MID VoIP Developer

Role Overview

Develop, customize, and maintain VoIP applications using Asterisk, FreeSWITCH, or other SIP-based platforms. You will be responsible for building reliable communication solutions, integrating third-party services, and ensuring high availability for production deployments.

Responsibilities

• 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
• 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
• 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
• 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, or Plivo is a plus
• Experience with MySQL, PostgreSQL, or MongoDB

Skills

• Asterisk
• FreeSWITCH
• SIP
• WebRTC
• Node.js