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Tarun Kumar
Tarun Kumar

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How I Built a Cloud VoIP System from Scratch on AWS (Kamailio, Asterisk & RTPengine)

#voip #aws #kamailio #sip

🧠 Why I Built This

I wanted to understand how real-world VoIP systems work beyond theory — especially SIP signaling, call routing, and media handling.

Instead of using managed services, I built a complete VoIP system from scratch using open-source tools on AWS EC2.

🏗️ Architecture Overview

I designed a layered VoIP architecture:

  • Client Layer: Linphone (mobile), MicroSIP (desktop)
  • SIP Layer: Kamailio (handles registration & signaling)
  • PBX Layer: Asterisk (handles routing & IVR)
  • Media Layer: RTPengine (handles audio flow)
  • Database: MySQL (stores users)

⚙️ Tech Stack

  • Kamailio (SIP Proxy)
  • Asterisk (PBX)
  • RTPengine (Media Proxy)
  • MySQL (Database)
  • AWS EC2 (Cloud)
  • SIP Protocol

📞 How Call Flow Works

Example call:

3001 (Linphone) → Kamailio → Asterisk → 3003 (MicroSIP)
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  • Kamailio handles SIP signaling
  • Asterisk processes dialplan & routing
  • RTPengine handles RTP audio between endpoints

✅ What I Implemented

  • SIP user registration & authentication
  • Call routing between users
  • IVR system (extension 1000)
  • Two-way audio with RTP handling
  • NAT traversal using RTPengine

🧪 Challenges I Solved

🔧 One-way Audio

Problem: Audio only worked in one direction
Fix: Integrated RTPengine for proper RTP handling

🔧 SIP 401 Unauthorized

Problem: Users couldn't register
Fix: Corrected MySQL subscriber configuration

🔧 Port Conflicts

Problem: Kamailio & Asterisk both using 5060
Fix: Moved Asterisk to port 5062

🔧 NAT Issues

Problem: Calls connected but no audio
Fix: Fixed SDP and contact handling

📂 Project Repository

👉 https://github.com/Itztarunk51/voip-sip-platform

Includes:

  • Full configs
  • Troubleshooting guide
  • Command reference
  • Screenshots

📚 What I Learned

  • SIP protocol (INVITE, ACK, BYE, REGISTER)
  • Kamailio–Asterisk integration
  • RTP vs SIP and NAT traversal
  • Debugging using sngrep, tcpdump
  • Cloud deployment on AWS

🔮 Future Enhancements

  • WebRTC (browser-based calling)
  • Call queues & contact center features
  • Monitoring dashboard (CDR + logs)
  • TLS (SIPS) and SRTP security

🎯 Final Thoughts

This project helped me understand how real VoIP systems work in production environments.

If you're interested in networking, VoIP, or cloud — building something like this is incredibly valuable.

💬 Feel free to connect or share feedback!
linkedin : https://www.linkedin.com/in/tarun-morry-03a278225/

voip #aws #networking #sip #asterisk #kamailio #devops

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