Job Description

Experience Required:

3-4 years (VOIP domain)

Location:

Gandhinagar

Shift Timing:

Mornin Shift

Joining:

Immediate

?

About the Role

We are expanding our VoIP engineering team and looking for a developer who understands real-time communication systems and VOIP architecture, including Free Switch, SIP signaling and media handling. You will work on VOIP project deployments, API integrations, troubleshooting, and enhancements.

Roles & Responsibilities

Configure and manage Free Switch, Asterisk, FreePBX, or other PBX platforms. Setup SIP trunks, DID routing, IVR, queues, agents, inbound/outbound dial plans. Develop or modify dial plans based on business needs. Troubleshoot call flow -- one-way audio, SIP errors, codec issues, firewall & NAT. Work with

Wireshark

or

SIP logs

for debugging. Integrate VoIP systems with CRMs or applications using APIs (REST / WebSocket). Deploy and maintain VOIP on cloud infrastructure (AWS / DigitalOcean / Azure). Ensure security best practices (SIP firewall rules, fail2ban, IP whitelisting, TLS).

Required Skills :

VOIP FrameworksAsterisk, FreePBX, FusionPBX OR similarProtocolsSIP, RTP, STUN, TURN, NATLinuxUbuntu / Debian - CLI

Debugging :

using tool (Wireshark / tcpdump / logs)

Advantage if :

Experience with WebRTC, Kamailio, OpenSIPS, Coturn (TURN/STUN) API / backend scripting knowledge (Node.js, Python, or PHP)
Job Types: Full-time, Permanent, Fresher

Pay: Up to ₹550,000.00 per year

Benefits:

Paid sick time Paid time off Provident Fund
Work Location: In person