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

🧠 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) 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