Develop, customize, and maintain VoIP applications using Asterisk, FreeSWITCH, or other SIP-based platforms
Configure and troubleshoot SIP trunks, gateways, extensions, IVRs, call routing, queues, and conferencing solutions
Develop and integrate REST APIs, WebRTC applications, and third-party communication services
Build and maintain telephony features such as call recording, voicemail, call forwarding, transfers, and reporting
Diagnose and resolve SIP signaling, RTP, codec, NAT, and media-related issues
Collaborate with frontend, backend, DevOps, and QA teams to deliver reliable communication solutions
Perform system monitoring, log analysis, and performance tuning for VoIP platforms
Participate in code reviews, technical documentation, and solution design discussions
Ensure system security, scalability, and high availability for production deployments
Stay updated with emerging VoIP technologies and industry best practices
Strong experience with Asterisk, FreeSWITCH, or similar IP PBX platforms
Solid understanding of SIP, RTP, SRTP, SDP, WebRTC, T.38, and DTMF
Experience configuring SIP trunks, gateways, DIDs, IVRs, queues, ring groups, and call routing
Good understanding of NAT traversal, STUN/TURN, and media flow troubleshooting
Knowledge of Kamailio or OpenSIPS is an advantage
Experience with Node.js, PHP, or Python for VoIP integrations and backend development
Experience building and consuming RESTful APIs.
Familiarity with AGI, AMI, ESL, ARI, or event-driven telephony integrations
Experience integrating communication platforms such as Twilio, Telnyx, Plivo, or similar providers is a plus
Experience with MySQL, PostgreSQL, or MongoDB
