Use this skill when building real-time, bidirectional streaming applications with the Gemini Live API. Covers WebSocket-based audio/video/text streaming, voice activity detection (VAD), native audio features, function calling, session management, ephemeral tokens for client-side auth, live translati

Gemini Live API Development Skill Overview The Live API enables low latency, real time voice and video interactions with Gemini over WebSockets. It processes continuous streams of audio, video, or text to deliver immediate, human like spoken responses. Key capabilities: Bidirectional audio streaming — real time mic to speaker conversations Video streaming — send camera/screen frames alongside audio Text input/output — send and receive text within a live session Audio transcriptions — get text transcripts of both input and output audio Voice Activity Detection (VAD) — automatic interruption handling Native audio — thinking (with configurable thinkingLevel ) Function calling — synchronous tool use Google Search grounding — ground responses in real time search results Session management — context compression, session resumption, GoAway signals Ephemeral tokens — secure client side authentication [!NOTE] The Live API currently only supports WebSockets . For WebRTC support or simplified integration, use a [partner integration]( partner integrations). Models gemini 3.1 flash live preview — Optimized for low latency, real time dialogue. Native audio output, thinking (via thinkingLevel ). 128k context window. This is the recommended model for all Live API use cases. gemini 3.5 transcribe live — Real time streaming speech to text with interim hypotheses, finalized transcripts, smart formatting, and Hybrid VAD. gemini 3.5 live translate preview — Real time streaming translation model. [!WARNING] The following Live API models are deprecated and will be shut down. Migrate to gemini 3.1 flash live preview . gemini 2.5 flash native audio preview 12 2025 — Migrate to gemini 3.1 flash live preview . gemini live 2.5 flash preview — Released June 17, 2025. Shutdown: December 9, 2025. gemini 2.0 flash live 001 — Released April 9, 2025. Shutdown: December 9, 2025. SDKs Python : google genai — pip install google genai JavaScript/TypeScript : @google/genai — npm install @google/genai [!WARNING] Legacy SDKs google generativeai (Python) and @google/generative ai (JS) are deprecated. Use the new SDKs above. Partner Integrations To streamline real time audio/video app development, use a third party integration supporting the Gemini Live API over WebRTC or WebSockets : [LiveKit](https://docs.livekit.io/agents/models/realtime/plugins/gemini/) — Use the Gemini Live API with LiveKit Agents. [Pipecat by Daily](https://docs.pipecat.ai/guides/features/gemini live) — Create a real time AI chatbot using Gemini Live and Pipecat. [Fishjam by Software Mansion](https://docs.fishjam.io/tutorials/gemini live integration) — Create live video and audio streaming applications with Fishjam. [Vision Agents by Stream](https://visionagents.ai/integrations/gemini) — Build real time voice and video AI applications with Vision Agents. [Voximplant](https://voximplant.com/products/gemini client) — Connect inbound and outbound calls to Live API with Voximplant. [Firebase AI SDK](https://firebase.google.com/docs/ai logic/live api?api=dev) — Get started with the Gemini Live API using Firebase AI Logic. Audio Formats Input : Raw PCM, little endian, 16 bit, mono. 16kHz native (will resample others). MIME type: audio/pcm;rate=16000 Output : Raw PCM, little endian, 16 bit, mono. 24kHz sample rate. [!IMPORTANT] Use send realtime input / sendRealtimeInput for all real time user input (audio, video, and text ). send client content / sendClientContent is only supported for seeding initial context history (requires setting initial history in client content in history config ). Do not use it to send new user messages during the conversation. [!WARNING] Do not use media in sendRealtimeInput . Use the specific keys: audio for audio data, video for images/video frames, and text for text input. Quick Start Authentication Python JavaScript Connecting to the Live API Python JavaScript Sending Text Python JavaScript Sending Audio Python JavaScript Sending Video Python JavaScript Receiving Audio and Text [!IMPORTANT] A single server event can contain multiple content parts simultaneously (e.g., audio chunks and transcript). Always process all parts in each event to avoid missing content. Python JavaScript Live Translation (Gemini Live Translate) The Live API supports real time, low latency streaming translation of speech (audio) across 70+ languages. For full details on options and capabilities, see the [Live Translate Guide](https://ai.google.dev/gemini api/docs/live api/live translate.md.txt). Model gemini 3.5 live translate preview — The recommended translation model for all Live Translate use cases. Configuration ( TranslationConfig ) To enable translation, specify a TranslationConfig object inside your live session setup: Python SDK : Configure the connection using translation config on LiveConnectConfig : Raw WebSockets : Place translationConfig inside generationConfig : Live Streaming Transcription (Gemini Live Transcribe) The Live API supports real time streaming speech to text over WebSockets with low latency interim hypotheses, finalized transcripts, and Hybrid VAD. For full details, see the [Live Transcription Guide](https://ai.google.dev/gemini api/docs/live api/live transcribe.md.txt) and [Colab Cookbook](https://colab.research.google.com/github/google gemini/cookbook/blob/main/quickstarts/Get started transcribe.ipynb). Model gemini 3.5 transcribe live Modes smart : cleans up filler words, resolves inline self corrections, and structures formatting. verbatim (default): exact word for word transcript. Python JavaScript Raw WebSockets Limitations Response modality — Only TEXT or AUDIO per session, not both. Native audio models only support audio. Audio only session — 15 min without compression Audio+video session — 2 min without compression Connection lifetime — ~10 min (use session resumption) Context window — 128k tokens (native audio) / 32k tokens (standard) Async function calling — Not yet supported; function calling is synchronous only. The model will not start responding until you've sent the tool response. Proactive audio — Not yet supported in Gemini 3.1 Flash Live. Remove any configuration for this feature. Affective dialogue — Not yet supported in Gemini 3.1 Flash Live. Remove any configuration for this feature. Code execution — Not supported URL context — Not supported Migrating from Gemini 2.5 Flash Live When migrating from gemini 2.5 flash native audio preview 12 2025 to gemini 3.1 flash live preview : 1. Model string — Update from gemini 2.5 flash native audio preview 12 2025 to gemini 3.1 flash live preview . 2. Thinking configuration — Use thinkingLevel ( minimal , low , medium , high ) instead of thinkingBudget . Default is minimal for lowest latency. 3. Server events — A single event can contain multiple content parts simultaneously (audio + transcript). Process all parts in each event. 4. Client content — send client content is only for seeding initial context history (set initial history in client content in history config ). Use send realtime input for text during conversation. 5. Turn coverage — Defaults to TURN INCLUDES AUDIO ACTIVITY AND ALL VIDEO instead of TURN INCLUDES ONLY ACTIVITY . If sending constant video frames, consider sending only during audio activity to reduce costs. 6. Async function calling — Not yet supported. Function calling is synchronous only. 7. Proactive audio & affective dialogue — Not yet supported. Remove any configuration for these features. Best Practices 1. Use headphones when testing mic audio to prevent echo/self interruption 2. Enable context window compression for sessions longer than 15 minutes 3. Implement session resumption to handle connection resets gracefully 4. Use ephemeral tokens for client side deployments — never expose API keys in browsers 5. Use send realtime input for all real time user input (audio, video, text). Reserve send client content only for seeding initial context history 6. Send audioStreamEnd when the mic is paused to flush cached audio 7. Clear audio playback queues on interruption signals 8. Process all parts in each server event — events can contain multiple content parts Documentation Lookup When MCP is Installed (Preferred) If the search docs tool (from the Google MCP server) is available, use it as your only documentation source: 1. Call search docs with your query 2. Read the returned documentation 3. Trust MCP results as source of truth for API details — they are always up to date. [!IMPORTANT] When MCP tools are present, never fetch URLs manually. MCP provides up to date, indexed documentation that is more accurate and token efficient than URL fetching. When MCP is NOT Installed (Fallback Only) If no MCP documentation tools are available, fetch from the official docs index: llms.txt URL : https://ai.google.dev/gemini api/docs/llms.txt This index contains links to all documentation pages in .md.txt format. Use web fetch tools to: 1. Fetch llms.txt to discover available documentation pages 2. Fetch specific pages (e.g., https://ai.google.dev/gemini api/docs/live session.md.txt ) Key Documentation Pages [!IMPORTANT] Those are not all the documentation pages. Use the llms.txt index to discover available documentation pages [Live API Overview](https://ai.google.dev/gemini api/docs/live.md.txt) — getting started, raw WebSocket usage [Live Transcription](https://ai.google.dev/gemini api/docs/live api/live transcribe.md.txt) — real time speech to text, interim hypotheses, smart formatting, and Hybrid VAD [Live Translate](https://ai.google.dev/gemini api/docs/live api/live translate.md.txt) — configuration options and capabilities for translation [Live API Capabilities Guide](https://ai.google.dev/gemini api/docs/live guide.md.txt) — voice config, transcription config, native audio (thinking), VAD configuration, media resolution [Live API Tool Use](https://ai.google.dev/gemini api/docs/live tools.md.txt) — function calling (sync and async), Google Search grounding [Session Management](https://ai.google.dev/gemini api/docs/live session.md.txt) — context window compression, session resumption, GoAway signals [Ephemeral Tokens](https://ai.google.dev/gemini api/docs/ephemeral tokens.md.txt) — secure client side authentication for browser/mobile [WebSockets API Reference](https://ai.google.dev/api/live.md.txt) — raw WebSocket protocol details Supported Languages The Live API supports 70 languages including: English, Spanish, French, German, Italian, Portuguese, Chinese, Japanese, Korean, Hindi, Arabic, Russian, and many more. Native audio models automatically detect and switch languages.

gemini-live-api-dev