Telephony Engineer (FreeSWITCH, VoIP) (Gurugram)

Telephony Engineer (FreeSWITCH, VoIP) (Gurugram)

08 Aug
|
TalentXO
|
Gurugram

08 Aug

TalentXO

Gurugram

Hiring for a Client - Eureka Digitisation & Automation Services (eDAS)

About the Role

We are looking for a FreeSWITCH Implementation Engineer with robust hands-on experience in designing, deploying, configuring, integrating, and supporting FreeSWITCH-based VoIP platforms in production environments. The ideal candidate should have deep expertise in SIP, RTP, WebRTC, IVR, dialplans, call routing, and third-party integrations.

Key Responsibilities

FreeSWITCH & Call Management

- Install, configure, and maintain FreeSWITCH in production environments.
- Design and implement call flows, dialplans, IVRs, conferencing, and call routing end-to-end.
- Configure SIP profiles, gateways, trunks, and extensions.
- Implement multi-tenant and high-availability FreeSWITCH architectures.

SIP, RTP & Networking

- Troubleshoot SIP signaling, RTP media, SDP negotiation, call drops, one-way/no audio, and registration failures.
- Configure NAT traversal, firewall rules, and RTP port ranges.
- Analyze SIP/RTP logs and traces using siptrace, sngrep, and tcpdump.

Dialplans & Scripting

- Develop FreeSWITCH dialplans using XML and scripting languages such as Lua, JavaScript, or Python.
- Implement call queues, time-based routing, skill-based routing, failover, and fallback logic.

Integrations & APIs

- Integrate FreeSWITCH with CRM/ERP systems, billing platforms, AI voice bots (ASR/TTS), REST APIs, and WebSockets.
- Work with Event Socket Library (ESL) for real-time call control and monitoring.

WebRTC & Media

- Configure WebRTC, DTLS-SRTP, ICE, and OPUS.
- Work with mod_verto and SIP over WSS.
- Implement browser-based calling solutions.

Monitoring & Troubleshooting

- Monitor system performance, call quality, logs, and CDRs.
- Optimize codec usage and RTP performance.
- Handle live production incidents, troubleshooting, and Root Cause Analysis (RCA).

Deployment & Operations





- Deploy and manage FreeSWITCH on Linux (Ubuntu/CentOS).
- Support cloud deployments on AWS, GCP, or Azure.
- Configure HA, load balancing, scaling, backups, and automation.
- Collaborate with DevOps teams on CI/CD and deployment workflows.

Mandatory Requirements

- 4+ years of overall experience, including at least 3+ years of hands-on FreeSWITCH implementation, configuration, and production support.
- Strong experience designing and implementing FreeSWITCH call flows, dialplans, IVRs, call routing, conferencing, SIP profiles, gateways, trunks, and extensions.
- Strong hands-on knowledge of SIP, RTP, RTCP, SDP, and WebRTC.
- Proven ability to troubleshoot call drops, one-way/no audio, registration failures, NAT traversal, and SIP/RTP issues.
- Hands-on experience developing FreeSWITCH dialplans using XML and at least one of Lua, JavaScript, or Python.
- Experience implementing call queues, time-based routing, skill-based routing, and failover logic.
- Experience integrating FreeSWITCH with CRM/ERP, REST APIs, WebSockets, AI Voice Bots, or Billing Platforms.
- Hands-on experience with Event Socket Library (ESL) for real-time call control.
- Strong Linux administration skills with experience managing FreeSWITCH on Ubuntu/CentOS.
- Comfortable with fs_cli, sngrep, tcpdump, and SQL databases such as MySQL/PostgreSQL for CDRs and reporting.
- Immediate Joiners or candidates who can join within 15 days will be highly preferred.
- Must be currently based in Delhi NCR.

Preferred

- Experience with AWS/Azure/GCP, HA, load balancing, CI/CD, automation, backups, and scalable FreeSWITCH deployments.
- Experience implementing WebRTC-based calling solutions, including mod_verto, SIP over WSS, DTLS-SRTP, ICE, and OPUS.

Ideal Candidate: A hands-on FreeSWITCH specialist with strong production implementation experience, deep VoIP troubleshooting skills, and expertise in SIP/RTP, dialplans, integrations, Linux, and WebRTC.

📌 Telephony Engineer (FreeSWITCH, VoIP) (Gurugram)
🏢 TalentXO
📍 Gurugram

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