The latest articles on DEV Community by Tarun Kumar (@itztarunk51). https://dev.to/itztarunk51 https://media2.dev.to/dynamic/image/width=90,height=90,fit=cover,gravity=auto,format=auto/https:%2F%2Fdev-to-uploads.s3.amazonaws.com%2Fuploads%2Fuser%2Fprofile_image%2F3865680%2Fc832bb2c-af1c-4493-8b5c-913c94d2a7fa.jpg https://dev.to/itztarunk51 en Tarun Kumar Tue, 07 Apr 2026 12:43:26 +0000 https://dev.to/itztarunk51/how-i-built-a-cloud-voip-system-from-scratch-on-aws-kamailio-asterisk-rtpengine-5b24 https://dev.to/itztarunk51/how-i-built-a-cloud-voip-system-from-scratch-on-aws-kamailio-asterisk-rtpengine-5b24 <h2> 🧠 Why I Built This </h2> <p>I wanted to understand how real-world VoIP systems work beyond theory — especially SIP signaling, call routing, and media handling.</p> <p>Instead of using managed services, I built a complete VoIP system from scratch using open-source tools on AWS EC2.</p> <h2> 🏗️ Architecture Overview </h2> <p>I designed a layered VoIP architecture:</p> <ul> <li> <strong>Client Layer:</strong> Linphone (mobile), MicroSIP (desktop)</li> <li> <strong>SIP Layer:</strong> Kamailio (handles registration &amp; signaling)</li> <li> <strong>PBX Layer:</strong> Asterisk (handles routing &amp; IVR)</li> <li> <strong>Media Layer:</strong> RTPengine (handles audio flow)</li> <li> <strong>Database:</strong> MySQL (stores users)</li> </ul> <p><a href="https://media2.dev.to/dynamic/image/width=800%2Cheight=%2Cfit=scale-down%2Cgravity=auto%2Cformat=auto/https%3A%2F%2Fdev-to-uploads.s3.amazonaws.com%2Fuploads%2Farticles%2Fw73b8d93nw4r31by73is.png" class="article-body-image-wrapper"><img src="https://media2.dev.to/dynamic/image/width=800%2Cheight=%2Cfit=scale-down%2Cgravity=auto%2Cformat=auto/https%3A%2F%2Fdev-to-uploads.s3.amazonaws.com%2Fuploads%2Farticles%2Fw73b8d93nw4r31by73is.png" alt=" " width="800" height="436"></a></p> <h2> ⚙️ Tech Stack </h2> <ul> <li>Kamailio (SIP Proxy)</li> <li>Asterisk (PBX)</li> <li>RTPengine (Media Proxy)</li> <li>MySQL (Database)</li> <li>AWS EC2 (Cloud)</li> <li>SIP Protocol</li> </ul> <h2> 📞 How Call Flow Works </h2> <p>Example call:<br> </p> <div class="highlight js-code-highlight"> <pre class="highlight plaintext"><code>3001 (Linphone) → Kamailio → Asterisk → 3003 (MicroSIP) </code></pre> </div> <ul> <li>Kamailio handles SIP signaling</li> <li>Asterisk processes dialplan &amp; routing</li> <li>RTPengine handles RTP audio between endpoints</li> </ul> <p><a href="https://media2.dev.to/dynamic/image/width=800%2Cheight=%2Cfit=scale-down%2Cgravity=auto%2Cformat=auto/https%3A%2F%2Fdev-to-uploads.s3.amazonaws.com%2Fuploads%2Farticles%2F970ika70ykc2mep8kasu.png" class="article-body-image-wrapper"><img src="https://media2.dev.to/dynamic/image/width=800%2Cheight=%2Cfit=scale-down%2Cgravity=auto%2Cformat=auto/https%3A%2F%2Fdev-to-uploads.s3.amazonaws.com%2Fuploads%2Farticles%2F970ika70ykc2mep8kasu.png" alt=" " width="800" height="414"></a></p> <p><a href="https://media2.dev.to/dynamic/image/width=800%2Cheight=%2Cfit=scale-down%2Cgravity=auto%2Cformat=auto/https%3A%2F%2Fdev-to-uploads.s3.amazonaws.com%2Fuploads%2Farticles%2Fqgmzwkskgeosla30bml0.png" class="article-body-image-wrapper"><img src="https://media2.dev.to/dynamic/image/width=800%2Cheight=%2Cfit=scale-down%2Cgravity=auto%2Cformat=auto/https%3A%2F%2Fdev-to-uploads.s3.amazonaws.com%2Fuploads%2Farticles%2Fqgmzwkskgeosla30bml0.png" alt=" " width="800" height="374"></a></p> <h2> ✅ What I Implemented </h2> <ul> <li>SIP user registration &amp; authentication</li> <li>Call routing between users</li> <li>IVR system (extension 1000)</li> <li>Two-way audio with RTP handling</li> <li>NAT traversal using RTPengine</li> </ul> <h2> 🧪 Challenges I Solved </h2> <h3> 🔧 One-way Audio </h3> <p><strong>Problem:</strong> Audio only worked in one direction<br> <strong>Fix:</strong> Integrated RTPengine for proper RTP handling</p> <h3> 🔧 SIP 401 Unauthorized </h3> <p><strong>Problem:</strong> Users couldn't register<br> <strong>Fix:</strong> Corrected MySQL subscriber configuration</p> <h3> 🔧 Port Conflicts </h3> <p><strong>Problem:</strong> Kamailio &amp; Asterisk both using 5060<br> <strong>Fix:</strong> Moved Asterisk to port 5062</p> <h3> 🔧 NAT Issues </h3> <p><strong>Problem:</strong> Calls connected but no audio<br> <strong>Fix:</strong> Fixed SDP and contact handling</p> <h2> 📂 Project Repository </h2> <p>👉 <a href="https://github.com/Itztarunk51/voip-sip-platform" rel="noopener noreferrer">https://github.com/Itztarunk51/voip-sip-platform</a></p> <p>Includes:</p> <ul> <li>Full configs</li> <li>Troubleshooting guide</li> <li>Command reference</li> <li>Screenshots</li> </ul> <h2> 📚 What I Learned </h2> <ul> <li>SIP protocol (INVITE, ACK, BYE, REGISTER)</li> <li>Kamailio–Asterisk integration</li> <li>RTP vs SIP and NAT traversal</li> <li>Debugging using sngrep, tcpdump</li> <li>Cloud deployment on AWS</li> </ul> <h2> 🔮 Future Enhancements </h2> <ul> <li>WebRTC (browser-based calling)</li> <li>Call queues &amp; contact center features</li> <li>Monitoring dashboard (CDR + logs)</li> <li>TLS (SIPS) and SRTP security</li> </ul> <h2> 🎯 Final Thoughts </h2> <p>This project helped me understand how real VoIP systems work in production environments.</p> <p>If you're interested in networking, VoIP, or cloud — building something like this is incredibly valuable.</p> <p>💬 Feel free to connect or share feedback!<br> linkedin : <a href="https://www.linkedin.com/in/tarun-morry-03a278225/" rel="noopener noreferrer">https://www.linkedin.com/in/tarun-morry-03a278225/</a></p> <h1> voip #aws #networking #sip #asterisk #kamailio #devops </h1> voip aws kamailio sip