WebRTC Video Engineer
About The Position
As a WebRTC Video Engineer on Zoom's real-time web media team, you will shape video quality for users by developing code across the full media pipeline, from device capture and encoding through network transmission, decoding, and rendering. You will diagnose and resolve complex quality issues across browsers, operating systems, and hardware environments, and collaborate with global engineering teams on product development and release cycles. This is a hybrid role requiring on-site presence three days per week at Zoom's San Jose, California office; remote arrangements are not available. The Audio Video team owns Zoom's real-time communication media pipeline, from device capture through processing, encoding, transmission, decoding, and rendering. It delivers audio and video experiences across Zoom Workplace's native client, web-based client, and Video SDK, supporting developers and enterprise customers in building reliable, high-quality communication tools. The team advances media quality and reliability in support of Zoom's mission to keep people connected and productive.
Requirements
- Bachelor's degree or higher in Computer Science, Electrical Engineering, or a related STEM field, or equivalent experience.
- At least 5 years of hands-on software engineering experience in real-time media or web communication systems.
- 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 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.
- Ability to investigate and resolve production issues, including hotfixes, memory leaks, and browser-specific regressions.
Nice To Haves
- Experience supporting enterprise SDK customers technically.
Responsibilities
- Implement and maintain robust WebRTC media flows across web browsers, native clients, and media servers, including interoperability across these environments.
- Develop and improve 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, 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, including 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.
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What This Job Offers
Job Type
Full-time
Career Level
Senior
