Engineering Manager - C++ VOIP

STL Digital Inc; India; On-site; INR 4,200,000 - 6,400,000; Full time

Job summary

STL Digital Inc. is seeking an Engineering Manager with deep VoIP expertise in SIP, SDP, and RTP. The role requires strong C++ development and Linux performance analysis to troubleshoot production SIP environments.

You will lead engineers across backend and UI, ensure robust, scalable VoIP components, and communicate complex issues to stakeholders. A strong command of Asterisk/OpenSIPS/Kamailio is preferred, with proven leadership skills.

Qualifications

• 10+ years building production VoIP systems with troubleshooting experience.
• Strong C++ development on Linux with debugging and performance analysis.
• Deep SIP/SDP/RTP/RTCP protocol understanding and VoIP failure handling.
• Solid Linux/networking fundamentals (Ubuntu, sockets, TLS).
• Experience leading engineers across backend and UI through reviews.
• Clear communication to non-VoIP engineers and stakeholders.

Responsibilities

• Lead engineers across a mixed stack (backend + UI) with technical direction.
• Conduct code reviews and set architecture/quality standards.
• Communicate complex VoIP issues clearly to technical and non-technical stakeholders.

Skills

VoIP expertise

C++ development

Linux performance

Code debugging

Leadership

Communication

Education

Bachelor's degree in Computer Science

Tools

Asterisk

OpenSIPS/OpenSIPS

Kamailio

FreeSWITCH

Job description

Description

Job Title: Engineering Manager

Work Mode: WFFO

This role is ideal for someone who can confidently dive into SIP traces and C++ code

Required Qualifications

• 10+ years building production systems in VoIP / SIP environments, with proven troubleshooting experience
• Strong C++ development experience (modern C++ preferred), including debugging and performance analysis in Linux
• Deep protocol understanding: SIP, SDP, RTP/RTCP (and common VoIP failure modes)
• Strong Linux and networking fundamentals (Ubuntu, system tuning, sockets, firewall/NAT behavior, TLS basics)

Experience integrating or operating SIP/media components such as:

• Asterisk (dialplan/call flows, AMI/ARI familiarity is a plus)
• OpenSIPS/OpenSIP or similar SIP proxy/SBC stacks (Kamailio, FreeSWITCH, etc.)

• Demonstrated ability to lead engineers across a mixed stack (backend + UI) through code reviews and technical direction
• Clear communication skills- able to explain complex VoIP issues to non-VoIP engineers and stakeholders