Voice AI Latency Benchmark: Sub-300ms Pipeline Architecture
An in-depth technical analysis of full-duplex conversational voice systems. Discover how streaming VAD, speculative inference, and chunked TTS achieve natural conversational cadence.
The Complete Full-Duplex Audio Turnaround Breakdown
Every millisecond counts when engineering human-like conversational responsiveness.
| Pipeline Stage | Observed Latency (p50 - p90) | Underlying Technology | Engineering Architecture |
|---|---|---|---|
| 1. Voice Activity Detection (VAD) | 30 – 45 ms | Streaming Silero VAD / Energy Thresholding | Detects the precise millisecond user speech finishes, filtering background acoustic noise without cutting off natural speech mid-sentence. |
| 2. Streaming Speech-to-Text (STT) | 85 – 110 ms | Deepgram Nova-2 / Whisper Streaming Engine | Converts binary PCM audio chunks into raw text tokens in real time, delivering partial transcripts before the sentence fully concludes. |
| 3. LLM Time to First Token (TTFT) | 60 – 85 ms | Groq LPU / Anthropic Claude 3.5 Haiku | Processes RAG context and system instructions to yield the first response token using high-throughput low-latency inference hardware. |
| 4. Streaming Text-to-Speech (TTS) | 65 – 85 ms | Cartesia Sonic / ElevenLabs Flash | Synthesizes the first chunk of natural voice audio immediately upon receiving the first LLM punctuation boundary, rather than waiting for the complete paragraph. |
| 5. Transport & Jitter Buffer | 20 – 35 ms | WebRTC Data Channel & Opus / SIP G.711 | Delivers outbound audio packets directly into the caller's mobile audio buffer with adaptive jitter compensation. |
| TOTAL CONVERSATIONAL TURNAROUND | 260 – 310 ms | End-to-End Orchestrated Pipeline | Under the human perceptual pause threshold (~300ms) |
Related Voice AI & Telephony Architecture
Frequently Asked Questions on Voice AI Latency
Everything you need to know about our AI CRMs, integrations, and deployment timelines.
Our full-duplex audio pipeline continuously listens while speaking. When the VAD detects inbound user speech, the system immediately cuts audio playback within 40ms, drops the active TTS buffer, and updates the conversational context.
Yes. Our edge server network in Mumbai and Delhi ensures round-trip packet transport times under 40ms across Indian mobile networks (Jio, Airtel).
Yes. The voice agent supports neutral Indian English, Hindi, Hinglish, and regional accents with authentic pronunciation.
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