Voice AI Architecture Guide: Cascaded vs Speech-to-Speech in 2026
Building a production voice AI system in 2026 requires architectural decisions that fundamentally impact latency, quality, and control. The choice between cascaded architectures (STT → LLM → TTS) and speech-to-speech models (GPT-4o Realtime, Gemini 2.5 Flash) determines everything from debugging complexity to user experience.
This technical guide breaks down the three core voice AI architectures, explains Voice Activity Detection (VAD) for natural turn-taking, covers tool calling in voice contexts, and provides a latency optimization framework for hitting that critical 300-500ms response window humans expect.
🎯 The Critical Architecture Decision
Human conversation requires 300-500ms response time to feel natural. Delays beyond 500 milliseconds feel unnatural. Your architecture determines whether you hit that target.
💡 The 2026 reality: Speech-to-speech models like GPT-4o Realtime and Gemini 2.5 Flash have closed the quality gap while cutting latency in half. But cascaded architectures still win for enterprise scenarios requiring specific compliance, custom models, or fine-grained debugging.
🔗 Architecture 1: Cascaded (Traditional)
How It Works
User speaks → Audio captured → STT model transcribes to text
Latency: 100-500ms (streaming vs batch)
Text processed → LLM generates response text
Latency: 200-2000ms (depends on model size, prompt complexity)
Response text → TTS synthesizes audio → User hears response
Latency: 200-800ms (varies with streaming, quality)
✅ Advantages
- Best-in-class components: Mix Deepgram STT + GPT-4 + ElevenLabs TTS
- Maximum control: Swap any component independently
- Easier debugging: Inspect text at each stage
- Compliance-friendly: Use specific certified models for regulated industries
- Custom model support: Fine-tuned models for domain-specific tasks
⚠️ Disadvantages
- Higher total latency: 2-4s end-to-end in good conditions
- Integration complexity: Coordinate 3+ separate services
- Lost audio nuance: Tone, emotion, laughter disappear in transcription
- More failure points: Each component can fail independently
- Network overhead: Multiple round-trips add 50-200ms each
Example: Cascaded Voice Agent with Deepgram + GPT-4 + ElevenLabs
import Deepgram from "@deepgram/sdk";
import OpenAI from "openai";
import { ElevenLabsClient } from "elevenlabs";
// Initialize clients
const deepgram = new Deepgram(process.env.DEEPGRAM_API_KEY);
const openai = new OpenAI({ apiKey: process.env.OPENAI_API_KEY });
const elevenlabs = new ElevenLabsClient({ apiKey: process.env.ELEVENLABS_API_KEY });
async function processVoiceInput(audioBuffer) {
// Step 1: STT - Transcribe audio (100-500ms)
const transcription = await deepgram.transcription.preRecorded({
buffer: audioBuffer,
mimetype: "audio/wav"
}, { model: "nova-3", language: "en" });
const userText = transcription.results.channels[0].alternatives[0].transcript;
console.log("User said:", userText);
// Step 2: LLM - Generate response (200-2000ms)
const completion = await openai.chat.completions.create({
model: "gpt-4",
messages: [
{ role: "system", content: "You are a helpful voice assistant." },
{ role: "user", content: userText }
],
max_tokens: 150
});
const responseText = completion.choices[0].message.content;
console.log("Assistant responds:", responseText);
// Step 3: TTS - Synthesize speech (200-800ms)
const audio = await elevenlabs.generate({
voice: "Rachel",
text: responseText,
model_id: "eleven_multilingual_v2"
});
return audio; // Total: 500-3300ms depending on streaming optimizations
} ⏱️ Latency breakdown: Best case with streaming: ~500-800ms total (STT 100ms + LLM 200ms + TTS 200ms + network 100ms). Typical production: 2-4 seconds without aggressive optimization.
🚀 Architecture 2: Speech-to-Speech (End-to-End)
How It Works
Models like GPT-4o Realtime, Gemini 2.5 Flash Native Audio, and Grok Voice API process audio directly without intermediate text transcription.
User speaks → Native audio model processes → Audio response generated
Latency: Sub-1 second end-to-end (Grok: <1s, GPT-4o Realtime varies)
The model is trained to analyze the raw acoustic signal, bypassing the need for intermediary text transcription (ASR) and text synthesis (TTS). This preserves tone, emotion, laughter, and non-verbal cues.
✅ Advantages
- Lowest latency: Sub-1s response time in production
- Preserves emotion: Tone, laughter, non-verbal cues maintained
- More natural: Responds to how user speaks, not just what they say
- Simpler integration: Single API call, fewer moving parts
- Better interruptions: Native handling of barge-in scenarios
⚠️ Disadvantages
- Less control: Can't swap STT, LLM, TTS independently
- Harder debugging: No intermediate text to inspect
- Vendor lock-in: Tied to provider's full stack
- Limited customization: Can't use fine-tuned domain models
- Compliance challenges: May not meet specific regulatory requirements
OpenAI GPT-4o Realtime API
The GPT-4o Realtime API lets you create a persistent WebSocket connection to exchange messages with GPT-4o, sending audio input and receiving audio responses in real-time. Available via WebRTC or WebSocket.
The new gpt-realtime model shows higher intelligence, can comprehend native audio with greater accuracy, captures non-verbal cues (like laughs), and switches languages mid-sentence. Pricing reduced by 20%: $32/1M audio input tokens and $64/1M audio output tokens.
Gemini 2.5 Flash Native Audio
Google's Gemini 2.5 Flash Native Audio is trained to analyze the raw acoustic signal itself, bypassing intermediary text transcription and synthesis. The recent update improves the model's ability to handle complex workflows, navigate user instructions, and hold natural conversations.
Emotional Intelligence: Models using Gemini Live API native audio can understand and respond appropriately to users' emotional expressions for more nuanced conversations. Better conversation quality with improved multi-turn context retrieval.
Example: GPT-4o Realtime API (WebSocket)
import WebSocket from "ws";
const ws = new WebSocket("wss://api.openai.com/v1/realtime", {
headers: {
"Authorization": `Bearer ${process.env.OPENAI_API_KEY}`,
"OpenAI-Beta": "realtime=v1"
}
});
ws.on("open", () => {
// Configure session for voice conversation
ws.send(JSON.stringify({
type: "session.update",
session: {
modalities: ["text", "audio"],
instructions: "You are a helpful voice assistant.",
voice: "alloy",
input_audio_format: "pcm16",
output_audio_format: "pcm16",
turn_detection: { type: "server_vad" } // Built-in VAD
}
}));
});
ws.on("message", (data) => {
const event = JSON.parse(data);
if (event.type === "response.audio.delta") {
// Stream audio chunks directly to speaker (sub-1s latency)
playAudioChunk(event.delta);
}
if (event.type === "response.audio.done") {
console.log("Response complete");
}
});
// Send audio input
function sendUserAudio(audioBuffer) {
ws.send(JSON.stringify({
type: "input_audio_buffer.append",
audio: audioBuffer.toString("base64")
}));
} 💡 When to choose Speech-to-Speech: Consumer-facing apps where latency is critical, emotional conversations (therapy, customer service), scenarios requiring natural interruptions, applications where "how" matters as much as "what" is said.
⚡ Architecture 3: Hybrid Approaches
Combining Best of Both Worlds
Hybrid architectures use speech-to-speech for general conversation but switch to cascaded for specific tasks requiring custom models or precise control.
- • General conversation and chitchat
- • Quick FAQs and simple queries
- • Scenarios requiring empathy/emotion
- • Medical/legal transcription (compliance)
- • Domain-specific fine-tuned models
- • Complex tool calling workflows
Fallback Strategies
Production systems implement fallbacks to maintain reliability:
- 1. Primary: Speech-to-Speech → Fast, natural response for 80% of queries
- 2. Fallback: Cascaded → When S2S fails or encounters unsupported language
- 3. Emergency: Static responses → Pre-recorded messages if all services down
Example: Customer service bot uses GPT-4o Realtime for conversations, but switches to Deepgram STT + custom LLM + ElevenLabs TTS when accessing proprietary knowledge base requiring fine-tuned embeddings.
🎙️ VAD: Voice Activity Detection Deep Dive
What is VAD?
Voice Activity Detection (VAD) is the process of determining when someone is speaking versus silence or background noise. It's the foundation of natural turn-taking in voice conversations.
Why it matters: Without VAD, your agent doesn't know when to start listening, when to stop, or when the user has finished speaking. Poor VAD = awkward pauses, cut-off responses, or missed input.
Types of VAD
📊 Energy-Based
Simple threshold detection based on audio amplitude.
🤖 ML-Based
Neural networks trained to detect speech vs non-speech.
☁️ Server-Side
VAD handled by the voice platform (GPT-4o, Gemini).
🧠 Semantic/CSR
Understands language context to detect turn completion, not just silence.
Silero VAD: The Industry Standard
Silero VAD is a pre-trained enterprise-grade Voice Activity Detector that supports 6000+ languages and performs well on audios from different domains with various background noise and quality levels. One audio chunk (30+ ms) takes less than 1ms to process on a single CPU thread.
- • Supports 8000 Hz and 16000 Hz sampling rates
- • JIT model ~2MB in size
- • Published under MIT license (zero restrictions)
- • PyTorch-based, ONNX runtime available
- • <1ms CPU processing per 30ms chunk
- • Handles noisy environments well
- • Used by LiveKit, Pipecat, and major platforms
- • High accuracy speech detection
Silero VAD Implementation (PyTorch)
import torch
# Load Silero VAD model from torch hub
model, utils = torch.hub.load(
repo_or_dir='snakers4/silero-vad',
model='silero_vad',
force_reload=False
)
(get_speech_timestamps, save_audio, read_audio, VADIterator, collect_chunks) = utils
# Read audio file (16kHz recommended)
wav = read_audio('audio.wav', sampling_rate=16000)
# Get speech timestamps with configurable parameters
speech_timestamps = get_speech_timestamps(
wav,
model,
threshold=0.5, # Speech probability threshold
sampling_rate=16000,
min_speech_duration_ms=250, # Minimum speech segment
min_silence_duration_ms=100 # Minimum silence to split
)
print(speech_timestamps)
# Output: [{'start': 3200, 'end': 12800}, {'start': 16000, 'end': 24000}] Deepgram Flux: Semantic Turn Detection (CSR)
Deepgram Flux is the first production Conversational Speech Recognition (CSR) model that fuses transcription with turn detection into a single unified system. Instead of relying on silence thresholds, Flux understands semantic completeness — recognizing that "because..." means the user isn't finished, while "Thanks." signals turn completion.
- • Detects "has the user finished their thought?" vs "is there silence?"
- • Replaces separate STT + VAD + endpointing pipeline with one model
- • Emits StartOfTurn and EndOfTurn events natively
- • ~30% fewer false interruptions vs silence-based approaches
- • Median end-of-turn detection within 1.5s (p95)
- • Nova-3 level transcription accuracy maintained
- • Two config params: eot_threshold + eot_silence_threshold_ms
- • "Eager" mode: speculative LLM calls for 150-250ms faster responses
Deepgram Flux Integration (replaces VAD + STT pipeline)
import { createClient, LiveTranscriptionEvents } from "@deepgram/sdk";
const deepgram = createClient(process.env.DEEPGRAM_API_KEY);
// Single connection replaces separate VAD + STT + endpointing
const connection = deepgram.listen.live({
model: "nova-3",
smart_format: true,
// Flux semantic turn detection — replaces silence-based VAD
endpointing: "flux",
eot_threshold: 0.5, // Semantic completion confidence
eot_silence_threshold_ms: 800, // Fallback silence threshold
});
connection.on(LiveTranscriptionEvents.Transcript, (data) => {
const transcript = data.channel.alternatives[0].transcript;
if (data.is_final) console.log("Partial:", transcript);
});
// Native turn events — no separate VAD needed
connection.on(LiveTranscriptionEvents.EndOfTurn, () => {
console.log("User finished speaking — generate response");
});
connection.on(LiveTranscriptionEvents.StartOfTurn, () => {
console.log("User started speaking — stop agent audio");
}); When to use Flux over Silero: Flux is ideal when you want to eliminate the VAD+endpointing layer entirely and get turn detection baked into transcription. Silero remains better for client-side preprocessing, offline use, or when you need VAD independent of any cloud STT provider.
Key VAD Concepts
Speech Probability Threshold
The confidence level (0.0-1.0) required to classify audio as "speech". Lower = more sensitive (catches whispers but more false positives). Higher = more strict (misses quiet speech but fewer false triggers).
Min Speech/Silence Duration
Min speech duration: Ignore speech segments shorter than this (filters
out random noise spikes). Typical: 250ms.
Min silence duration: How long silence must last to consider speech ended.
Typical: 100-500ms depending on use case.
Padding (Pre/Post Speech)
Extra audio to include before/after detected speech to avoid cutting off the start or end of words. Pre-padding: ~100-200ms. Post-padding: ~100-300ms.
Turn-Taking and Interruption Handling
Turn detection determines when a user begins or ends their "turn" in a conversation, letting the agent know when to start listening and when to respond. Barge-in detection handles when users interrupt the agent mid-response.
- • Detect silence after speech ends
- • Tuned via min_silence_duration
- • Fast but prone to false triggers on pauses
- • Model understands if thought is complete
- • "because..." = not done, "Thanks." = done
- • ~30% fewer false cutoffs
- • VAD detects speech during agent playback
- • Immediately stop agent audio output
- • Buffer user speech, restart turn detection
Best practice: For silence-based VAD, process audio in 10-20ms intervals using Silero ONNX. For semantic turn detection, use Deepgram Flux with eot_threshold tuned to your use case — lower values respond faster but risk more interruptions.
🔧 Tool Calling / Function Calling in Voice AI
What is Tool Calling in Voice Context?
Tool calling (or function calling) in voice AI allows the agent to execute external actions during a conversation—checking databases, updating CRMs, pulling live data, sending emails, booking appointments—all while maintaining natural dialogue.
Examples: "Book me an appointment" → Agent calls calendar API.
"Check my balance" → Agent queries account database.
"Send an email to Sarah" → Agent triggers email service.
Architectures for Tool Calling
⏸️ Synchronous (User Waits)
Agent pauses conversation, executes tool, returns result, then continues.
Agent: [calls weather API]
Agent: "It's 72°F and sunny."
🔄 Asynchronous (Background Execution)
Agent continues conversation while tool executes in background.
Agent: "Sending that now. Anything else?"
[Email sends in background]
Implementation Patterns
In Cascaded Architecture
The LLM decides to call a tool, you execute it, then TTS speaks the result.
// Example: GPT-4 with function calling in cascaded pipeline
const completion = await openai.chat.completions.create({
model: "gpt-4",
messages: [
{ role: "user", content: "What's the weather in San Francisco?" }
],
tools: [
{
type: "function",
function: {
name: "get_weather",
description: "Get current weather for a location",
parameters: {
type: "object",
properties: {
location: { type: "string", description: "City name" }
},
required: ["location"]
}
}
}
]
});
// LLM decides to call get_weather function
const toolCall = completion.choices[0].message.tool_calls[0];
const weatherData = await getWeather(JSON.parse(toolCall.function.arguments).location);
// Feed result back to LLM
const finalResponse = await openai.chat.completions.create({
model: "gpt-4",
messages: [
{ role: "user", content: "What's the weather in San Francisco?" },
completion.choices[0].message,
{
role: "tool",
tool_call_id: toolCall.id,
content: JSON.stringify(weatherData)
}
]
});
// TTS speaks the final response
const audio = await elevenlabs.generate({
text: finalResponse.choices[0].message.content,
voice: "Rachel"
}); In Speech-to-Speech (GPT-4o Realtime)
Native function calling with asynchronous support—the model can continue a fluid conversation while waiting on tool results.
// Define tools in session configuration
ws.send(JSON.stringify({
type: "session.update",
session: {
tools: [
{
type: "function",
name: "check_order_status",
description: "Check the status of a customer order",
parameters: {
type: "object",
properties: {
order_id: { type: "string" }
},
required: ["order_id"]
}
}
]
}
}));
// When model calls a tool, you receive event
ws.on("message", async (data) => {
const event = JSON.parse(data);
if (event.type === "response.function_call_arguments.done") {
const { call_id, name, arguments } = event;
// Execute tool asynchronously
const result = await checkOrderStatus(JSON.parse(arguments).order_id);
// Send result back to session
ws.send(JSON.stringify({
type: "conversation.item.create",
item: {
type: "function_call_output",
call_id: call_id,
output: JSON.stringify(result)
}
}));
}
}); Providers with Tool Calling
Native async function calling. Model continues conversation while tools execute. Improved precision with gpt-realtime model.
Advanced function calling during conversations. Check databases, update CRMs, pull live data mid-call. Supports custom tool integrations.
Tools called automatically based on conversation context. Search web, query documents, execute business logic. 100+ languages supported.
Standardized tool connections. MCP-compliant servers use tools/list and tools/call protocol. Works with Realtime API.
🎯 Best Practices: Confirm before destructive actions ("Should I delete this?"), provide progress updates for long operations ("Checking that now..."), handle errors gracefully in voice context ("I couldn't find that, could you try again?"), use async for long-running tools to avoid awkward silence.
⚡ Latency Optimization
The Human Conversation Threshold
Human conversation requires a 300-500ms response window to feel natural. Delays beyond 500 milliseconds feel unnatural. Based on analysis of 4M+ production voice agent calls, hitting this threshold requires aggressive optimization at every stage.
- • Cascaded: 2-4s typical end-to-end (exceeds natural threshold)
- • Cascaded optimized: 500-800ms achievable with streaming
- • Speech-to-Speech: Sub-1s typical (within natural range)
Where Latency Comes From
Streaming reduces this to ~100-200ms. Batch processing can take 500ms+.
The LLM is typically the highest latency component. Model size, prompt complexity, and provider speed all matter.
Streaming TTS dramatically reduces perceived latency. Time to first byte (TTFB) matters more than total synthesis time.
Each API call adds network latency. Cascaded architecture has 3+ round-trips.
ML-based VAD (Silero) adds minimal latency when optimized. Energy-based VAD is nearly instant.
Optimization Techniques
1. Streaming at Every Stage
STT streams partial transcripts, LLM generates tokens incrementally, TTS synthesizes audio chunks as tokens arrive. This eliminates waiting for complete processing.
2. Speculative Generation
Start TTS synthesis before LLM completes full response. If LLM changes direction, discard speculative audio. Works well for predictable responses.
3. Edge Deployment
Deploy latency-sensitive components (VAD, lightweight STT) on edge nodes close to users. Heavy LLM inference can stay centralized.
4. Connection Pooling / Keep-Alive
Use WebSocket persistent connections instead of HTTP request/response. Eliminate TCP handshake and TLS negotiation overhead on every interaction.
Latency Budget Breakdown
To hit the 300ms natural conversation threshold, here's how to allocate your latency budget across a cascaded architecture with aggressive streaming optimization:
| Component | Typical (No Opt) | Optimized (Streaming) | Optimization |
|---|---|---|---|
| VAD Detection | 10-20ms | 1-5ms | ONNX runtime, CPU optimization |
| STT Processing | 300-500ms | 100-150ms | Streaming STT (Deepgram Nova-3) |
| LLM First Token | 500-2000ms | 100-200ms | Fast inference (Groq), small prompts |
| TTS First Byte | 200-800ms | 40-75ms | Cartesia Turbo, ElevenLabs Flash |
| Network Overhead | 150-300ms | 50-100ms | WebSocket, edge deployment, co-location |
| Total End-to-End | 2-4 seconds | 300-530ms | Streaming + fast providers |
* Latencies are approximate and vary based on network conditions, model selection, and implementation quality. These represent production-tested ranges from real-world deployments.
🎯 Choosing Your Architecture: Decision Framework
Decision Tree
→ Yes: Use Speech-to-Speech (GPT-4o Realtime, Gemini 2.5 Flash, Grok)
→ No: Continue...
→ Yes: Use Cascaded with approved STT/LLM/TTS
→ No: Continue...
→ Yes: Use Cascaded with custom models
→ No: Continue...
→ Yes: Use Cascaded (inspect text at each stage)
→ No: Use Speech-to-Speech for simplicity and speed
🔗 Choose Cascaded
- Medical/Legal: Compliance requires specific certified models
- Enterprise B2B: Custom fine-tuned models for domain jargon
- High-stakes debugging: Need to inspect intermediate outputs
- Best-in-class quality: Willing to trade latency for top providers
🚀 Choose Speech-to-Speech
- Consumer apps: Chatbots, virtual assistants, customer service
- Low latency priority: Natural conversation feel is critical
- Emotional contexts: Therapy, coaching, empathetic interactions
- Simple workflows: General Q&A, FAQs, basic assistance
⚡ Choose Hybrid
- Complex enterprise: Mix of general conversation + specific tasks
- Fallback scenarios: S2S primary, cascaded when it fails
- Cost optimization: S2S for most, cascaded for cheaper specific tasks
- Phased migration: Start cascaded, migrate to S2S gradually
Cost Considerations at Scale
At production scale (100k+ conversations/month), pricing models differ significantly:
| Architecture | Typical Cost/Min | 100k hrs/mo Cost | Notes |
|---|---|---|---|
| Cascaded (Premium) | $0.23-$0.33/min | $1.4M-$2M | Deepgram + GPT-4 + ElevenLabs |
| Cascaded (Budget) | $0.10-$0.15/min | $600k-$900k | Whisper + Llama + Cartesia |
| Speech-to-Speech (GPT-4o) | Varies by tokens | ~$1M | $32/$64 per 1M audio tokens |
| Speech-to-Speech (Grok) | $0.05/min | $300k | Flat rate, all-in-one |
* Costs are estimates based on published pricing as of February 2026. Actual costs vary based on conversation length, features used, and negotiated enterprise pricing.
💡 Hidden cost consideration: Cascaded architectures require managing 3+ vendor relationships, separate invoices, and integration maintenance. Speech-to-speech platforms (Grok, GPT-4o Realtime) offer simpler billing but less flexibility. Factor in engineering time when comparing costs.
❓ Frequently Asked Questions
What is the difference between cascaded and speech-to-speech voice AI architectures?
Cascaded architecture (STT → LLM → TTS) processes voice in three separate stages: speech-to-text transcription, language model response generation, and text-to-speech synthesis. Speech-to-speech models like GPT-4o Realtime process audio directly without intermediate text, achieving sub-1s latency but offering less granular control. Cascaded offers maximum control and debuggability (2-4s latency typical), while speech-to-speech prioritizes natural conversation speed (<1s) at the cost of flexibility.
What is VAD (Voice Activity Detection) and why does it matter?
Voice Activity Detection (VAD) determines when someone is speaking versus silence or background noise. It's critical for natural turn-taking in voice conversations—knowing when to start listening, when to stop, and when the user has finished speaking. Without proper VAD, agents produce awkward pauses, cut-off responses, or missed input. Modern ML-based VAD like Silero VAD processes 30ms audio chunks in <1ms on CPU, handling noisy environments with high accuracy across 6000+ languages.
How can I achieve sub-500ms latency for natural voice conversations?
Hitting the 300-500ms natural conversation threshold requires: (1) Streaming at every stage—STT streams partial transcripts, LLM generates tokens incrementally, TTS synthesizes chunks as they arrive. (2) Fast provider selection—Deepgram Nova-3 for STT (100-150ms), Groq for LLM inference (100-200ms first token), Cartesia Turbo or ElevenLabs Flash for TTS (40-75ms TTFB). (3) Network optimization—WebSocket persistent connections, edge deployment, service co-location. (4) Or use speech-to-speech models (GPT-4o Realtime, Gemini 2.5 Flash) which achieve sub-1s naturally.
When should I use cascaded architecture vs speech-to-speech?
Choose cascaded when: (1) You need specific compliance or certified models (healthcare, legal), (2) Fine-tuned domain models are critical (medical terminology, industry jargon), (3) Debugging/observability is essential (inspect intermediate text), (4) You want best-in-class components mixed together. Choose speech-to-speech when: (1) Sub-500ms latency is critical (consumer apps), (2) Emotional context matters (therapy, empathetic conversations), (3) Simple general Q&A workflows, (4) Natural interruption handling is important. Use hybrid for complex enterprise scenarios requiring both.
How does tool calling work in voice AI?
Tool calling (function calling) allows voice agents to execute external actions during conversation—checking databases, updating CRMs, sending emails, booking appointments. In cascaded architectures, the LLM decides to call a tool, you execute it, then TTS speaks the result. In speech-to-speech (GPT-4o Realtime), native async function calling lets the model continue conversation while tools execute in background. Modern platforms like Vapi, Grok Voice API, and OpenAI Realtime API support tool calling with varying levels of synchronous vs asynchronous execution.
What is Silero VAD and how do I implement it?
Silero VAD is a pre-trained enterprise-grade Voice Activity Detector supporting 6000+ languages with <1ms processing time per 30ms audio chunk on CPU. It uses PyTorch-based deep neural networks to classify speech vs non-speech with high accuracy in noisy environments. Implementation: Install PyTorch, load model from torch.hub (repo: snakers4/silero-vad), configure threshold (0.5 balanced), min speech duration (250ms), and min silence duration (100-500ms). The model is ~2MB, MIT licensed, and widely used by LiveKit, Pipecat, and production voice platforms.
📚 Related Reading
Ready to Build Production Voice AI?
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Last updated: February 5, 2026 • Research sourced from OpenAI, Google, Silero, AssemblyAI, and production voice AI platforms