Kamailio

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.