Open Source Telecom Engineering
Build, deploy, and master modern communication systems with practical tutorials on Asterisk, SIP, WebRTC, Linux servers, cloud PBX, and AI-powered voice technologies.
Explore Technologies
Modern telecom engineering and open-source communication systems
Asterisk
Learn dialplans, PJSIP, ARI, AGI, queues, IVR systems, and production-grade PBX deployments.
SIP & VoIP
Master SIP signaling, RTP troubleshooting, codecs, NAT handling, and VoIP infrastructure.
WebRTC
Create browser-based calling systems using SIP.js and modern real-time communication.
Linux Servers
Deploy and manage Ubuntu, Debian, Nginx, Docker, and cloud telecom infrastructure.
Latest Tutorials
Real-world deployment and troubleshooting guides
Asterisk
Install Asterisk 22 on Ubuntu
Step-by-step installation and configuration guide for production-ready deployments.
SIP
Fix One-Way Audio Issues
Complete troubleshooting guide for NAT, RTP, and firewall-related audio problems.
WebRTC
Build WebRTC Calling
Create browser-based calling applications with SIP.js and WebRTC technologies.
Configuration Examples
Production-ready telecom commands and snippets
apt update && apt upgrade -y apt install asterisk nginx mariadb-server ufw allow 5060/udp ufw allow 10000:20000/udp [pjsip.conf] [transport-udp] type=transport protocol=udp bind=0.0.0.0
Platform Highlights
Practical telecom engineering resources
100+
Tutorials
50+
SIP Configurations
25+
WebRTC Guides
24/7
Learning Resources