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.