Configure and troubleshoot SIP trunks, gateways, extensions, IVRs, call routing, queues, and conferencing solutions

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

Requirements:
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

Valuable 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

📌 Mid Voip Developer (Ahmedabad)
🏢 Moon Technolabs
📍 Ahmedabad