About the Role
Were looking for a Senior Real-Time Communications Engineer to design, build, and scale our audio/video infrastructure on LiveKit. You’ll work deep in the WebRTC stack using Go and Pion, building low-latency media services that support thousands of concurrent users. This role suits someone who enjoys working close to the protocol layer and owns systems from architecture through production.
Key Responsibilities
- Design, develop, and maintain real-time audio/video applications and services built on LiveKit (self-hosted and/or LiveKit Cloud).
- Extend and customize LiveKit server components, including SFU behavior, room management, egress/ingress, and webhooks.
- Build Go services with Pion WebRTC for custom media pipelines, bots, recording, transcoding, or media relay.
- Implement and debug signaling, ICE/STUN/TURN connectivity, NAT traversal, and peer connection lifecycle management.
- Optimize media quality and latency through simulcast, SVC, bandwidth estimation, congestion control, and adaptive bitrate strategies.
- Integrate real-time AI/voice agents (for example, the LiveKit Agents framework, STT/TTS, and LLM pipelines) where applicable.
- Deploy and scale WebRTC infrastructure on Kubernetes and cloud platforms (AWS/GCP/Azure), including multi-region setups.
- Monitor production systems, analyze WebRTC stats, and troubleshoot issues like packet loss, jitter, and connection failures.
- Collaborate with frontend and mobile teams on client SDK integration (JavaScript, React, iOS, Android, Flutter).
- Write clean, tested, well-documented code and contribute to architecture decisions and code reviews.
Required Qualifications
- 5+ years of professional software engineering experience, with at least 3 years in real-time communication or WebRTC.
- Strong proficiency in
Go (Golang)
, including concurrency patterns (goroutines, channels), profiling, and performance tuning. - Hands-on experience with
Pion WebRTC
, such as building peer connections, handling tracks, and working with RTP/RTCP packets. - Production experience with
LiveKit
, including server deployment, configuration, and SDK usage. - Solid understanding of WebRTC internals: SDP, ICE, DTLS, SRTP, RTP/RTCP, and codecs (Opus, VP8, VP9, H.264, AV1).
- Experience with SFU architectures and media routing concepts.
- Experience deploying and operating STUN/TURN servers (e.g., coturn or Pion TURN).
- Familiarity with Redis, Docker, Kubernetes, and CI/CD pipelines.
- Strong debugging skills using tools like chrome://webrtc-internals, Wireshark, and Prometheus/Grafana.
- Good communication skills and the ability to work independently in a distributed team.
Preferred Qualifications
- Contributions to LiveKit, Pion, or other open-source WebRTC projects.
- Experience with GStreamer or FFmpeg for media processing, recording, or streaming.
- Knowledge of streaming protocols such as RTMP, WHIP/WHEP, HLS, or SRT.
- Experience building voice AI agents or integrating real-time speech pipelines.
- Familiarity with other SFUs/media servers (mediasoup, Janus, Jitsi, Kurento).
- Experience with frontend WebRTC development in JavaScript/TypeScript.
- Background in networking fundamentals (UDP/TCP, QoS, latency optimization).
