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.
Puede gestionar cientos de miles de llamadas simultáneas sin sudar.
Procesa las decisiones de encaminamiento de llamadas en microsegundos
Utilizado por las principales empresas de telecomunicaciones del mundo
No hay que pagar licencias tan caras como las alternativas propietarias
Puede adaptarse a las necesidades específicas de la empresa
100% compliant to RFC rules, easy to integrate
Si necesita fiabilidad hoy y capacidad de ampliación mañana, ¡hablemos!