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
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