Kamailio SIP Server on Azure

Deploy a production-ready Kamailio SIP server on Microsoft Azure in minutes. Built for VoIP engineers, Internet Telephony Service Providers (ITSPs) and platform teams that need carrier-grade SIP routing, registration and load balancing in front of media servers such as Asterisk or FreePBX, without spending days compiling modules, configuring database backends or hardening the operating system.

Why Kamailio, and Why This Image

Kamailio, formerly known as OpenSER, is one of the industry's leading open-source SIP proxy and registrar platforms, capable of routing thousands of SIP transactions per second on modest hardware. It is widely deployed as a SIP registrar, proxy layer and least-cost routing frontend for carrier and enterprise VoIP platforms.

Deploying Kamailio securely requires assembling modules, configuring database connectivity and securing the operating system before exposing SIP services to the Internet. This image provides a hardened, production-ready deployment from first boot, recognising that Internet-facing SIP services are routinely scanned within minutes of becoming available.

What You Get Out of the Box

• Latest stable Kamailio release installed from the official project repositories.
• Commonly used modules enabled, including registrar, usrloc, dispatcher, dialog, TLS, WebSocket and database connectivity.
• WebSocket support prepared for WebRTC gateway deployments.
• Security-focused defaults including Fail2ban SIP scanner protection, Pike module rate limiting and a firewall configuration exposing only the ports you explicitly enable.
• kamctl and kamcmd management utilities installed, configured and documented.

Common Use Cases

• SIP Load Balancer – Distribute SIP registrations and call traffic across Asterisk or FreePBX clusters using the Dispatcher module.
• Registrar and Proxy Layer – Accept SIP registrations at the network edge while keeping backend media servers private.
• ITSP Platform – Provide routing, authentication, registration and rate limiting for service provider infrastructure.
• WebRTC Gateway – Support WebSocket SIP connections from browsers while forwarding standard SIP traffic to backend systems.
• Geographically Redundant Deployments – Build active-active SIP proxy infrastructures across multiple Azure regions using Kamailio's stateless architecture.

Sizing Guidance

Kamailio is highly efficient because it processes SIP signalling rather than RTP media. Production sizing depends primarily on registration volume, routing table size and concurrent SIP transactions rather than CPU-intensive media processing.

• Evaluation and small deployments: Standard_B2s (2 vCPU, 4 GB RAM)
• Production SIP proxy or registrar: Standard_D2s_v5 (2 vCPU, 8 GB RAM)
• High registration churn, large routing tables or carrier-scale deployments: Standard_D8s_v5 (8 vCPU, 32 GB RAM). Larger deployments typically benefit more from additional memory than CPU cores.

Trademark Notice

Kamailio® is a registered trademark of its respective trademark holder. This product is independently packaged and maintained and is not affiliated with or endorsed by the Kamailio project.