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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
Job ID: 142257847