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7 min readJanuary 28, 2026

AI Voice Agents: Achieving Sub-500ms End-to-End Latency with WebRTC

How to build conversational voice bots that feel human using streaming speech-to-text, fast LLM token generation, and neural audio synthesis.

A
Avernus Engineering Team
AI Systems Architecture
[ BLOG COVER: AI Voice Agents ]

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Key Takeaways
  • Human conversational comfort threshold is below 600ms; anything higher feels unnatural and disjointed.
  • Replacing HTTP request-response loops with WebRTC audio streams eliminates transport overhead.
  • Sentence-boundary audio streaming enables TTS synthesis to begin while the LLM is still generating subsequent words.

The Latency Budget of Voice Interaction

Building a realistic voice agent requires breaking down the end-to-end latency waterfall into three strict budgets: Speech-to-Text (STT, max 150ms), LLM First Token Generation (max 200ms), and Text-to-Speech (TTS First Chunk, max 150ms).

StageLegacy ApproachModern Ultra-Low Latency Pipeline
Audio TransportREST POST .wav file (1200ms)WebRTC bidirectional stream (<40ms)
TranscriptionBatch Whisper API (800ms)Streaming Deepgram / Whisper Live (<150ms)
LLM GenerationWait for complete sentence (600ms)Streaming tokens with punctuation chunking (<150ms)
Voice SynthesisSynthesize entire paragraph (900ms)Cartesia / ElevenLabs Turbo stream (<120ms)

Handling User Interruptions Gracefully

A critical test of voice bots is handling user speech overlap. When voice activity detection (VAD) senses human speech, the client must immediately cancel the active audio buffer and send an abort signal to the LLM generation thread.

Topics Covered:
#Voice AI#WebRTC#TTS#STT#Real-Time

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