LiveKit Agents
Open-source real-time WebRTC infrastructure for building ultra-low-latency voice and multimodal AI agents
About LiveKit Agents
LiveKit Agents is an open-source real-time communication framework engineered to build conversational voice, video, and multimodal AI agents with sub-500ms latency. Leveraging WebRTC, it connects speech-to-text (Deepgram, Whisper), LLMs (OpenAI, Anthropic), and text-to-speech (Cartesia, ElevenLabs) in a tightly synchronized bidirectional stream. From customer service avatars to interactive language tutors and hands-free coding copilots, LiveKit Agents provides the enterprise infrastructure for real-time human-AI interaction.
Building real-time voice agents requires solving audio interruption, packet jitter, and latency stacking. LiveKit Agents abstracts these challenges with native Voice Activity Detection (VAD), turn-taking management, and edge-routed audio pipelines. Available in Python and Node.js with client SDKs across React, iOS, Android, and Flutter, LiveKit enables developers to self-host their agent backend or deploy onto LiveKit Cloud.
Install the LiveKit Agents SDK and configure your STT, LLM, and TTS providers.
Define agent event handlers for audio stream ingress, user interruptions, and tool calls.
Connect client frontend via LiveKit React or mobile SDK.
Deploy on local servers or auto-scale with LiveKit Cloud.
Capabilities & Features
Common Use Cases
Building conversational AI voice bots with real-time interruption handling
Deploying interactive multimodal AI vision agents for video calls
Low-latency speech translation and multilingual live audio streaming
Frequently Asked Questions
What is LiveKit Agents?
LiveKit Agents is an open-source real-time WebRTC framework for building conversational voice and multimodal AI assistants with ultra-low latency.
Free Plan
Open-source framework is 100% free to self-host with generous Cloud free tier (50,000 min/mo).
Paid Plan
Usage-based Cloud scaling starting at $0.004/min with enterprise SLAs.
Direct link · Verified & reader-supported
Pros & Cons
Ultra-low latency (<500ms voice response) with adaptive turn-taking and VAD
100% open-source core with comprehensive Python and Node.js SDKs
Native WebRTC transport ensures seamless connection across mobile and web
Requires software engineering expertise in backend streaming pipelines
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