
Voice AI agents are becoming increasingly common in applications such as customer support automation, AI call centers, and real-time conversational assistants. Modern voice systems can process speech in real time, understand conversational context, handle interruptions, and respond with natural-sounding speech while maintaining low latency.
I wanted to understand what it actually takes to build a production-ready voice AI agent using modern tools.
In this guide, I explain how to build a voice AI agent using LiveKit Agents, an open-source framework designed for real-time voice applications. The goal is not just to build a prototype, but to understand the architecture, core components, and practical considerations required to run voice agents reliably at scale.
Voice AI agents are software systems that can understand spoken language, process the request using artificial intelligence, and respond with synthesized speech in real time.
They combine technologies such as speech recognition (STT), large language models (LLMs), and text-to-speech (TTS) to enable natural, conversational interactions between humans and machines.
Voice AI agents are commonly used in applications like AI assistants, customer support automation, voice-enabled applications, and AI call centers, where users interact through speech instead of traditional text or graphical interfaces.
A production-ready voice AI agent must listen to spoken input, detect when a user is speaking, convert speech to text, process the request using a large language model (LLM), and generate a natural speech response. All of this must happen in real time, typically targeting under 500ms latency, while also handling interruptions during conversation.
Frameworks such as LiveKit Agents treat the voice agent as a WebRTC participant, enabling real-time bidirectional audio streaming and support for multimodal interactions.
A typical voice AI system includes:
A basic prototype can often be built in 1–2 hours, while production deployments usually require additional time for testing, scaling, and infrastructure setup.
Before building a voice AI agent, make sure the following tools and resources are available.
Programming languagesPython 3.9+ (recommended) or Node.js 18+
HardwareA standard laptop is enough for development. A GPU is helpful for local models or high-concurrency testing.
Accounts and API keys
Development tools
Browser support
A modern browser such as Chrome or Firefox is recommended for testing WebRTC-based voice interactions.
A basic setup usually takes 15–30 minutes. For quick experimentation, LiveKit also provides Agent Builder, a browser-based tool that allows you to prototype voice agents without writing code.
There are two common architectures used to build voice AI agents.
Audio → VAD/STT → LLM → TTS → Audio
This is the most common approach for production systems. Each component in the pipeline can be customized depending on the use case.
For example:
Frameworks such as LiveKit Agents manage this pipeline through AgentSession, handling streaming audio, interruptions, and conversation state. This approach works well for applications that require RAG, tool calling, or complex workflows.
In this approach, a single multimodal model processes audio input and generates audio output directly.
Examples include:
These models preserve speech characteristics such as prosody, emotion, and accents, and can achieve very low latency (often under 200ms). However, they provide less control over individual components compared to pipeline architectures.
For most production systems, the cascaded pipeline architecture provides greater control, scalability, and flexibility.
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Realtime speech-to-speech models are useful for low-latency conversational demos or rapid prototyping, and they can also be integrated into hybrid pipelines when needed.
A voice AI agent typically relies on several core components that work together in a real-time pipeline.
Speech-to-Text (STT) – Converts user speech into text.Examples: AssemblyAI Universal Streaming, Deepgram Nova-2.
Voice Activity Detection (VAD) – Detects when a user starts and stops speaking.Common choice: Silero VAD.
Turn Detection – Determines when the user has finished speaking so the agent can respond.LiveKit’s MultilingualModel improves speech completion detection across languages.
Large Language Model (LLM) – Interprets the request, generates responses, and handles reasoning, tool calls, or RAG.Examples: GPT-4.1-mini, Groq-hosted Llama 3.1, Claude, xAI Grok.
Text-to-Speech (TTS) – Converts the generated response into natural audio output.Examples: Cartesia Sonic-3, ElevenLabs Turbo v2, Rime.
Noise Cancellation – Cleans background noise for clearer voice input.Options: BVC (general use) and BVCTelephony (for phone calls).
Observability & Monitoring – Tracks transcripts, latency, and performance during conversations.
Developers typically build voice AI agents using either open-source frameworks or managed platforms. The right approach depends on the level of control, customization, and infrastructure management required.
| Aspect | Open-Source Frameworks | Platform-Based Tools |
Control | Full control over the voice pipeline and infrastructure | Limited customization |
Flexibility | High — supports RAG, tools, and custom workflows | Restricted to platform features |
Setup Speed | Requires development setup | Faster to deploy |
Infrastructure | Must be managed manually | Fully managed |
Cost | Usually lower long-term | Often higher due to platform pricing |
Examples | LiveKit Agents, Pipecat | Vapi, Retell, Bland |
For developers building custom or production voice AI systems, open-source frameworks like LiveKit Agents offer greater flexibility and control.
Managed platforms can be useful for rapid prototyping, but they may introduce higher costs and vendor lock-in as systems scale.
Install LiveKit CLI & Authenticate:
brew install livekit-cli # Or curl/winget |
Initialize Project:
For Python:
uv init livekit-voice-agent --bare |
For Node.js:
npm init -y |
Install Dependencies:
Python:
uv add "livekit-agents[silero,turn-detector]~=1.0" "livekit-plugins-noise-cancellation~=0.2" python-dotenv |
Node.js: npm install @livekit/agents @livekit/components-core dotenv
Set Up Environment:
lk app env -w # Generates .env.local with LIVEKIT keys |
Create Basic Agent (agent.py - Python Example):
from dotenv import load_dotenv |
Download Models:
python agent.py download-files # For VAD/turn models |
For Node.js equivalent, see GitHub examples
(github.com/livekit/agents/tree/main/examples/node).
In a voice AI system, multiple components operate together in a real-time pipeline. LiveKit’s AgentSession manages this orchestration by coordinating audio streaming, AI processing, and response generation.
The typical pipeline looks like this:
1. Audio IngressUser speech is streamed through WebRTC or SIP, and noise cancellation is applied to improve audio quality.
2. Detection & TranscriptionThe system detects when the user is speaking using VAD (Voice Activity Detection).The audio is then converted into text using Speech-to-Text (STT).
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3. ReasoningThe LLM processes the transcript, using conversation history and context.If required, it can call tools, APIs, or retrieve knowledge using RAG.
4. Response GenerationThe generated response is converted into speech using Text-to-Speech (TTS).
5. Audio OutputThe audio response is streamed back to the user in real time.
6. Monitoring & Error HandlingObservability tools track transcripts, latency, and failures, while the system performs adaptive retries for network issues.
Recent improvements such as LiveKit Inference help reduce latency by running models closer to the edge. Integrations like the Grok Voice Agent API further enhance voice-native orchestration.
Voice AI agents become more useful when they can interact with external systems, retrieve information, and perform actions.
Common integrations include:
These integrations allow voice agents to move beyond simple conversations and execute real-world workflows.
Code Snippet for Tools (Python):
from livekit.agents import function_tool |
Building a voice AI agent today is far more accessible thanks to modern real-time AI frameworks. With tools like LiveKit Agents, developers can combine speech recognition, LLM reasoning, and text-to-speech into a reliable conversational pipeline.
By starting with a simple prototype and gradually adding integrations such as RAG, tools, and monitoring, teams can move from experimentation to production-ready voice applications.
A voice AI agent is a software system that can understand spoken language, process the request using AI, and respond with natural speech in real time using technologies like STT, LLMs, and TTS.
A typical voice AI system requires Speech-to-Text (STT), Voice Activity Detection (VAD), a Large Language Model (LLM), and Text-to-Speech (TTS) along with real-time infrastructure such as WebRTC.
Most voice AI agents follow a real-time pipeline:
Audio Input → VAD → STT → LLM → TTS → Audio Output
This pipeline enables the system to listen, understand, reason, and respond during a conversation.
Yes. Modern voice AI systems use streaming speech processing and low-latency models to maintain response times typically under 300–500 ms, enabling natural conversations.
Common frameworks and platforms include LiveKit Agents, Vapi, Retell, Pipecat, and custom WebRTC pipelines combined with LLM providers such as OpenAI, Groq, or Anthropic.
Yes. Voice AI agents can connect to APIs, databases, CRMs, and knowledge bases, allowing them to perform actions such as retrieving information, booking appointments, or answering domain-specific questions.
Walk away with actionable insights on AI adoption.
Limited seats available!