Kamailio SIP Switch: Beginner to Pro

Categories: Telephony Systems
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About Course

Master Kamailio, the high-performance open-source SIP server, from your first configuration to carrier-grade deployments. This course covers all the core Kamailio modules plus integrations with Asterisk, FreeSWITCH, VICIdial, WebRTC and external APIs.

What Will You Learn?

  • Understand SIP and how Kamailio routes it
  • Write and structure kamailio.cfg routing logic
  • Registration, authentication and database backends
  • NAT traversal and media relay with RTPengine
  • Load balancing, failover and least cost routing
  • Security: anti-flood, fraud prevention and TLS
  • WebRTC, presence and messaging
  • Integrate with Asterisk, FreeSWITCH, VICIdial and APIs
  • Build highly available, scalable SIP platforms

Course Content

Introduction to VoIP, SIP and Kamailio

Installation and kamailio.cfg structure

SIP routing logic and request handling

Registration and user location (registrar, usrloc)

Authentication and database backends (auth_db)

NAT traversal (nathelper) and media with RTPengine

Load balancing and failover (dispatcher)

Least cost and carrier routing (lcr, drouting)

Dialogs, accounting and CDRs (dialog, acc)

Security: anti-flood, fraud prevention and TLS (pike, htable, tls)

WebRTC with WebSockets

Presence, messaging and push notifications

Integrations: Asterisk, FreeSWITCH, VICIdial and APIs (JSON-RPC, HTTP)

High availability, scaling and monitoring

Final project: a carrier-grade SIP platform