Walk-in Interview – VoIP Engineer

startelelogic; Dadri; On-site; INR 350,000 - 600,000; Full time

Job summary

startelelogic invites a VoIP Engineer with 1+ year of hands-on experience to join the telecom engineering team in Uttar Pradesh. You will develop, configure, troubleshoot, and maintain carrier‑grade VoIP systems using SIP, FreeSWITCH, and Kamailio on Linux.

The role requires troubleshooting SIP call flows, analyzing SIP messages, and working with trunks, gateways, and routing. On-site walk-in interview in Ithaira, UP.

Qualifications

• 1+ year hands‑on experience in VoIP/SIP engineering.
• Basic to intermediate understanding of VoIP and SIP.
• Hands‑on experience with FreeSWITCH and/or Kamailio.
• Good understanding of Linux.
• Ability to read and troubleshoot SIP traces and call flows.

Responsibilities

• Configure, troubleshoot, and maintain FreeSWITCH and Kamailio platforms.
• Understand and troubleshoot SIP signaling and VoIP call flows.
• Analyze SIP messages such as INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS, 200 OK, 4xx, and 5xx.
• Troubleshoot common VoIP issues (one-way audio, call failures, NAT issues, etc).
• Work with SIP trunks, extensions, gateways, DIDs, and VoIP routing.
• Monitor and troubleshoot production VoIP services.
• Use tools like sngrep, tcpdump, Wireshark, netstat/ss for debugging.
• Write or modify basic scripts (Python, Lua, Perl, Bash).
• Collaborate with senior engineers to implement new VoIP features.

Skills

VoIP

SIP

FreeSWITCH

Kamailio

Linux

Networking basics

Python/Lua/Bash

C/C++ basics

SIP traces

Troubleshooting

Education

B.Tech/B.E. in CS/EC

Tools

Wireshark

tcpdump

FreeSWITCH

Kamailio

Job description

Walk-in Interview – VoIP EngineerAbout startelelogic

startelelogic is an AI-powered Omnichannel Customer Experience (CX) platform that helps businesses deliver seamless and intelligent interactions across voice, chat, email, WhatsApp, social media, and more.

Our platform brings together Conversational AI, Voice AI, AI Agents, automation, analytics, and human-assisted workflows to simplify customer support, sales, and engagement.

We are looking for a motivated VoIP Engineer with 1+ year of hands‑on experience in VoIP and SIP technologies to join our telecom engineering team.

The ideal candidate should have practical knowledge of SIP, VoIP call flows, Linux, FreeSWITCH, and/or Kamailio, along with basic programming and scripting skills. The candidate will work on developing, configuring, troubleshooting, and maintaining reliable, carrier-grade VoIP systems.

Key Responsibilities

• Configure, troubleshoot, and maintain FreeSWITCH and Kamailio platforms.
• Understand and troubleshoot SIP signaling and VoIP call flows.
• Analyze SIP messages such as INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS, 200 OK, 4xx, and 5xx responses.
• Troubleshoot common VoIP issues, including:
• One-way or no audio
• Call failures
• SIP registration issues
• DTMF problems
• NAT-related issues
• Call routing failures
• Work with SIP trunks, extensions, gateways, DIDs, and VoIP routing.
• Monitor and troubleshoot production VoIP services.
• Use tools such as sngrep, tcpdump, Wireshark, netstat/ss, and other Linux utilities for debugging.
• Work with FreeSWITCH dial plans, XML configurations, and SIP profiles.
• Assist with Kamailio routing and configuration.
• Perform SIP trunk testing, call‑flow analysis, and connectivity troubleshooting.
• Write or modify basic scripts using Python, Lua, Perl, or Bash.
• Collaborate with senior engineers to implement new VoIP features and telecom integrations.
• Monitor system performance and help maintain service availability.
• Document configurations, troubleshooting procedures, and technical changes.

Required Skills

• 1+ year of hands‑on experience in VoIP/SIP engineering.
• Basic to intermediate understanding of VoIP and SIP.
• Hands‑on experience with FreeSWITCH and/or Kamailio.
• Good understanding of Linux.
• Ability to read and troubleshoot SIP traces and call flows.
• Understanding of TCP/IP, UDP, DNS, NAT, and basic networking concepts.
• Basic knowledge of RTP and media flow.
• Basic knowledge of Python, Lua, Perl, Bash, or Golang.
• Basic knowledge of C/C++ is an advantage.
• Strong troubleshooting, analytical, and problem‑solving skills.
• Ability to work effectively in a production telecom environment.

Good to Have

• Experience with OpenSIPS or other SIP proxy technologies.
• Knowledge of RTP, SRTP, and media handling.
• Familiarity with SIP over UDP, TCP, and TLS.
• Knowledge of codecs such as G.711, G.729, Opus, and GSM.
• Exposure to REST APIs and telecom integrations.
• Basic understanding of databases such as MySQL, PostgreSQL, or Redis.
• Experience with Git and Linux-based development environments.
• Exposure to Docker, Kubernetes, or cloud-based VoIP deployments is a plus.

Education

• B.Tech/B.E. in Computer Science, Electronics, Electronics & Communication, or Telecommunications is required.
• BCA, MCA, diploma courses, or other non-engineering qualifications will not be considered.
• Candidates should have a strong academic foundation in computer networks, telecommunications, Linux, and VoIP/SIP technologies.
• Fresh B.Tech/B.E. graduates with strong VoIP/SIP fundamentals and relevant practical projects may also be considered.

Date: September 5, 2026,Saturday

Time: 11:00 AM – 3:00 PM

Experience: 1+ Year

Open Positions: 5

Employment Type: Full-Time

Work Mode: On-site

First Floor, NX ONE, T1, 117 & 118

Ithaira, Uttar Pradesh – 201318

What You Will Work On

As a VoIP Engineer at startelelogic, you will work on real‑world telecom and carrier-grade communication systems, including VoIP infrastructure, SIP routing, FreeSWITCH, Kamailio, SIP trunks, carrier connectivity, call routing, troubleshooting, and AI-powered customer experience platforms.

This role provides an opportunity to gain hands‑on experience across VoIP, CCaaS, CPaaS, AI-powered communications, and cloud communication technologies while working with a growing telecom technology company.

• Please bring an updated copy of your resume/CV and be prepared for a technical discussion/interview focused on VoIP, SIP, Linux, FreeSWITCH, Kamailio, and networking fundamentals.