Scaling Asterisk Using Kamailio and RTPProxy: A Step-by-Step Guide

Efficiently Scaling VoIP Infrastructure: Load Balancing SIP and RTP Traffic with Kamailio and RTPProxy

Architecture of the Asterisk cluster setup using Kamailio and RTPProxy

Introduction:

In this article, we will walk you through the process of scaling an Asterisk cluster using Kamailio and RTPProxy. This solution provides efficient load balancing of SIP calls and media traffic across multiple Asterisk nodes, allowing you to achieve improved performance and high availability for your VoIP infrastructure.

Prerequisites:

• Two or more Asterisk nodes deployed using Docker (or any other deployment method)

• A dedicated server or container for running Kamailio and RTPProxy

Step 1: Install Kamailio and RTPProxy

Install Kamailio and RTPProxy on a separate server or container that will act as the load balancer for your Asterisk cluster.

On Debian-based systems, you can use the following command:

sudo apt-get update sudo apt-get install kamailio kamailio-mysql-modules rtpproxy

Step 2: Configure RTPProxy

Edit the RTPProxy configuration file (typically located at 

/etc/default/rtpproxy

USER="rtpproxy" GROUP="rtpproxy" PIDFILE="/var/run/rtpproxy/rtpproxy.pid" LOG_LEVEL="3" EXTRA_OPTS="-l <RTPProxy_IP_Address>"

Replace 

<RTPProxy_IP_Address>

Restart the RTPProxy service to apply the changes:

sudo systemctl restart rtpproxy

Step 3: Configure Kamailio

Edit the Kamailio configuration file (typically located at 

/etc/kamailio/kamailio.cfg

A minimal configuration example:

# Load required modules loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "textopsx.so" loadmodule "rtpproxy.so" loadmodule "dispatcher.so" loadmodule "nathelper.so"# Global parameters cfgengine "native" fork=yes children=4# RTPProxy settings modparam("rtpproxy", "rtpproxy_sock", "udp:<RTPProxy_IP_Address>:7722")# Dispatcher settings modparam("dispatcher", "list_file", "/etc/kamailio/dsipatcher.list")# Main request routing logic route { # Max forward check if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; } # Record routing for stateful processing record_route(); # Load balance using dispatcher if (!ds_select_dst("0", "4")) { sl_send_reply("500", "No destination"); exit; } # Engage RTPProxy for RTP traffic if (is_method("INVITE")) { force_rtp_proxy(); } else if (is_method("BYE")) { end_media_session(); } # Forward the request t_relay(); }

Replace 

<RTPProxy_IP_Address>

Create a dispatcher list file at 

/etc/kamailio/dispatcher.list

1 sip:<Asterisk_Node1_IP_Address>:5060 1 sip:<Asterisk_Node2_IP_Address>:5060

Replace 

<Asterisk_Node1_IP_Address>

<Asterisk_Node2_IP_Address>

Step 4: Restart Kamailio

Restart the Kamailio service to apply the changes:

sudo systemctl restart kamailio

Step 5: Update Asterisk configuration

Update the 

rtp.conf

[general] rtpstart=10000 rtpend=20000

Step 6: Point your SIP clients to Kamailio

Configure your SIP clients or devices to use the Kamailio load balancer as their SIP server. All SIP traffic should now be handled by Kamailio, which will load balance calls between your Asterisk nodes and use RTPProxy to handle RTP traffic.

Conclusion:

With this setup, Kamailio handles SIP signaling, allowing you to load balance and distribute calls across your Asterisk nodes. RTPProxy takes care of the RTP traffic, relaying media streams between your SIP clients and Asterisk nodes without the need to expose a large number of RTP ports on each container. This approach provides a more efficient and scalable solution for your Asterisk cluster, enabling you to achieve improved performance and high availability for your VoIP infrastructure.