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Learn How to Deploy Kamailio on Ubuntu VPS! This article provides a guide demonstrating how to deploy Kamailio on Ubuntu VPS. Kamailio is a high-performance SIP (Session Initiation Protocol) server used for VoIP infrastructures, SIP trunking, IMS deployments, SBCs, and large-scale telecommunications platforms. It can handle thousands of concurrent calls while maintaining low resource usage. This guide demonstrates how to deploy Kamailio 6.x on an Ubuntu VPS using the official Kamailio repositories. Prerequisites Before attempting to deploy Kamailio on Ubuntu VPS, ensure you have: Ubuntu 22.04 or Ubuntu 24.04 VPS Root or sudo access Public IP address Minimum: 2 GB RAM 2 CPU cores 20 GB storage Firewall access to open SIP ports How to Deploy Kamailio on Ubuntu VPS To deploy Kamailio on Ubuntu VPS, follow the steps outlined below: Update the Server Update all installed packages: sudo apt update sudo apt upgrade -y Install required utilities: sudo apt install -y curl gnupg2 wget Add the Official Kamailio Repository Import the Kamailio repository signing key: curl -fsSL https://deb.kamailio.org/kamailiodebkey.gpg | \ sudo gpg --dearmor -o /usr/share/keyrings/kamailio.gpg For Ubuntu 24.04 (Noble): echo "deb https://deb.kamailio.org/kamailio60 noble main" | \ sudo tee /etc/apt/sources.list.d/kamailio.list For Ubuntu 22.04 (Jammy): echo "deb https://deb.kamailio.org/kamailio60 jammy main" | \ sudo tee /etc/apt/sources.list.d/kamailio.list The Kamailio project maintains official repositories for Kamailio 6.x releases on Ubuntu and Debian systems. Update package indexes: sudo apt update Install Kamailio 6 Install the core package: sudo apt install -y kamailio Verify installation: kamailio -V Example output: version: kamailio 6.0.x Install Useful Kamailio Modules For most production deployments, install additional modules: sudo apt install -y \ kamailio-mysql-modules \ kamailio-tls-modules \ kamailio-websocket-modules \ kamailio-utils-modules These modules provide: Module Purpose MySQL User authentication and location storage TLS Secure SIP over TLS WebSocket WebRTC support Utils Administrative tools Configure Firewall Open SIP ports: sudo ufw allow 5060/udp sudo ufw allow 5060/tcp sudo ufw allow 5061/tcp Enable firewall: sudo ufw enable Verify: sudo ufw status Configure Kamailio Backup the default configuration: sudo cp /etc/kamailio/kamailio.cfg \ /etc/kamailio/kamailio.cfg.backup Edit configuration: sudo nano /etc/kamailio/kamailio.cfg Locate the listening directive and configure your VPS IP: listenudp:YOUR_SERVER_IP:5060 Example: listenudp:203.0.113.10:5060 Set server alias: alias"sip.example.com" Replace with your actual domain. Enable Database Support (Optional) Install MariaDB: sudo apt install -y mariadb-server Secure database: sudo mysql_secure_installation Edit Kamailio database settings: sudo nano /etc/kamailio/kamctlrc Configure: DBENGINEMYSQL DBHOSTlocalhost SIP_DOMAINsip.example.com Create Kamailio database: sudo kamdbctl create Follow the prompts to create: Database Tables Permissions Database-backed deployments are commonly used for subscriber authentication, registration storage, and accounting. Enable Authentication Modules Open: sudo nano /etc/kamailio/kamailio.cfg Enable the following definitions near the top: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB These enable: Database storage SIP authentication User location services Call accounting Validate Configuration Before restarting Kamailio: sudo kamailio -c Expected result: configuration file OK Fix any reported syntax errors before proceeding. Start Kamailio Enable service at boot: sudo systemctl enable kamailio Start service: sudo systemctl start kamailio Check status: sudo systemctl status kamailio You should see: Active: active (running) Kamailio includes native systemd service files for modern Ubuntu deployments. Create a SIP User Add a SIP account: sudo kamctl add testuser password123 Verify: sudo kamctl ul show Test SIP Registration Configure a SIP softphone such as: Zoiper Linphone MicroSIP Use: Username: testuser Password: password123 Domain: sip.example.com Port: 5060 Transport: UDP Attempt registration. Monitor logs: sudo journalctl -u kamailio -f Monitor SIP Traffic Install sngrep: sudo apt install -y sngrep Launch: sudo sngrep You can now watch: REGISTER requests INVITE messages SIP responses Call flows Enable SIP over TLS TLS encrypts SIP signaling traffic, protecting registrations, authentication credentials, and call setup messages. Install TLS Modules sudo apt install -y kamailio-tls-modules openssl Create TLS Certificates For production, use Let's Encrypt. Install Certbot: sudo apt install -y certbot Generate certificate: sudo certbot certonly --standalone -d sip.example.com Certificates will be stored in: /etc/letsencrypt/live/sip.example.com/ Create Kamailio TLS Configuration Create: sudo nano /etc/kamailio/tls.cfg Example: method TLSv1.2+ private_key /etc/letsencrypt/live/sip.example.com/privkey.pem certificate /etc/letsencrypt/live/sip.example.com/fullchain.pem ca_list /etc/ssl/certs/ca-certificates.crt verify_certificate no require_certificate no Enable TLS in Kamailio Edit: sudo nano /etc/kamailio/kamailio.cfg Enable: #!define WITH_TLS Add TLS listener: listentls:YOUR_SERVER_IP:5061 Load TLS module if not already loaded: loadmodule "tls.so" modparam("tls", "config", "/etc/kamailio/tls.cfg") Open Firewall sudo ufw allow 5061/tcp Validate Configuration sudo kamailio -c Restart Kamailio: sudo systemctl restart kamailio Verify: sudo ss -tlnp | grep 5061 Expected: LISTEN 0 128 0.0.0.0:5061 Configure Softphones Set: Transport: TLS Port: 5061 You should now see TLS registrations in: sudo sngrep Install and Configure RTPEngine RTPEngine provides media proxying for: NAT traversal SRTP support WebRTC Codec transcoding RTP anchoring Install RTPEngine Ubuntu package: sudo apt install -y rtpengine Configure Kernel Module Load: sudo modprobe xt_RTPENGINE Verify: lsmod | grep RTPENGINE Configure RTPEngine Edit: sudo nano /etc/rtpengine/rtpengine.conf Example: table 0 interface public/YOUR_PUBLIC_IP listen-ng 127.0.0.1:2223 port-min 10000 port-max 20000 foreground false Open RTP Ports sudo ufw allow 10000:20000/udp Start RTPEngine sudo systemctl enable rtpengine sudo systemctl restart rtpengine Verify: sudo systemctl status rtpengine Enable RTPEngine in Kamailio Install module: sudo apt install -y kamailio-extra-modules Verify: ls /usr/lib/x86_64-linux-gnu/kamailio/modules/rtpengine.so Add to kamailio.cfg: loadmodule "rtpengine.so" modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") Route SIP Traffic Through RTPEngine Inside request_route: if (is_method("INVITE")) { rtpengine_offer(); } if (is_method("BYE")) { rtpengine_delete(); } Inside onreply_route: if (status ~ "") { rtpengine_answer(); } Validate: sudo kamailio -c Restart: sudo systemctl restart kamailio Configure SIP Load Balancing with Dispatcher Kamailio's Dispatcher module allows SIP traffic distribution across multiple PBX servers. Example topology: Internet | Kamailio | +----+----+ | | Asterisk1 Asterisk2 Install Dispatcher Module Usually included: sudo apt install -y kamailio Verify: find /usr -name dispatcher.so Enable Dispatcher Add: loadmodule "dispatcher.so" modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") Create Backend List sudo nano /etc/kamailio/dispatcher.list Example: 1 sip:10.0.0.11:5060 1 sip:10.0.0.12:5060 1 sip:10.0.0.13:5060 Group 1 contains three SIP servers. Configure Round-Robin Routing Add to request_route: if (is_method("INVITE")) { if (!ds_select_dst("1", "4")) { send_reply("500", "No Destination"); exit; } t_relay(); exit; } Dispatcher algorithm values: Value Algorithm 0 Hash Call-ID 2 Hash From URI 4 Round Robin 8 Random 10 Weight Based Recommended: ds_select_dst("1", "4") for even distribution. Enable Backend Health Checks Add: modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_mode", 1) Kamailio will automatically probe servers and remove failed nodes from rotation. Reload Dispatcher Without Restart After editing dispatcher.list: kamcmd dispatcher.reload View status: kamcmd dispatcher.list Example: SET: 1 sip:10.0.0.11:5060 ACTIVE sip:10.0.0.12:5060 ACTIVE sip:10.0.0.13:5060 ACTIVE Recommended Production Architecture Internet | TLS 5061 / UDP 5060 | +----------------+ | Kamailio | | Load Balancer | +----------------+ | RTPEngine | +---------------+---------------+ | | +--------------+ +--------------+ | Asterisk #1 | | Asterisk #2 | +--------------+ +--------------+ | | SIP Users SIP Users This architecture provides: SIP over TLS encryption RTP media anchoring via RTPEngine NAT traversal SIP load balancing High availability Horizontal scaling for thousands of concurrent calls WebRTC readiness (when combined with WSS and SRTP) Useful Commands Restart Kamailio: sudo systemctl restart kamailio Reload configuration: sudo kamcmd cfg.reload Check running processes: ps aux | grep kamailio View logs: sudo journalctl -u kamailio Next Steps Once Kamailio is running successfully, consider adding: WebRTC support Dispatcher module Failover routing SBC functionality FreeSWITCH or Asterisk integration These features transform Kamailio from a basic SIP proxy into a carrier-grade VoIP platform capable of handling large-scale deployments. Conclusion You now know how to deploy Kamailio on Ubuntu VPS. Related: How to Install and Deploy FusionPBX on Debian VPS 🚀 How to Deploy Issabel PBX on Rocky Linux VPS

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