Sr. Asterisk / FreeSWITCH VoIP Developer

Moon Technolabs Ahmedabad, Gujarat, Ahmedabad, Gujarat, IN (On-site) 1 week ago
Full Time/Permanent
Mid-Senior level

Job Description

? 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.

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