voice-agents

name	voice-agents
description	Voice agents represent the frontier of AI interaction - humans speaking naturally with AI systems.
risk	safe
source	vibeship-spawner-skills (Apache 2.0)
date_added	"2026-02-27T00:00:00.000Z"

Voice Agents

Voice agents represent the frontier of AI interaction - humans speaking naturally with AI systems. The challenge isn't just speech recognition and synthesis, it's achieving natural conversation flow with sub-800ms latency while handling interruptions, background noise, and emotional nuance.

This skill covers two architectures: speech-to-speech (OpenAI Realtime API, lowest latency, most natural) and pipeline (STT→LLM→TTS, more control, easier to debug). Key insight: latency is the constraint. Humans expect responses in 500ms. Every millisecond matters.

84% of organizations are increasing voice AI budgets in 2025. This is the year voice agents go mainstream.

Principles

• Latency is the constraint - target <800ms end-to-end
• Jitter (variance) matters as much as absolute latency
• VAD quality determines conversation flow
• Interruption handling makes or breaks the experience
• Start with focused MVP, iterate based on real conversations
• Combine best-in-class components (Deepgram STT + ElevenLabs TTS)

Capabilities

• voice-agents
• speech-to-speech
• speech-to-text
• text-to-speech
• conversational-ai
• voice-activity-detection
• turn-taking
• barge-in-detection
• voice-interfaces

Scope

• phone-system-integration → backend
• audio-processing-dsp → audio-specialist
• music-generation → audio-specialist
• accessibility-compliance → accessibility-specialist

Tooling

Speech_to_speech

• OpenAI Realtime API - When: Lowest latency, most natural conversation Note: gpt-4o-realtime-preview, native voice, sub-500ms
• Pipecat - When: Open-source voice orchestration Note: Daily-backed, enterprise-grade, modular

Speech_to_text

• OpenAI Whisper - When: Highest accuracy, multilingual Note: gpt-4o-transcribe for best results
• Deepgram Nova-3 - When: Production workloads, 54% lower WER Note: 150-184ms TTFT, 90%+ accuracy on noisy audio
• AssemblyAI - When: Real-time streaming, speaker diarization Note: Good accuracy-latency balance

Text_to_speech

• ElevenLabs - When: Most natural voice, emotional control Note: Flash model 75ms latency, V3 for expression
• OpenAI TTS - When: Integrated with OpenAI stack Note: gpt-4o-mini-tts, 13 voices, streaming
• Deepgram Aura-2 - When: Cost-effective production TTS Note: 40% cheaper than ElevenLabs, 184ms TTFB

Frameworks

• Pipecat - When: Open-source voice agent orchestration Note: Silero VAD, SmartTurn, interruption handling
• Vapi - When: Managed voice agent platform Note: No infrastructure management
• Retell AI - When: Low-latency voice agents Note: Best context preservation on interruption

Patterns

Speech-to-Speech Architecture

Direct audio-to-audio processing for lowest latency

When to use: Maximum naturalness, emotional preservation, real-time conversation

SPEECH-TO-SPEECH ARCHITECTURE:

""" [User Audio] → [S2S Model] → [Agent Audio]

Advantages:

• Lowest latency (sub-500ms)
• Preserves emotion, emphasis, accents
• Most natural conversation flow

Disadvantages:

• Less control over responses
• Harder to debug/audit
• Can't easily modify what's said """

OpenAI Realtime API

""" import { RealtimeClient } from '@openai/realtime-api-beta';

const client = new RealtimeClient({ apiKey: process.env.OPENAI_API_KEY, });

// Configure for voice conversation client.updateSession({ modalities: ['text', 'audio'], voice: 'alloy', input_audio_format: 'pcm16', output_audio_format: 'pcm16', instructions: You are a helpful customer service agent. Be concise and friendly. If you don't know something, say so rather than making things up., turn_detection: { type: 'server_vad', // or 'semantic_vad' threshold: 0.5, prefix_padding_ms: 300, silence_duration_ms: 500, }, });

// Handle audio streams client.on('conversation.item.input_audio_transcription', (event) => { console.log('User said:', event.transcript); });

client.on('response.audio.delta', (event) => { // Stream audio to speaker audioPlayer.write(Buffer.from(event.delta, 'base64')); });

// Send user audio client.appendInputAudio(audioBuffer); """

Use Cases:

• Real-time customer support
• Voice assistants
• Interactive voice response (IVR)
• Live language translation

Pipeline Architecture

Separate STT → LLM → TTS for maximum control

When to use: Need to know/control exactly what's said, debugging, compliance

PIPELINE ARCHITECTURE:

""" [Audio] → [STT] → [Text] → [LLM] → [Text] → [TTS] → [Audio]

Advantages:

• Full control at each step
• Can log/audit all text
• Easier to debug
• Mix best-in-class components

Disadvantages:

• Higher latency (700-1200ms typical)
• Loses some emotion/nuance
• More components to manage """

Production Pipeline Example

""" import { Deepgram } from '@deepgram/sdk'; import { ElevenLabsClient } from 'elevenlabs'; import OpenAI from 'openai';

// Initialize clients const deepgram = new Deepgram(process.env.DEEPGRAM_API_KEY); const elevenlabs = new ElevenLabsClient(); const openai = new OpenAI();

async function processVoiceInput(audioStream) { // 1. Speech-to-Text (Deepgram Nova-3) const transcription = await deepgram.transcription.live({ model: 'nova-3', punctuate: true, endpointing: 300, // ms of silence before end });

transcription.on('transcript', async (data) => { if (data.is_final && data.speech_final) { const userText = data.channel.alternatives[0].transcript; console.log('User:', userText);

  // 2. LLM Processing
  const completion = await openai.chat.completions.create({
    model: 'gpt-4o-mini',
    messages: [
      { role: 'system', content: 'You are a concise voice assistant.' },
      { role: 'user', content: userText }
    ],
    max_tokens: 150,  // Keep responses short for voice
  });

  const agentText = completion.choices[0].message.content;
  console.log('Agent:', agentText);

  // 3. Text-to-Speech (ElevenLabs)
  const audioStream = await elevenlabs.textToSpeech.stream({
    voice_id: 'voice_id_here',
    text: agentText,
    model_id: 'eleven_flash_v2_5',  // Lowest latency
  });

  // Stream to user
  playAudioStream(audioStream);
}

});

// Pipe audio to transcription audioStream.pipe(transcription); } """

Optimization Tips:

• Start TTS while LLM still generating (streaming)
• Pre-compute first response segment during user speech
• Use Flash/turbo models for latency

Voice Activity Detection Pattern

Detect when user starts/stops speaking

When to use: All voice agents need VAD for turn-taking

VOICE ACTIVITY DETECTION (VAD):

""" VAD Types:

1. Energy-based: Simple, fast, noise-sensitive
2. Model-based: Silero VAD, more accurate
3. Semantic VAD: Understands meaning, best for conversation """

Silero VAD (Popular Open Source)

""" import { SileroVAD } from '@pipecat-ai/silero-vad';

const vad = new SileroVAD({ threshold: 0.5, // Speech probability threshold min_speech_duration: 250, // ms before speech confirmed min_silence_duration: 500, // ms of silence = end of turn });

vad.on('speech_start', () => { console.log('User started speaking'); // Stop any playing TTS (barge-in) audioPlayer.stop(); });

vad.on('speech_end', () => { console.log('User finished speaking'); // Trigger response generation processTranscript(); });

// Feed audio to VAD audioStream.on('data', (chunk) => { vad.process(chunk); }); """

OpenAI Semantic VAD

""" // In Realtime API session config client.updateSession({ turn_detection: { type: 'semantic_vad', // Uses meaning, not just silence // Model waits longer after "ummm..." // Responds faster after "Yes, that's correct." }, }); """

Barge-In Handling

""" // When user interrupts: function handleBargeIn() { // 1. Stop TTS immediately audioPlayer.stop();

// 2. Cancel pending LLM generation llmController.abort();

// 3. Reset state conversationState.checkpoint();

// 4. Listen to new input startListening(); }

// VAD triggers barge-in vad.on('speech_start', () => { if (audioPlayer.isPlaying) { handleBargeIn(); } }); """

Latency Optimization Pattern

Achieving <800ms end-to-end response time

When to use: Production voice agents

LATENCY OPTIMIZATION:

""" Target Metrics:

• End-to-end: <800ms (ideal: <500ms)
• Time-to-First-Token (TTFT): <300ms
• Barge-in response: <200ms
• Jitter variance: <100ms std dev """

Pipeline Latency Breakdown

""" Typical breakdown:

• VAD processing: 50-100ms
• STT first result: 150-200ms
• LLM TTFT: 100-300ms
• TTS TTFA: 75-200ms
• Audio buffering: 50-100ms

Total: 425-900ms """

Optimization Strategies

1. Streaming Everything

""" // Stream STT results as they come stt.on('partial_transcript', (text) => { // Start processing before final transcript llmPreprocessor.prepare(text); });

// Stream LLM output to TTS const llmStream = await openai.chat.completions.create({ stream: true, // ... });

for await (const chunk of llmStream) { tts.appendText(chunk.choices[0].delta.content); } """

2. Pre-computation

""" // While user is speaking, predict and prepare stt.on('partial_transcript', async (text) => { // Pre-fetch relevant context const context = await retrieveContext(text);

// Pre-compute likely first sentence const firstSentence = await generateOpener(context); }); """

3. Use Low-Latency Models

""" // STT: Deepgram Nova-3 (150ms TTFT) // LLM: gpt-4o-mini (fastest GPT-4 class) // TTS: ElevenLabs Flash (75ms) or Deepgram Aura-2 (184ms) """

4. Edge Deployment

""" // Run inference closer to user // - Cloud regions near user // - Edge computing for VAD/STT // - WebSocket over HTTP for lower overhead """

Conversation Design Pattern

Designing natural voice conversations

When to use: Building voice UX

CONVERSATION DESIGN:

Voice-First Principles

""" Voice is different from text:

• No undo button - say it right the first time
• Linear - user can't scroll back
• Ephemeral - easy to miss information
• Emotional - tone matters as much as words """

Response Design

"""

Keep responses short (10-20 seconds max)

Front-load the answer

Use signposting for lists

Bad: "I found several options. The first is... second is..." Good: "I found 3 options. Want me to go through them?"

Confirm understanding

Bad: "I'll transfer $500 to John." Good: "So that's $500 to John Smith. Should I proceed?" """

Prompting for Voice

""" system_prompt = ''' You are a voice assistant. Follow these rules:

1. Be concise - keep responses under 30 words
2. Use natural speech - contractions, casual language
3. Never use formatting (bullets, numbers in lists)
4. Spell out numbers and abbreviations
5. End with a question to keep conversation flowing
6. If unclear, ask for clarification
7. Never say "I'm an AI" unless asked

Good: "Got it. I'll set that reminder for three pm. Anything else?" Bad: "I have set a reminder for 3:00 PM. Is there anything else I can assist you with today?" ''' """

Error Recovery

""" // Handle recognition errors gracefully const errorResponses = { no_speech: "I didn't catch that. Could you say it again?", unclear: "Sorry, I'm not sure I understood. You said [repeat]. Is that right?", timeout: "Still there? I'm here when you're ready.", };

// Always offer human fallback for complex issues if (confidenceScore < 0.6) { response = "I want to make sure I get this right. Would you like to speak with a human agent?"; } """

Sharp Edges

Response Latency Exceeds 800ms

Severity: CRITICAL

Situation: Building a voice agent pipeline

Symptoms: Conversations feel awkward. Users repeat themselves. "Are you there?" questions. Users hang up or give up. Low satisfaction scores despite correct answers.

Why this breaks: In human conversation, responses typically arrive within 500ms. Anything over 800ms feels like the agent is slow or confused. Users lose confidence and patience. Every component adds latency: VAD (100ms) + STT (200ms) + LLM (300ms) + TTS (200ms) = 800ms.

Recommended fix:

Measure and budget latency for each component:

Target latencies:

• VAD processing: <100ms
• STT time-to-first-token: <200ms
• LLM time-to-first-token: <300ms
• TTS time-to-first-audio: <150ms
• Total end-to-end: <800ms

Optimization strategies:

1. Use low-latency models:

   • STT: Deepgram Nova-3 (150ms) vs Whisper (500ms+)
   • TTS: ElevenLabs Flash (75ms) vs standard (200ms+)
   • LLM: gpt-4o-mini streaming

2. Stream everything:

   • Don't wait for full STT transcript
   • Stream LLM output to TTS
   • Start audio playback before TTS finishes

3. Pre-compute:

   • While user speaks, prepare context
   • Generate opening phrase in parallel

4. Edge deployment:

   • Run VAD/STT at edge
   • Use nearest cloud region

Measure continuously:

Log timestamps at each stage, track P50/P95 latency

Response Time Variance Disrupts Rhythm

Severity: HIGH

Situation: Voice agent with inconsistent response times

Symptoms: Conversations feel unpredictable. User doesn't know when to speak. Sometimes agent responds immediately, sometimes after long pause. Users talk over agent. Agent talks over users.

Why this breaks: Jitter (variance in response time) disrupts conversational rhythm more than absolute latency. Consistent 800ms feels better than alternating 400ms and 1200ms. Users can't adapt to unpredictable timing.

Recommended fix:

Target jitter metrics:

• Standard deviation: <100ms
• P95-P50 gap: <200ms

Reduce jitter sources:

1. Consistent model loading:

   • Keep models warm
   • Pre-load on connection start

2. Buffer audio output:

   • Small buffer (50-100ms) smooths playback
   • Don't start playing until buffer filled

3. Handle LLM variance:

   • gpt-4o-mini more consistent than larger models
   • Set max_tokens to limit long responses

4. Monitor and alert:

   • Track response time distribution
   • Alert on jitter spikes

Implementation:

const MIN_RESPONSE_TIME = 400; // ms

async function respondWithConsistentTiming(text) { const startTime = Date.now(); const audio = await generateSpeech(text);

const elapsed = Date.now() - startTime; if (elapsed < MIN_RESPONSE_TIME) { await delay(MIN_RESPONSE_TIME - elapsed); }

playAudio(audio); }

Using Silence Duration for Turn Detection

Severity: HIGH

Situation: Detecting when user finishes speaking

Symptoms: Agent interrupts user mid-thought. Or waits too long after user finishes. "Let me think..." triggers premature response. Short answers have awkward pause before response.

Why this breaks: Simple silence detection (e.g., "end turn after 500ms silence") doesn't understand conversation. Humans pause mid-sentence. "Yes." needs fast response, "Well, let me think about that..." needs patience. Fixed timeout fits neither.

Recommended fix:

Use semantic VAD:

OpenAI Semantic VAD:

client.updateSession({ turn_detection: { type: 'semantic_vad', // Waits longer after "umm..." // Responds faster after "Yes, that's correct." }, });

Pipecat SmartTurn:

const pipeline = new Pipeline({ vad: new SileroVAD(), turnDetection: new SmartTurn(), });

// SmartTurn considers: // - Speech content (complete sentence?) // - Prosody (falling intonation?) // - Context (question asked?)

Fallback: Adaptive silence threshold:

function calculateSilenceThreshold(transcript) { const endsWithComplete = transcript.match(/[.!?]$/); const hasFillers = transcript.match(/um|uh|like|well/i);

if (endsWithComplete && !hasFillers) { return 300; // Fast response } else if (hasFillers) { return 1500; // Wait for continuation } return 700; // Default }

Agent Doesn't Stop When User Interrupts

Severity: HIGH

Situation: User tries to interrupt agent mid-sentence

Symptoms: Agent talks over user. User has to wait for agent to finish. Frustrating experience. Users give up and abandon call. "STOP! STOP!" doesn't work.

Why this breaks: Without barge-in handling, the TTS plays to completion regardless of user input. This violates basic conversational norms - in human conversation, we stop when interrupted.

Recommended fix:

Implement barge-in detection:

Basic barge-in:

vad.on('speech_start', () => { if (ttsPlayer.isPlaying) { // 1. Stop audio immediately ttsPlayer.stop();

// 2. Cancel pending TTS generation
ttsController.abort();

// 3. Checkpoint conversation state
conversationState.save();

// 4. Listen to new input
startTranscription();

} });

Advanced: Distinguish interruption types:

vad.on('speech_start', async () => { if (!ttsPlayer.isPlaying) return;

// Wait 200ms to get first words await delay(200); const firstWords = getTranscriptSoFar();

if (isBackchannel(firstWords)) { // "uh-huh", "yeah" - don't interrupt return; }

if (isClarification(firstWords)) { // "What?", "Sorry?" - repeat last sentence repeatLastSentence(); } else { // Real interruption - stop and listen handleFullInterruption(); } });

Response time target:

• Barge-in response: <200ms
• User should feel heard immediately

Generating Text-Length Responses for Voice

Severity: MEDIUM

Situation: Prompting LLM for voice agent responses

Symptoms: Agent rambles. Users lose track of information. "Can you repeat that?" requests. Users interrupt to ask for shorter version. Low comprehension of conveyed information.

Why this breaks: Text can be scanned and re-read. Voice is linear and ephemeral. A 3-paragraph response that works in chat is overwhelming in voice. Users can only hold ~7 items in working memory.

Recommended fix:

Constrain response length in prompts:

system_prompt = ''' You are a voice assistant. Keep responses UNDER 30 WORDS. For complex information, break into chunks and confirm understanding between each.

Instead of: "Here are the three options. First, you could... Second... Third..."

Say: "I found 3 options. Want me to go through them?"

Never list more than 3 items without pausing for confirmation. '''

Enforce at generation:

const response = await openai.chat.completions.create({ max_tokens: 100, // Hard limit // ... });

Chunking pattern:

if (information.length > 3) { response = I have ${information.length} items. Let's go through them one at a time. First: ${information[0]}. Ready for the next?; }

Progressive disclosure:

"I found your account. Want the balance, recent transactions, or something else?" // Don't dump all info at once

Using Bullets/Numbers/Markdown in Voice

Severity: MEDIUM

Situation: Formatting LLM output for voice

Symptoms: "First bullet point: item one" read aloud. Numbers read as "one two three" instead of "one, two, three." Markdown artifacts in speech. Robotic, unnatural delivery.

Why this breaks: TTS models read what they're given. Text formatting intended for visual display sounds robotic when read aloud. Users can't "see" structure in audio.

Recommended fix:

Prompt for spoken format:

system_prompt = ''' Format responses for SPOKEN delivery:

• No bullet points, numbered lists, or markdown
• Spell out numbers: "twenty-three" not "23"
• Spell out abbreviations: "United States" not "US"
• Use verbal signposting: "There are three things. First..."
• Never use asterisks, dashes, or special characters '''

Post-processing:

function prepareForSpeech(text) { return text // Remove markdown .replace(/[*_#`]/g, '') // Convert numbers .replace(/\d+/g, numToWords) // Expand abbreviations .replace(/\betc\b/gi, 'et cetera') .replace(/\be.g./gi, 'for example') // Add pauses .replace(/. /g, '... ') .replace(/, /g, '... '); }

SSML for precise control:

The total is $49.99. Want to proceed?

VAD/STT Fails in Noisy Enprojectnments

Severity: MEDIUM

Situation: Users in cars, cafes, outdoors

Symptoms: "I didn't catch that" frequently. Background noise triggers false starts. Fan/AC causes continuous listening. Car engine noise confuses STT.

Why this breaks: Default VAD thresholds work for quiet enprojectnments. Real-world usage includes background noise that triggers false positives or masks speech, causing false negatives.

Recommended fix:

Implement noise handling:

1. Noise reduction in STT:

const transcription = await deepgram.transcription.live({ model: 'nova-3', noise_reduction: true, // or smart_format: true, });

2. Adaptive VAD threshold:

// Measure ambient noise level const ambientLevel = measureAmbientNoise(5000); // 5 sec sample

vad.setThreshold(ambientLevel * 1.5); // Above ambient

3. Confidence filtering:

stt.on('transcript', (data) => { if (data.confidence < 0.7) { // Low confidence - probably noise askForRepeat(); return; } processTranscript(data.transcript); });

4. Echo cancellation:

// Prevent agent's voice from being transcribed const echoCanceller = new EchoCanceller(); echoCanceller.reference(ttsOutput); const cleanedAudio = echoCanceller.process(userAudio);

STT Produces Incorrect or Hallucinated Text

Severity: MEDIUM

Situation: Processing unclear or accented speech

Symptoms: Agent responds to something user didn't say. Names consistently wrong. Technical terms misheard. "I said X, not Y" frustration.

Why this breaks: STT models can hallucinate, especially on proper nouns, technical terms, or accented speech. These errors propagate through the pipeline and produce nonsensical responses.

Recommended fix:

Mitigate STT errors:

1. Use keywords/biasing:

const transcription = await deepgram.transcription.live({ keywords: ['Acme Corp', 'ProductName', 'John Smith'], keyword_boost: 'high', });

2. Confirmation for critical info:

if (containsNameOrNumber(transcript)) { response = I heard "${name}". Is that correct?; }

3. Confidence-based fallback:

if (confidence < 0.8) { response = I think you said "${transcript}". Did I get that right?; }

4. Multiple hypothesis handling:

// Some STT APIs return n-best list const alternatives = transcription.alternatives; if (alternatives[0].confidence - alternatives[1].confidence < 0.1) { // Ambiguous - ask for clarification }

5. Error correction patterns:

promptPattern = User may correct previous mistakes. If they say "no, I said X" or "not Y, Z", update your understanding accordingly.;

Validation Checks

Missing Latency Measurement

Severity: ERROR

Voice agents must track latency at each stage

Message: Voice pipeline without latency tracking. Add timestamps at each stage to measure performance.

Using Batch STT Instead of Streaming

Severity: WARNING

Streaming STT reduces latency significantly

Message: Using batch transcription. Consider streaming for lower latency in voice agents.

TTS Without Streaming Output

Severity: WARNING

Streaming TTS reduces time to first audio

Message: TTS without streaming. Stream audio to reduce time to first audio.

Hardcoded VAD Silence Threshold

Severity: WARNING

Fixed silence thresholds don't adapt to conversation

Message: Fixed silence threshold. Consider semantic VAD or adaptive thresholds for better turn-taking.

Missing Barge-In Handling

Severity: WARNING

Voice agents should stop when user interrupts

Message: VAD without barge-in handling. Stop TTS when user starts speaking.

Voice Prompt Without Length Constraints

Severity: WARNING

Voice prompts should constrain response length

Message: Voice prompt without length constraints. Add 'Keep responses under 30 words' to system prompt.

Markdown Formatting Sent to TTS

Severity: WARNING

Markdown will be read literally by TTS

Message: Check for markdown in TTS input. Strip formatting before sending to TTS.

STT Without Error Handling

Severity: WARNING

STT can fail or return low confidence

Message: STT without error handling. Check confidence scores and handle failures.

WebSocket Without Reconnection

Severity: WARNING

Realtime APIs need reconnection handling

Message: Realtime connection without reconnection logic. Handle disconnects gracefully.

Missing Noise Handling

Severity: INFO

Real-world audio includes background noise

Message: Consider adding noise handling for real-world audio quality.

Collaboration

Delegation Triggers

• user needs phone/telephony integration -> backend (Twilio, Vonage, SIP integration)
• user needs LLM optimization -> llm-architect (Model selection, prompting, fine-tuning)
• user needs tools for voice agent -> agent-tool-builder (Tool design for voice context)
• user needs multi-agent voice system -> multi-agent-orchestration (Voice agents working together)
• user needs accessibility compliance -> accessibility-specialist (Voice interface accessibility)

Related Skills

Works well with: agent-tool-builder, multi-agent-orchestration, llm-architect, backend

When to Use

• User mentions or implies: voice agent
• User mentions or implies: speech to text
• User mentions or implies: text to speech
• User mentions or implies: whisper
• User mentions or implies: elevenlabs
• User mentions or implies: deepgram
• User mentions or implies: realtime api
• User mentions or implies: voice assistant
• User mentions or implies: voice ai
• User mentions or implies: conversational ai
• User mentions or implies: tts
• User mentions or implies: stt
• User mentions or implies: asr

Limitations

• Use this skill only when the task clearly matches the scope described above.
• Do not treat the output as a substitute for enprojectnment-specific validation, testing, or expert review.
• Stop and ask for clarification if required inputs, permissions, safety boundaries, or success criteria are missing.