Walk-in Interview - VoIP Engineer

startelelogic · India

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
• Codec negotiation issues
• 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 programming or scripting knowledge in 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.

Walk-in Interview Details

Date: September 5, 2026,   Saturday

Time: 11:00 AM – 3:00 PM

Position: VoIP Engineer

Experience: 1+ Year

Open Positions: 5

Employment Type: Full-Time

Work Mode: On-site

Interview Venue

First Floor, NX ONE, T1, 117 & 118

Greater Noida West, Ghaziabad

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.

⚠️ Important

• 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 .