Secure VoIP Infrastructure with Kamailio

Kamailio is an open-source SIP server used for handling high-volume voice, video, and messaging services in VoIP networks.

Kamailio is ideal for SIP signaling and routing at scale, while Asterisk/FreeSWITCH are better for media processing. Often, they are used together.

Yes, it’s built for carrier-grade usage and is widely used by telecom operators and VoIP wholesalers.

Yes, we set up WebRTC with secure signaling and media handling using RTPEngine or MediaProxy.

Absolutely. With IP whitelisting, DoS protection, TLS encryption, and SIP authentication, it ensures enterprise-level security.

Yes, we can link Kamailio with RADIUS, Diameter, or custom billing APIs for pre-paid/post-paid models.

We use tools like Prometheus, Grafana, and Kamailio’s built-in modules for monitoring, scaling, and logging.

Both are forks of the same base, but Kamailio is often preferred for low-level SIP routing and custom C modules, while OpenSIPS is known for its modular control.

Yes, we provide cloud-based deployments and managed Kamailio hosting with 24/7 support.

Yes, we offer complete support, security patching, version upgrades, and SLA-based managed services.