Job Title

We are seeking a Lead VoIP Engineer to design and build high-performance modules within our Voice platform. You’ll work on the core telephony stack involving signaling, media processing, NAT traversal, and RTP relaying. This is a hands-on execution role ideal for engineers who love building, debugging, and optimizing real-time communication systems.What You’ll Do:• Implement core voice capabilities using FreeSWITCH, Kamailio/OpenSIPs, and RTPEngine• Build and optimize SIP call routing logic, RTP media relays, failover mechanisms, and NAT traversal • Develop and manage configurations for scalability, codec negotiation, SIP trunk registration • Implement and test features like call recording, IVR, voicemail, DTMF detection • Monitor live traffic and participate in 24x7 on-call rotation for critical escalations• Collaborate with QA on stress/load testing and with Backend teams on provisioning APIs • Document design decisions, configurations, and troubleshooting runbooks What Makes You Qualified:• 5+ years of experience building and operating VoIP systems or CPaaS platforms • Solid expertise with SIP signaling, RTP, and media relay techniques • Strong hands-on with FreeSWITCH, Kamailio/OpenSIPs, RTPEngine • Hands-on experience with Session Border Controller (SBC), Media Servers and WebRTC. • Experience in managing telephony infrastructure for uptime, latency, and call quality optimization • Strong systems programming and debugging skills in C/C++ • Good scripting/debugging skills (Bash, Python, or Lua for FreeSWITCH modules) • Proficiency with diagnostic tools (Wireshark, tcpdump etc) • Experience working with geographically distributed infrastructure or HA deployments