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Setting Up Basic SIP Servers with Kamailio: Beginner’s Guide

Setting Up Basic SIP Servers with Kamailio: Beginner’s Guide
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SIP servers play a key role in the world of Internet communication. They handle voice and video calls over the Internet, making them essential for any VoIP system. But setting up SIP servers can be tricky, especially for those new to the field. This is where Kamailio comes in. Kamailio is a powerful, open-source SIP server that is known for its flexibility and performance.

If you want to learn how to set up basic SIP servers using Kamailio, this article is the right place for you. It’ll walk you through each step. This makes sure you understand the process and avoid common mistakes, hence allowing you to create a working SIP server. But first, let’s take a quick look at the preconditions for installing basic SIP servers.

Prerequisites

Before setting up any SIP servers, you should acquire the basics of SIP and Kamailio. This helps you be better equipped to configure a server that meets your needs.

Basic Understanding of SIP Protocol

Basic understanding of SIP protocol

SIP stands for Session Initiation Protocol. It’s the method used to start, maintain, and end real-time communication sessions. These sessions could be voice calls, video calls, or messaging between two or more parties. 

SIP works by sending and receiving messages between different components of the SIP system. These components include the User Agent (UA), the Proxy Server, and the Registrar. 

The User-Agent is the device or application used by the person making or receiving a call. It could be a VoIP phone, a softphone app on a computer, or even a mobile app. The Proxy Server is like a middleman. It forwards requests from the User-Agent to the correct destination. This server also helps with routing calls and ensuring that the communication reaches the right place. The Registrar, meanwhile, is responsible for keeping track of where users are located. When a user logs in, the Registrar updates their location so that calls can reach them.

Kamailio plays a central role in this setup as well. It acts as the brain of the SIP infrastructure. Kamailio can handle tasks like routing, registration, and user authentication. This makes it a key piece of the puzzle. 

Kamailio’s architecture is modular, which means it can be customized to fit different needs. At its core, Kamailio uses modules to handle different functions. For instance, one module might manage user registrations, while another handles call routing. This flexibility allows you to build a SIP server that meets specific requirements, whether simple or complex.

Understanding these basics will help you navigate the setup process with Kamailio. With this foundation, you’re ready to start configuring your SIP server. 

System Requirements

System requirements

Before setting up a basic SIP server, it’s important to ensure that your system meets the necessary hardware and software requirements. The specific requirements can vary depending on the expected call volume, desired features, and overall complexity of your deployment. However, here’s a general guideline for a basic setup:

Hardware Requirements:

  • CPU: A single-core processor is enough for a small-scale deployment. But larger deployments with high call volume require a multi-core processor for good performance. 
  • RAM: At least 1GB of RAM is needed for a basic setup.
  • Disk Space: A few hundred megabytes of disk space is required for the Kamailio installation and configuration files. You can acquire extra space for logs, databases, and other data.
  • Network Interface: A network interface card (NIC) with sufficient bandwidth can handle expected call traffic.

Software Requirements:

  • Operating System: Kamailio runs primarily on Linux-based systems. These systems are highly preferred among developers for their flexibility and strong development environments. Ubuntu ranks as the third most popular system, following Windows and MacOS. Other Linux-based distributions include Debian, Fedora, and Arch. 
  • Kamailio: This is the core SIP server software. Make sure you download the latest stable version from the official Kamailio website. This ensures you have all the recent updates and security patches.
  • Database: While not strictly required for a basic setup, a database is often used to store user information, call records, and other data. Popular choices include MySQL, PostgreSQL, and SQLite.
  • Additional Software: Depending on your specific requirements, you might need additional tools and software like:
    • Web server (Apache, Nginx) for management interfaces.
    • Programming languages (Python, Lua) for custom scripts.
    • Development tools, including compilers (GCC, Clang), debuggers (GDB, LLDB), text editors (Nano, VS Code), networking tools (Wireshark, TCPdump), etc.

Preliminary Setup

Preliminary setup

Before diving into the setup of your SIP server, you should prepare your environment. This ensures that everything runs smoothly and that you don’t encounter unexpected issues later on. Below is what you can do:

  • Update Your System: Outdated software can cause problems. So, the first thing is to make sure your system is up-to-date. In other words, make sure that your system has the latest security patches and software versions. Remember to back up important data before performing a system update.
  • Install Necessary Packages: Your SIP server needs specific packages, or dependencies, to function properly. They’re compilers, SSL libraries, and other essential tools that lay the groundwork for Kamailio. 
  • Set Up a Non-Root User: This practice helps your SIP server avoid running as the root user for security reasons. It does so by creating a new user with sudo privileges to manage your server.
  • Configure Your Firewall: Next, ensure that your firewall is configured to allow traffic on the necessary ports. 
  • Install Git: Git is essential to download Kamailio from its source repository. It’s a version control system that eases the management of the Kamailio installation process. After installing Git, you can download Kamailio and proceed with the installation. 

5 Steps to Create Basic SIP Servers in Kamailio

5 steps to create basic sip servers in Kamailio

Once everything has been ready, let’s embark on the journey of building basic SIP servers in Kamailio. 

Step 1: Install Kamailio

The first step is to install Kamailio. Kamailio has different versions, each with its own set of features and stability. Choosing the right version of Kamailio is crucial for optimal performance and compatibility. 

If you’re setting up a basic SIP server, choose the latest stable version. This version is well-tested and has all the necessary features for a reliable server. If you’re looking for specific advanced features or cutting-edge updates, consider the latest development version. However, the stable version is recommended for most users, especially if you’re new to Kamailio.

Once you’ve chosen the right version, it’s time to install Kamailio by updating your packaging lists and adding the Kamailio repository. After installation, you should verify whether Kamailio has been correctly installed. You can do this by checking the version number and ensuring the service is running with the following commands:

Check the version:

  kamailio -v

Check the service status:

  sudo systemctl status kamailio

Step 2: Configure Kamailio for Basic SIP Servers

To configure Kamailio for basic SIP servers, you need to understand its configuration file, known as kamailio.cfg

Generally, Kamailio’s core functionality revolves around this file. As the heart of your SIP server, it guides how to handle SIP messages, monitor user accounts, and interact with the network. The configuration file is divided into key sections, including the module section, global parameters, and routing logic. Each is responsible for a specific aspect of the server’s operation.

To edit the configuration file for a basic SIP server setup, consider the following things:

1. Configure SIP Domains: First, define your SIP domain in the kamailio.cfg. The SIP domain is the part of the SIP address after the “@” symbol. For instance, if your SIP address is user@example.com, “example.com” is your SIP domain. You can define it like this:

 # Define your SIP domain

  alias="example.com"

2. Set Listening Ports: Next, configure the ports on which Kamailio will listen for SIP traffic. The default SIP port is 5060, but you can specify any port you prefer:

  listen=udp:127.0.0.1:5060

3. Network Interfaces: Also, you can specify the network interfaces that Kamailio will use. This tells Kamailio which IP addresses to bind to and listen for incoming SIP messages:

  listen=udp:eth0:5060

These basic settings establish the core functionality of your SIP server. This allows it to handle SIP requests and route them appropriately. 

With the basic configuration in place, you then can create SIP user accounts. These accounts enable users to register with your SIP server and make or receive calls. To create these accounts, add an entry to the kamailio.cfg file. For example:

 # Create a user account

  auth_user="user1"

  auth_pass="password123"

Each user needs a unique username and password. Further, you can manage user accounts by adding, removing, or updating entries in the configuration file. Kamailio can also connect to an external database to manage a large number of users more efficiently.

Once you’ve configured the basic settings and added user accounts, it’s important to test your setup. This helps check whether the configuration file has any syntax error or whether users can register with your SIP server.

Step 3: Set Up SIP Routing

The next step is to set up SIP routing. Briefly speaking, SIP routing is the process of directing SIP messages to their intended destinations. It’s a core function of any SIP server. 

Kamailio handles SIP routing by analyzing incoming SIP requests, like INVITEs or REGISTERs. It then determines where to send them based on predefined rules. These rules can be simple (e.g., directing all traffic to a single destination) or complex (e.g., involving numerous routes and conditions.

To set up SIP routing in Kamailio, you need to define routing rules in the `kamailio.cfg` file. Start by setting up a basic rule that routes all incoming SIP INVITE requests to a specific destination. Here’s the command you can add to the configuration file:

  # Route all INVITE requests to a specific destination

  if (method == "INVITE") {

    t_relay("sip:destination@domain.com");

  }

Next, consider SIP registrations. They allow users to register their location with the SIP server so they can receive calls. Kamailio can manage these registrations by storing them in a location database. You can handle SIP registrations by adding the following lines to the file:

  # Handle SIP REGISTER requests

  if (method == "REGISTER") {

    save("location");

    exit;

  }

In addition to routing and registrations, Kamailio can enable basic call features like call forwarding. To handle this feature, for example, you need to define a rule that checks for call forwarding settings and redirects calls as needed.

For more complex needs, Kamailio offers a wide range of advanced configuration features. They include load balancing, failover & redundancy, real-time monitoring and analytics, etc. You can enable these features by loading additional modules and writing more complex routing logic. For example, you can configure Kamailio to distribute incoming calls among multiple servers based on their current load.

Step 4: Secure Your SIP Server

The next important step is to secure your SIP server. Without proper safeguards, your server can be exposed to various threats, like unauthorized access or eavesdropping. To protect your SIP server, implement basic security measures. They include:

  • Transport Layer Security (TLS): TLS encrypts the communication between your SIP server and clients. This makes it harder for attackers to intercept and read messages. To enable TLS in Kamailio, configure the kamailio.cfg file and set up SSL certificates.
  • Secure Real-Time Transport Protocol (SRTP): SRTP adds an extra layer of encryption to the media streams (audio and video) in SIP calls. This prevents unauthorized parties from listening to the communication. You can enable SRTP by loading the appropriate module and configuring it in Kamailio.
  • Access Control Lists (ACLs): ACLs restrict access to your SIP server based on IP addresses or other criteria. This ensures that only trusted devices can connect to your server. 

Further, Kamailio tools for setting up logging and real-time monitoring. These tools track your server’s activity and identify potential security breaches. To enable logging, follow the command below:

  # Enable logging

  log_stderror = yes

  log_facility = LOG_LOCAL0

  ```

Meanwhile, you can use monitoring tools like siptrace or rtpengine to track SIP and RTP traffic in real time. 

As your SIP server grows, you may need more advanced security measures to address emerging threats and ensure ongoing protection. Some of them include Intrusion Detection and Prevention Systems (IDS/IPS), Dynamic Blacklisting, and Encryption of Sensitive Data.

Step 5: Test and Troubleshoot

After setting up your SIP server, testing its functionality is crucial. This ensures everything is working as expected. Some tools like SIPp or Wireshark can help you with this. SIPp allows you to simulate SIP traffic and check how your server performs. Meanwhile, Wireshark is a network protocol analyzer that lets you capture and analyze SIP traffic.

Even with careful setup and testing, you might encounter some common issues. Here’s how to address them:

  • Issue: SIP Clients Cannot Register:
    • Solution: Check if the SIP domain and ports are correctly configured in the Kamailio configuration file. Ensure that the firewall allows traffic on the SIP ports (usually 5060 for UDP/TCP, 5061 for TLS).
  • Issue: One-Way Audio During Calls:
    • Solution: This problem often arises from NAT issues. Verify that the NAT settings are correct in the Kamailio configuration and that the media relay is properly configured if needed.
  • Issue: High CPU Usage
    • Solution: High CPU usage can be due to an inefficient configuration or excessive SIP traffic. Review your routing rules, remove any unnecessary processing, and consider load balancing if the traffic is too high for a single server.

Once your SIP server is up and running, you may want to fine-tune the configuration for optimal performance. Accordingly, you can optimize routing rules to ensure they’re efficient, adjust timer values, and monitor server performance. All these practices can help your SIP server run smoothly and process growing demands over time.

Conclusion

In this guide, we walked through the steps to set up a basic SIP server with Kamailio. You started by understanding the prerequisites and installing Kamailio. Then, you configured the server, set up SIP routing, and ensured security. Finally, you tested and troubleshooted your setup to make sure everything worked as expected. 

Now that your SIP server is up and running, you might want to explore more advanced configurations. To do so, look into high availability setups, complex SIP routing, or even integrating additional features for a more robust system. 

If you want to expand your learning, check out the official Kamailio documentation, forums, and community. Also, subscribe to our blog to receive more info about SIP servers!

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