Sr. Asterisk / FreeSWITCH VoIP Developer

? Join the Innovation at Moon Technolabs !!

We're looking for a skilled Sr. Asterisk / FreeSWITCH VoIP Developer to build, optimize, and manage scalable VoIP communication systems while ensuring high performance and reliable connectivity.

? Asterisk / FreeSWITCH VoIP Developer - 5+ Yrs

? Location: Sola, Ahmedabad

? Work Mode: Onsite Only

Primary Responsibilities

• Troubleshoot SIP signaling, RTP media flow, one-way/no-audio issues, codec negotiation, jitter, latency, packet loss, and NAT traversal problems.
• Perform advanced VoIP and network troubleshooting using tools like sngrep, tcpdump, Wireshark, RTP analysis, traceroute, ping, and network monitoring tools.
• Design, develop, and maintain scalable VoIP and telephony solutions using Asterisk and/or FreeSWITCH.
• Configure and manage SIP trunking, dial plans, IVR systems, call routing, conferencing, and PBX features.
• Develop and optimize real-time voice communication systems for high availability and low latency.
• Integrate telephony systems with CRMs, APIs, third-party services, and AI-based voice platforms.
• Troubleshoot SIP signaling, RTP media flow, NAT traversal, codec negotiation, and call quality issues.
• Implement and maintain WebRTC-based communication solutions and SIP over WebSocket (WSS).
• Develop custom modules, AGI/ESL scripts, and automation tools for telephony workflows.
• Monitor system performance, identify bottlenecks, and optimize server scalability and reliability.
• Work with DevOps and infrastructure teams for deployment, monitoring, and production support.
• Participate in requirement analysis, architecture planning, estimation, and technical documentation.
• Mentor junior developers and support technical decision-making across VoIP projects.
• Ensure system security, failover mechanisms, and compliance with communication standards.

Technical Requirements

• Strong hands-on experience with Asterisk, FreeSWITCH, FusionPBX, or FreePBX.
• Deep understanding of SIP, RTP, SRTP, WebRTC, ICE, STUN/TURN, and NAT traversal.
• Experience in configuring SIP trunks, SBCs, gateways, IVRs, queues, and call center solutions.
• Proficiency in scripting/programming using Python, Lua, PHP, Node.js, or Bash.
• Experience with ESL (Event Socket Library), AGI, AMI, or ARI integrations.
• Knowledge of VoIP troubleshooting tools such as sngrep, tcpdump, Wireshark, and RTP analysis.
• Experience with Linux server administration (Ubuntu/CentOS/Debian).
• Familiarity with Kamailio/OpenSIPS and SIP proxy concepts is a plus.
• Experience integrating telephony with cloud providers like Twilio, Telnyx, Plivo, or similar.
• Knowledge of databases such as MySQL/PostgreSQL and API integrations.
• Understanding of containerization and deployment using Docker/Kubernetes is preferred.
• Familiarity with CI/CD and monitoring tools for production systems.