The industry’s standard, carrier grade SIP Server
Kamailio is a high-performance, open-source SIP server (proxy/registrar/redirect) that powers carrier-grade VoIP, SBC, and real-time communications platforms.
Built for massive scale and low latency, it efficiently routes SIP traffic for PBXs, softswitches, WebRTC gateways, and IMS/VoLTE deployments—while keeping operational costs down.
Extreme scalability & reliability: Proven to handle very high CPS (calls and millions of registrations with minimal memory footprint.
Flexible routing logic: A powerful, scriptable engine plus 200+ modules (dispatcher, tm, permissions, dialog, pike, nathelper, rtjson, etc.) enable nuanced call flows and policy control.
Enterprise-grade security: TLS/SRTP, topology hiding, rate limiting/DoS mitigation, ACLs, authentication, and fraud-prevention features out of the box.
Resilience & load balancing: Advanced failover, health checks, clustering, and horizontal scaling for always-on services.
Extensible by design: Integrates with databases and external apps; extend in Lua, Python, or JavaScript for custom logic and AI-driven workflows.
WebRTC & SIP interoperability: Standards-compliant, making it easy to bridge browsers, SIP trunks, and heterogeneous networks.
Choose Kamailio to build fast, secure, and scalable real-time communications—ready for today’s VoIP and tomorrow’s RTC innovation.
Can handle hundreds of thousands of simultaneous calls without breaking a sweat
Processes call routing decisions in microseconds
Used by major telecommunications companies worldwide
No expensive licensing fees like proprietary alternatives
Can be tailored to specific business requirements
100% compliant to RFC rules, easy to integrate
If you need reliability today and room to scale tomorrow, let’s talk!