About the Role
Role details below.
Responsibilities
- 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
- 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.
Nice to Have
- Familiarity with programming in Python, Elixir or Go.
Additional Information
- The role involves working with users around the globe to solve communications challenges.
- Collaboration with software developers, business teams, sales, and operations is expected.
- The candidate is expected to be comfortable learning new technologies and systems.
- The candidate must be able to triage escalated issues and work with all involved parties until resolution.
- The candidate is expected to be the subject matter expert for telephony-related topics.
