Job Details
- Required Skills
- DockerJenkinsKubernetesWebRTCLinuxAnsible
Requirements
- Degree in Computer Science, Information Technology, Telecommunications or similar
- Strong understanding of IP telephony (VoIP)
- Strong understanding of 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
- Experience with 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
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
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