Role description
Job Title: C++ VOIP Engineering Manager
Location: Ahmedabad (WFO) | Experience: 9+ Years
Who We Are at STL Digital, we don't just build software; we engineer digital
transformations. We partner with global enterprises with a comprehensive portfolio of
services, including Product Engineering, Cloud and Cybersecurity, Data and AI, and
Enterprise SaaS.
We're hiring a hands-on Engineering Manager to lead and actively build the core of our voice communications stack. You will be a player-coach: owning critical VoIP/SIP/C++ development while guiding developers across multiple components and languages (PHP, JavaScript/Node.js, Java/Spring Boot, Struts, Perl, databases, and Linux infrastructure).
Key Responsibilities
This role is ideal for someone who can confidently dive into SIP traces and C++ code, then unblock UI/backend teams and drive delivery.
Hands-on VoIP Core Engineering
- Design and implement features in C++ for call control/core PBX capabilities (multi-tenant behaviour, routing, services, etc.)
- Own and improve SIP signalling behaviour across proxy/B2BUA/core integration (registrations, routing, NAT handling, failover patterns)
- Work deeply with SIP/SDP/RTP flows and real-world interoperability issues (carriers, ITSPs, IP phones, softphones)
- Integrate and optimize the media layer (Asterisk): call flows, recordings, announcements, transcoding considerations, capacity planning
- Lead root-cause analysis of production issues (e.g., one-way audio, jitter, registration drops, codec mismatches, NAT/firewall problems) using tooling like SIP traces and packet captures
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 behaviour, 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.
