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Measuring response delay, barge-in interruption handling, and turn-taking heuristics during WebSocket audio streaming.
Full-duplex WebSocket channels combined with client-side WASM VAD reduce voice turn-taking latency to <600ms while supporting user barge-in interruptions.
To architect a streaming Voice AI pipeline supporting real-time hands-free conversation with sub-600ms latency, vocal intonation turn-taking, and barge-in interruptions.
Voice Transport (WebRTC / WebSocket)
↓
Audio Processing & WASM VAD (Silero VAD)
↓
Streaming STT (Deepgram Nova-2 / Whisper Live)
↓
Turn-Taking Engine (Dynamic Intonation Analyzer)
↓
Cortex Conversation Manager (Warborn OS)
↓
LLM Router (Ollama Local / Gemini Cloud)
↓
Response Generation Stream
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Streaming TTS (ElevenLabs Turbo v2 / Piper)
↓
Audio Output Transport
| Stage | Target Latency | Experimental Latency |
|---|---|---|
| Voice Transport | <30ms | 22ms |
| VAD & Chunking | <50ms | 45ms |
| Streaming STT | <150ms | 140ms |
| Turn-Taking Engine | <400ms | 400ms |
| LLM Prefill / Generation | <200ms | 185ms |
| Streaming TTS Synthesis | <150ms | 135ms |
| Total Roundtrip Latency | <980ms | 927ms |
Read the full technical specification in Voice AI Architecture (docs/rnd/voice-ai-architecture.md).