Wish to scale your VoIP system? One of the best methods to have high performance and flexibility is through Kamailio and Asterisk integration. Kamailio is a high-performance SIP proxy, and Asterisk is the media server and PBX system. When used together, they provide an efficient and scalable SIP routing solution for any business size.
we’ll walk through how to integrate Kamailio with Asterisk and highlight the key benefits of this architecture.
1. Why Combine Kamailio and Asterisk?
Asterisk is great at handling calls, IVRs, conferencing, and voicemail. However, when it comes to handling thousands of simultaneous SIP registrations and routing, Kamailio steps in. Kamailio and Asterisk integration helps offload SIP signaling and routing from Asterisk, improving performance and scalability.
2. Getting to Know SIP Roles in Integration
In a standard Kamailio Asterisk SIP proxy configuration, Kamailio processes all incoming SIP requests. Kamailio determines where calls are forwarded—either to Asterisk for handling or directly to endpoints. This separates the signaling from the media processing, making it more efficient.
3. Install and set up Kamailio
To start SIP routing with Kamailio and Asterisk, install Kamailio on your server:
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sudo apt-get install kamailio kamailio-mysql-modules
Configure the kamailio.cfg file to set up SIP domains, user authentication, and forwarding logic. Ensure Kamailio is set to forward SIP INVITE messages to your Asterisk server.
4. Asterisk Configuration for Kamailio Integration
Next, configure Asterisk to accept requests from Kamailio. You’ll need to adjust the sip.conf (or pjsip.conf for newer versions) to trust Kamailio's IP and handle SIP messages properly.
Example in sip.conf:
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[kamailio]
type=peer
host=kamailio_server_ip
context=from-kamailio
This is a critical step for Kamailio and Asterisk integration to seamlessly work together.
5. Secure the Communication
To prevent SIP attacks or abuse, set up firewalls to permit only Kamailio and trusted endpoints. You can also use TLS and SRTP for secure signaling and media.
6. Test the SIP Routing
Once you have your setup going, test the SIP signaling path by having users register with Kamailio and initiating a call that goes through to Asterisk. You can trace SIP packets using sngrep and see everything go down as expected.
7. Monitoring and Maintenance
Use utilities such as kamctl, Asterisk CLI, and system logs to trace call flows and debug problems. Keeping both Kamailio and Asterisk up-to-date ensures your system is secure and stable.
Final Thoughts
You now know how to combine Kamailio with Asterisk for reliable SIP routing. This design harnesses the best of both worlds—Kamailio for processing thousands of SIP messages per second, and Asterisk for full-featured telephony.
Whether you’re building a commercial VoIP service or enhancing your internal PBX, SIP routing with Kamailio and Asterisk offers unmatched scalability, flexibility, and per
