1. The Sub-300ms Perceptual Latency Barrier
In human speech conversation, turn-taking delays above 300ms feel unnatural and awkward. Traditional sequential voice stacks (STT -> REST API LLM -> TTS) accumulate over 1,200ms of roundtrip latency.
To achieve true conversational fluidness, InexpensiveCoders builds Full-Duplex WebRTC Streaming Engines that process audio chunks concurrently across Faster-Whisper STT, vLLM Speculative Tokens, and Streaming XTTS v2 speech synthesis.
Sub-300ms Voice Directives
- WebRTC PCM Audio Streaming: Eliminates HTTP POST overhead with bi-directional WebSockets
- Chunked Audio VAD: Dynamic Silero Voice Activity Detection cuts silence processing overhead
- Barge-In Interrupt Protocol: Instantly cancels ongoing TTS generation when user speaks mid-sentence
2. Real-Time Full-Duplex Latency Breakdown
| Voice Stack Component | Traditional REST Stack | InexpensiveCoders WebRTC Pipeline | |
|---|---|---|---|
| Speech-to-Text (STT) | 350 ms (Whisper API) | 85 ms (Faster-Whisper CTranslate2) | |
| LLM First Token (TTFT) | 450 ms (GPT-4) | 65 ms (vLLM Llama-3 8B) | |
| Text-to-Speech (TTS) | 500 ms (ElevenLabs API) | 110 ms (Chunked Streaming XTTS) | |
| Total Roundtrip Latency | 1 | 300 ms | 260 ms (Sub-300ms Human SLA) |
