Building a Real-Time Voice Agent with LiveKit Agents SDK
How we architected a sub-300ms real-time voice agent for enterprise deployment — and why those latency decisions directly impact user trust and adoption.
Real-time voice agents have a hard requirement: latency below 300ms or the conversation feels broken. Getting there requires careful architecture, not just picking the right model.
The Stack That Works
We use LiveKit Agents SDK as the real-time transport layer, paired with OpenAI's Realtime API for turn-taking and function calling. Supabase handles session state and any data lookups the agent needs mid-call.
Key Architectural Decisions
1. Keep tool calls tight: Every function call the agent makes during a live call adds latency. We pre-load context into the session prompt and only call tools when the user explicitly requests data.
2. Interrupt handling matters: Users interrupt themselves and each other. LiveKit handles the WebRTC layer, but you need to explicitly design your agent to handle mid-sentence interruptions gracefully — most docs gloss over this.
3. Warm up the pipeline: Cold-starting a voice session has a baseline cost. We pre-warm agent instances during business hours for clients with predictable call volumes.
4. Fallback to text: Not every environment can guarantee audio quality. We build every voice agent with a text fallback that shares the same tool and context layer.
What We Learned
The hardest part of voice agents isn't the AI — it's the plumbing. Audio encoding, silence detection thresholds, and session cleanup are where most production issues come from. LiveKit handles most of this correctly by default; don't fight the defaults until you have a reason to.
Why This Matters for Decision Makers
- Improves adoption rates by eliminating perceived latency friction
- Reduces session instability risk in production environments
- Prevents latency-driven user drop-off before value is delivered
