WebRTC Video Engineer at Zoom Communications in San Jose, CA

Job description

What You Can Expect As a WebRTC Video Engineer on Zoom's real-time web media team, you will directly shape the video quality that millions of users depend on for clear, reliable communication. Working across the full media pipeline, from device capture and encoding through network transmission, decoding, and rendering, you will develop code to deliver the best video quality to the customers, diagnose and resolve complex quality issues across browsers, operating systems, and hardware environments. You will collaborate closely with global engineering teams to drive product development and release cycles. This is a hybrid position that requires on-site presence 3 days per week at our San Jose, CA office. Remote arrangements are not available for this role. About the Team The Audio Video team is Zoom's technical foundation for real-time communication, owning the full media pipeline from device capture through processing, encoding, transmission, decoding, and rendering. The team delivers the audio and video experience across Zoom Workplace's native client, web-based client, and Video SDK, enabling developers and enterprise customers to build reliable, high-quality communication tools. By continuously advancing media quality and reliability, the team plays a central role in supporting Zoom's mission to keep people connected and productive. Responsibilities Implement and maintain robust WebRTC media flows across web browsers, native clients, and media servers, including interoperability across these environments. Develop and improve core modules for SDP negotiation, peer connection management, ICE/STUN/TURN, simulcast, packet switch architecture, and adaptive bitrate and congestion control. Investigate and resolve audio/video quality issues, including codec compatibility for H.264, VP8/VP9, and AV1, hardware decoder edge cases, and browser-specific regressions. Optimize end-to-end audio/video workflows across capture, encoding, transport, decoding, and rendering for conferencing use cases. Monitor QoS performance, implement failover metrics, and analyze quality regressions using data-driven methods. Ensure production stability through hotfix deployment, memory leak resolution, and proactive issue identification across production clusters. Deliver US-facing technical support for enterprise SDK customers, covering requirements discussions, ticket investigation, troubleshooting, and integration guidance. Build and maintain team knowledge bases, internal documentation, and best practices for WebRTC systems, and contribute to open-source communities such as Chromium and WebRTC upstream where applicable. What We're Looking For Bachelor's degree or higher in Computer Science, Electrical Engineering, or a related STEM field, or equivalent experience, with 5 or more years of hands-on software engineering experience in real-time media or web communication systems. Solid, production-proven WebRTC development experience, including real-world debugging, protocol-level issue resolution, and cross-browser compatibility work. Proficiency in JavaScript and/or TypeScript for front-end media development, with strong debugging skills including packet-level analysis and use of media and protocol tooling. Working knowledge of real-time media protocols, including ICE, STUN/TURN, DTLS/SRTP, RTP/RTCP, and SDP, with practical experience in SDP parsing, negotiation, and RFC compliance across browser environments. Familiarity with video codec internals, particularly H.264 (including SPS/PPS, key frames, and hardware decoder compatibility), and awareness of VP8/VP9 and AV1. Experience with simulcast, packet switch flows, or adaptive bitrate mechanisms in a media SDK or media server context. Demonstrated ability to investigate and resolve production issues, including hotfixes, memory leaks, and browser-specific regressions, with experience supporting enterprise SDK customers technically as a plus. Salary Range or On Target Earnings: $127,700.00 - $255,400.00 In a