CampusTalk: A Secure VoIP Platform using Asterisk

🚀 Project Overview

CampusTalk is a secure VoIP communication platform built using Asterisk, designed to enhance internal organizational communication. This project aims to provide a cost-effective, feature-rich, and highly secure VoIP solution to address the challenges associated with traditional PBX systems, such as high costs, limited scalability, and security vulnerabilities.

By leveraging Asterisk’s open-source telephony framework, CampusTalk enables SIP-based voice communication, voicemail services, IVR, call queuing, and advanced security mechanisms, ensuring reliable and encrypted voice transmission over the network.

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🔧 Key Features

✅ SIP-based VoIP Calls

• Users register as SIP clients and communicate over a secure network.
• Supports multiple SIP clients (Zoiper5, CSipSimple, Softphone) for cross-platform communication.

✅ Multi-server Architecture

• Two Asterisk servers deployed on Ubuntu and Kali Linux simulate a distributed VoIP infrastructure.
• Enables cross-network communication between different servers while ensuring low latency.

✅ Advanced Security Mechanisms

• TLS encryption & SRTP protocol ensure end-to-end secure voice transmission.
• MD5 hashing is used for secure SIP authentication.
• Firewall and access control rules protect against unauthorized access and SIP attacks.

✅ Voicemail & IVR

• Custom voicemail system with email notifications for missed calls.
• Interactive Voice Response (IVR) for automated call handling, allowing users to navigate options.

✅ Call Routing & Queuing

• Custom call forwarding and call transfer (blind & attended) improve call management.
• Call queuing system ensures efficient customer support handling.
• Time-based call restrictions for enhanced control over call availability.

✅ Scalability & Optimization

• Designed for low-latency communication with efficient call routing and media transmission using SIP and RTP protocols.
• Supports multiple concurrent connections while maintaining high availability.

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🖥️ Technologies Used

• Asterisk 20.2.0 – Open-source VoIP framework
• Ubuntu 20.2 & Kali Linux – Server deployment
• Zoiper5, CSipSimple, Softphone – SIP clients for testing
• MD5, TLS, SRTP – Encryption & security protocols
• Bash scripting & Asterisk configuration – Custom call handling

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📌 Installation & Setup

🔹 Prerequisites

1. Linux-based operating system (Ubuntu 20.2 or Kali Linux)
2. Asterisk 20.2.0 installed
3. SIP clients (Zoiper5, CSipSimple, Softphone)
4. Network configuration (firewall and routing setup)

🔹 Step-by-Step Installation

1. Install Asterisk:

   sudo apt update && sudo apt upgrade -y
   sudo apt install asterisk -y

2. Configure SIP Clients:
   • Add users to sip.conf:

   [natnael]
   type=peer
   context=phones
   allow=ulaw,alaw
   md5secret=your_md5_hashed_password
   host=dynamic
   transport=udp,tls,tcp

3. Configure Call Routing in extensions.conf:

   [phones]
   exten => 100,1,Dial(SIP/natnael,10,T)
   same => n,Voicemail(${EXTEN},b)

4. Enable Security Measures:

   sudo ufw allow 5060/udp
   sudo ufw allow 10000:20000/udp

5. Start Asterisk CLI:

   sudo asterisk -rvvv

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📌 Code Structure & Configuration Files

📂 Key Configuration Files

• sip.conf: Defines SIP clients and servers.
• extensions.conf: Manages call routing and dialing rules.
• voicemail.conf: Handles voicemail services and email notifications.
• queues.conf: Configures call queuing for customer support.
• features.conf: Defines call transfer, forwarding, and security policies.

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📌 Testing & Validation

✅ Functional Testing

• Registration Test: Ensured successful SIP client registration using sip show peers.
• Call Test: Verified successful call establishment between users.
• Voicemail Test: Checked voicemail recording and email notifications.
• IVR Test: Simulated menu selection in the interactive voice response system.

✅ Security Testing

• SIP Authentication: Tested MD5 encryption and TLS secure connections.
• Penetration Testing: Used Kali Linux tools to assess security vulnerabilities.
• Firewall Rules: Ensured restricted access to unauthorized users.

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📌 Future Enhancements

🔹 AI-based Call Analytics: Implement machine learning models to analyze call patterns and optimize network performance.

🔹 Cloud Deployment: Host the platform on AWS or Azure for better scalability and global accessibility.

🔹 Integration with WebRTC: Enable browser-based VoIP calling for improved accessibility.

🔹 Enhanced Security Measures: Implement end-to-end encryption with Zero Trust Architecture.

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Authors

📌 Horieb Mesfun

📌 Natnael Haile

📌 Siem Hagos