The Challenge#
Traditional telephony systems are expensive and rigid. I needed to build a communication system that was cost-effective, programmable, and capable of handling complex routing without physical hardware constraints. The goal was to eliminate traditional hardware landlines in favor of a flexible, cloud-integrated solution.
The Solution#
Designed and deployed a scalable Voice over IP (VoIP) telecommunication system bridging local infrastructure with the global telephone network (PSTN). I engineered a hybrid solution using FreePBX (on top of Asterisk) as the internal Private Branch Exchange (PBX) and Twilio Elastic SIP as the SIP Trunking provider.
Key Features#
Twilio SIP Trunking:
- Configured Twilio Elastic SIP Trunks to handle inbound/outbound calls, enabling global connectivity with low latency.
- Configured the Termination and Origination URIs to route calls dynamically.
- Implemented IP Access Control Lists (ACLs) for security.
PBX Management:
- Deployed FreePBX (on top of Asterisk) to manage internal extensions, call routing, and IVR (Interactive Voice Response) menus.
- Set up inbound routes to direct customer calls to specific departments via IVR (“Press 1 for Support”).
- Configured outbound routes with failover logic to ensure call reliability.
Softphone Integration:
- Configured endpoint connectivity for desktop and mobile clients (Zoiper/MicroSIP) allowing users to make secure calls from anywhere.
Security Hardening:
- Implemented fail2ban and configured firewall rules (iptables) to prevent SIP brute-force attacks and unauthorized toll fraud.
- Optimized SIP packets (UDP/TCP) for low latency voice transmission.
- Secured the transmission using TLS and SRTP encryption to prevent eavesdropping.
Technologies Used#
- Core: FreePBX, Asterisk
- Cloud Provider: Twilio (Elastic SIP)
- OS: Linux (CentOS/Debian)
- Protocols: SIP, RTP, TCP/UDP
- Softphones: Zoiper, MicroSIP
- Security: fail2ban, iptables, TLS, SRTP