Responsibilities
- Implement core voice capabilities using FreeSWITCH, Kamailio, 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.
- Troubleshoot and debug production issues, providing timely resolutions.
- Stay up-to-date with emerging technologies and industry trends, identifying opportunities for innovation and improvement.
- Collaborate with DevOps teams to ensure smooth deployment and operation of backend services in the AWS cloud environment.
Requirements
- 8+ 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 (Homer , VOIPMonitoring, Wireshark, tcpdump etc).
- Experience working with geographically distributed infrastructure or HA deployments.
