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

Requirements

Solid 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

📌 Sr. Asterisk / FreeSWITCH VoIP Developer (India)
🏢 Moon Technolabs
📍 India