Understand how all Telephony services work and how they are integrated to the platform.
Operate, maintain, expand and support SIP proxies based on Kamailio or OpenSIPS applications.
Handle level 3 troubleshooting escalations and triaging.
Analyze telephony traffic patterns and identify issues and anomalies.
React to critical alerts in order to rapidly return to a full-service state.
Troubleshoot and resolve voice and network protocol communication issues.
Interface with partner organizations for interconnections and expansions.
Design, build, test, deploy and maintain monitoring, alerting, QA and logging tools for Telephony applications.
Coordinate system maintenance and deployment events.
Requirements
A degree in Computer Science, Information Technology, Telecommunications or similar.
Strong understanding of IP telephony (VoIP), TCP/IP Networks and related protocols (SIP, RTP, RTCP, ISUP, TLS, STUN, TURN, WebRTC).
Experience with Open Source VoIP applications such as Kamailio, OpenSIPS, FreeSWITCH, RTPEngine, Asterisk and open source tools such as Wireshark, sngrep and Homer.
Experience with Linux, Open Source tools and shell scripting.
Experience with containers and automation/orchestration tools such as Docker, Ansible, Jenkins, Kubernetes.
Familiarity with programming in Python, Elixir or Go are a plus.