Talentxo
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Network EngineeringDevOps & Cloud

Telephony Engineer (FreeSWITCH, VoIP )

On-site · Gurugram
Posted August 4, 2026
Full-time

Overview

This role is for a Telephony Engineer with 4-10 years of experience to design, deploy, configure, integrate, and maintain FreeSWITCH-based VoIP platforms in production. The engineer will own call flows, dialplans, IVR, SIP/RTP media handling, WebRTC support, and external integrations while ensuring high availability, call quality, and scalability in cloud environments.

What You'll Do8

  • 1Install, configure, and maintain FreeSWITCH in production environments from scratch.
  • 2Design and implement call flows, dialplans, IVRs, conferencing, and call routing with complete ownership.
  • 3Configure SIP profiles, gateways, trunks, extensions, NAT traversal, firewall rules, and RTP port ranges.
  • 4Troubleshoot SIP/RTP issues including one-way audio, no-audio, call drops, and registration failures using siptrace, sngrep, and tcpdump.
  • 5Develop dialplans and call logic using XML, Lua, JavaScript, or Python, including call queues, time-based routing, skill-based routing, failover, and fallback handling.
  • 6Integrate FreeSWITCH with CRM/ERP systems, billing platforms, AI voice bots (ASR/TTS), and external WebSocket/REST services using Event Socket Library (ESL).
  • 7Configure WebRTC support (DTLS-SRTP, ICE, OPUS) with mod_verto and SIP over WSS for browser-based calling.
  • 8Deploy on Linux (CentOS/Ubuntu) and cloud platforms (AWS/GCP/Azure), configure high availability, load balancing, scaling, and work with DevOps teams for CI/CD, backups, and automation.

Requirements7

  • 14-10 years of experience in VoIP/telephony engineering with strong hands-on FreeSWITCH experience.
  • 2Solid knowledge of VoIP protocolsSIP, RTP, RTCP, and WebRTC — including SDP negotiation and media handling.
  • 3Strong Linux system administration skills and experience managing services on CentOS/Ubuntu.
  • 4Proficiency in dialplan development using XML, Lua, JavaScript, or Python.
  • 5Experience with Event Socket Library (ESL) for inbound/outbound call control and real-time monitoring.
  • 6Familiarity with debugging tools such as fs_cli, sngrep, and tcpdump for SIP/RTP troubleshooting.
  • 7Knowledge of SQL/MySQL/PostgreSQL for CDR and reporting, plus experience with HA, load balancing, and cloud deployment (AWS/GCP/Azure).

Who Should Apply

The ideal candidate is a hands-on VoIP/Telephony engineer who enjoys owning FreeSWITCH deployments end-to-end, from initial installation to production troubleshooting. You should be comfortable working in live environments, debugging complex SIP/RTP media issues, and collaborating with DevOps and integration teams on high-availability telephony systems.

Salary Insight

Open to Discussion

Location

TypeOn-site
LocationGurugram
Eligible countriesIndia

Required Skills

Dialplan (XML/Lua/JavaScript/Python)Event Socket Library (ESL)FreeSWITCHLinuxMySQL/PostgreSQLRTP/RTCPSIPWebRTCsngreptcpdump

Application Tip

Highlight specific FreeSWITCH production projects you have owned, especially those involving high-availability architectures, dialplan scripting, and resolution of issues like one-way audio or call drops. Mention the exact tools and languages you used — fs_cli, sngrep, XML, Lua, Python — so your hands-on experience is immediately visible.

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