducing context switches and memory usage.
• Extensive backend integration – native modules for MySQL, PostgreSQL, Oracle, Redis, MongoDB, Cassandra, LDAP, Radius, and Memcached.
• Rich API surface – JSON/XML‑RPC control interface, SNMP monitoring, and an embedded HTTP server for RESTful management.
• Security‑first – TLS/DTLS, SRTP support, and configurable authentication (digest, SIP 401, 407) with pluggable auth modules.

• Language: C (core), Lua/Python/Perl for scripting modules.
• Frameworks: Custom event‑driven engine; optional integration with OpenSSL for TLS, libuv for async I/O.
• Databases: Relational (MySQL/PostgreSQL/Oracle), NoSQL (Redis, MongoDB, Cassandra), LDAP/Radius.
• Build system: CMake with optional autotools fallback; modules compiled as shared objects (.so).

• Routing logic – configurable via kamailio.cfg, supporting regular expressions, routing tables, and dynamic lookup.
• Presence & IM – embedded XCAP server and MSRP relay for SIP‑SIMPLE instant messaging.
• Media gateway – SIP-to-PSTN gateways, media server integration via rtpengine or sngateway.
• Accounting & QoS – SIP‑C, CDR generation, and integration with external accounting systems.

Kamailio is lightweight (~30 MB binary) and can run on bare‑metal, VMs, or containers. It supports multi‑threaded UDP reception (since v6) and can be deployed behind HAProxy or LVS for high availability. Docker images are available on Docker Hub, and Helm charts simplify Kubernetes deployments. The server can be tuned for low‑latency environments (e.g., carrier‑grade networks) or embedded devices (e.g., IoT gateways).

• Plugin system – modules are loaded at runtime; developers can write custom modules in C or embed scripting languages.
• Webhooks & Callbacks – HTTP callbacks for events (e.g., registration, call start) enable integration with external services.
• API hooks – tm (transaction module), sl (session lock), and xmlrpc provide programmatic control over call flows.

• Configuration – human‑readable kamailio.cfg with extensive comments; configuration templates for common scenarios.
• Documentation – comprehensive online docs, wiki, and API reference; active community forums and mailing lists.
• Licensing – GPLv2+ for core, BSD for many modules; permissive enough to embed in commercial products while retaining open‑source benefits.

• Carrier SIP Trunking – high‑volume VoIP operators routing millions of calls with least‑cost logic.
• Enterprise PBX – lightweight front‑end to Asterisk or FreeSWITCH, providing fail‑over and load balancing.
• WebRTC Gateways – WebSocket/SCTP support for browser‑based real‑time communication.
• IoT Communication – lightweight SIP stack on edge devices for device discovery and control.

Kamailio offers unmatched performance thanks to its asynchronous engine, flexibility through a modular architecture, and extensibility via plugins and scripting. Its dual licensing model (GPLv2+ core, BSD modules) allows both open‑source and commercial deployments without legal friction. The strong community ensures rapid bug fixes, feature updates, and a wealth of third‑party integrations, making it the go‑to choice for developers building large‑scale SIP infrastructures.