WebRTC Engineer (LiveKit)

About the Role

We’re looking for a WebRTC Engineer to design, build, and scale real-time audio and video experiences using LiveKit. You’ll own our real-time communication stack end to end, from media servers and signaling to client SDK integration, with a focus on low latency, reliability, and call quality at scale.

Key Responsibilities

Design, build, and maintain real-time audio/video features using LiveKit (server and client SDKs).

Deploy, configure, and scale self-hosted LiveKit infrastructure or LiveKit Cloud, including SFU configuration, TURN/STUN servers, and multi-region setups.

Integrate LiveKit into web and mobile clients (JavaScript/TypeScript, React, iOS, Android, Flutter, or React Native).

Build backend services for room management, access tokens, webhooks, recording (Egress), and streaming ingest (Ingress).

Monitor and improve media quality: latency, jitter, packet loss, bandwidth adaptation, and simulcast/SVC behavior.

Troubleshoot connectivity issues across NATs, firewalls, and restrictive networks.

Build or integrate real-time AI features such as voice agents, transcription, or media processing using LiveKit Agents (if applicable).

Collaborate with product, frontend, and DevOps teams to ship reliable real-time features.

Required Qualifications

7+ years of software engineering experience, with at least 5 years working directly with WebRTC.

Hands-on production experience with LiveKit (rooms, tracks, participants, tokens, webhooks, Egress/Ingress).

Strong understanding of WebRTC fundamentals: ICE, STUN/TURN, SDP, DTLS/SRTP, RTP/RTCP, and codecs (Opus, VP8, VP9, H.264, AV1).

Experience with SFU architectures and concepts like simulcast, SVC, and adaptive bitrate.

Proficiency in Go, Node.js/TypeScript, or Python for backend services.

Solid experience with JavaScript/TypeScript and browser media APIs (getUserMedia, MediaStream, RTCPeerConnection).

Experience deploying and operating services with Docker, Kubernetes, and a major cloud provider (AWS, GCP, or Azure).

Ability to debug media issues using tools like chrome://webrtc-internals, Wireshark, or getStats() data.

Preferred Qualifications

Experience with LiveKit Agents or building real-time voice AI applications (STT, TTS, LLM integration).

Experience with other media servers such as Janus, mediasoup, Jitsi, or Pion.

Mobile real-time experience (native iOS/Android or cross-platform frameworks).

Knowledge of Redis, which LiveKit uses for multi-node deployments.

Familiarity with observability tools (Prometheus, Grafana, Datadog) for media metrics.

Contributions to open-source WebRTC or LiveKit projects.