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Kamailio project preview

Kamailio

High-performance Open Source SIP Server capable of handling thousands of call setups per second.

LicenseGPL-2.0
GitHub stars3.2k
Last commit1 weeks ago
Tags5 topics
Telecom RoutingC Sip EngineDevopsSip ServerVoip Gateway
Overview

Why consider Kamailio?

Kamailio is an enterprise-grade open-source SIP server written in C. Powering telecom carriers, VoIP providers, and enterprise PBX backbones, Kamailio handles thousands of call routing setups per second with TLS encryption, WebRTC gateway bridging, and prepaid billing integration.

Guided learning

Learn Kamailio by building

Practical setup notes, real use cases, and copy-ready examples in one focused guide.

1 min read 2 sections
In this guide2 sections

Overview of Kamailio

Kamailio routes high-throughput voice and video SIP sessions with carrier-grade reliability.

Quickstart

bash
1# Start Kamailio SIP daemon
2kamailio -f /etc/kamailio/kamailio.cfg

Kamailio is licensed under the GNU General Public License v2.0 (GPL-2.0).

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Kamailio FAQs

Kamailio is listed as a Developer Tools tool on TiloBox. Review the overview, features, and official documentation on this page to decide whether it solves your specific workflow.

Start with the project's GitHub repository and official website for supported installation and deployment instructions. Test the setup with representative data or a small project before rolling it out more widely.

Kamailio is listed under the GPL-2.0 license. Read the complete license text and the project's notices before using, modifying, or distributing the software.

Production readiness depends on your requirements. Review maintenance activity, security practices, documentation, backup and upgrade procedures, and compatibility with your stack; then validate it in a non-production environment.

Kamailio is listed as an alternative to Cisco Unified SIP. Compare the core workflow, deployment model, integrations, and licensing against your must-have requirements before switching.