Looking for a LiveKit / WebRTC performance expert (paid consulting) We run a self-hosted LiveKit production setup handling 1000s of voice calls/day, fully in production. Current state • Self-hosted LiveKit (K8s) • Telephony + AI voice agents • < 2s end-to-end latency (caller → AI → response) • Stable at scale, but seeing intermittent latency spikes & audio interruptions What we want to optimize • Media pipeline latency (ingress → SFU → egress) • Jitter, packet loss handling, buffer tuning • CPU scheduling, GC, and thread contention (Java/Python services involved) • LiveKit SFU config, TURN/STUN, networking, kernel & NIC tuning • Call concurrency stress scenarios What we’re looking for • Someone who has operated LiveKit / WebRTC at scale • Comfortable diving into metrics, pprof, traces, RTP stats • Short-term paid engagement to diagnose + optimize If you’ve done this in production and enjoy deep performance work, please DM or reply with your background.