WebRTC Engineer

As a WebRTC Engineer you understand how the Telnyx WebRTC platform works as a whole while also being able to identify each underlying service and its purpose proxies back-to-back-user agents SDK clients push notification services etc. You use this knowledge plus your experience working with WebRTC technologies to design new features and guide software developers on their implementation. You assist other teams by triaging escalated issues and then working with all the involved parties until they are resolved. You are the subject matter expert for all topics that are WebRTC related and when you don't know the answer you will know who to ask.

You work with users around the globe where you help by solving their communications challenges. You’ll also get to work with other engineers to build delightful features that span various parts of the system as well as our business sales and operations teams to understand and solve our users’ pain points.

We’re looking for people with a strong background in WebRTC technologies and interest in building successful products or systems; you’re comfortable in dealing with lots of moving pieces; you have exquisite attention to detail; and you’re comfortable learning new technologies and systems.

In This Role You Will

• Understand how all WebRTC services work and how they are integrated to the platform.

• Handle level 3 troubleshooting escalations and triaging.

• Analyze 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 WebRTC applications.

• Coordinate system maintenance and deployment events.

You May Be A Fit For This Role If You Have

• A degree in Computer Science Information Technology Telecommunications or similar.

• Experience with open source WebRTC technologies such as Janus Verto Pion SIP.js.

• 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 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 C/C++ or Go are a plus.

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