Kamailio
High-performance SIP server for VoIP and real-time communications
Kamailio is an open source SIP signaling server used to build and scale VoIP and real-time communication platforms. It focuses on high performance and flexibility, and is commonly deployed as a SIP proxy, registrar, edge proxy, or signaling core in carrier and enterprise environments.
Key Features
- High-throughput SIP routing and proxying suitable for large deployments
- SIP load balancing, failover routing, and least-cost routing capabilities
- Registrar and location service for user registration and contact lookup
- Security and access control features, including authentication/authorization and TLS
- WebSocket support for WebRTC signaling
- Extensible modular architecture with integrations for multiple backend systems
- Support for IPv4 and IPv6 and various SIP transport options (UDP/TCP and more)
Use Cases
- Scaling SIP PBX, SIP-to-PSTN gateways, or media servers with a dedicated signaling layer
- Building carrier-grade SIP routing cores, including IMS/VoLTE-oriented deployments
- Deploying SIP edge proxy and SIP firewall-style protection in front of RTC infrastructure
Limitations and Considerations
- Provides SIP signaling (call control) and typically requires external media servers for RTP/media handling
- Configuration is powerful but can be complex, especially for advanced routing and multi-module setups
Kamailio is a mature SIP server with continuous development since 2001 and a large ecosystem. It is well-suited for operators and organizations that need a robust, customizable signaling component for VoIP, WebRTC, and unified communications.
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Tech Stack:
PostgreSQL
Redis
Cassandra
GNU Make
C
Linux
MariaDB
CMake
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