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

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

Positive 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

Disclaimer: This job posting has been aggregated from external source. Role details, content, and availability are subject to change. Applicants are advised to confirm the latest information directly on the company website before applying.

📌 MID VoIP Developer (Ahmedabad)
🏢 Moon Technolabs
📍 Ahmedabad