Sub-800ms Voice Telephony: The Engineering Behind Real-Time Voice AI

Conversational latency is the single biggest factor separating frustrating voice bots from indistinguishable AI receptionists. Human conversation operates on natural turn-taking gaps of roughly 200ms to 400ms. If your voice system introduces 1.5s of latency, callers talk over the assistant or hang up.

The Voice AI Latency Budget

To stay under 800ms end-to-end, every layer of the architecture must be tuned for streaming:

1. VAD (Voice Activity Detection): 80ms - 120ms to detect turn completion without false positives.
2. Streaming STT (Speech-to-Text): Partial transcriptions pushed via bidirectional WebSockets.
3. LLM Time-to-First-Token (TTFT): Sub-250ms using optimized speculative decoding and edge deployments.
4. Streaming TTS (Text-to-Speech): Synthesizing audio chunk-by-chunk to start playback immediately on first phoneme.

SIP & WebRTC Edge Architecture

By terminating SIP calls directly on low-latency edge gateways, we bypass standard telecom transcode hops. Audio frames stream straight into memory buffers, eliminating roundtrip penalty.