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gemini-live-api-dev

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, and all Live API configuration options. SDKs covered - google-genai (Python), @google/genai (JavaScript/TypeScript).

Packaged view

This page reorganizes the original catalog entry around fit, installability, and workflow context first. The original raw source lives below.

Stars
2,270
Hot score
99
Updated
March 20, 2026
Overall rating
C4.0
Composite score
4.0
Best-practice grade
A85.2

Install command

npx @skill-hub/cli install google-gemini-gemini-skills-gemini-live-api-dev

Repository

google-gemini/gemini-skills

Skill path: skills/gemini-live-api-dev

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, and all Live API configuration options. SDKs covered - google-genai (Python), @google/genai (JavaScript/TypeScript).

Open repository

Best for

Primary workflow: Run DevOps.

Technical facets: Full Stack, Backend, Security.

Target audience: everyone.

License: Unknown.

Original source

Catalog source: SkillHub Club.

Repository owner: google-gemini.

This is still a mirrored public skill entry. Review the repository before installing into production workflows.

What it helps with

  • Install gemini-live-api-dev into Claude Code, Codex CLI, Gemini CLI, or OpenCode workflows
  • Review https://github.com/google-gemini/gemini-skills before adding gemini-live-api-dev to shared team environments
  • Use gemini-live-api-dev for development workflows

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Claude CodeCodex CLIGemini CLIOpenCode

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Original source / Raw SKILL.md

---
name: gemini-live-api-dev
description: 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, and all Live API configuration options. SDKs covered - google-genai (Python), @google/genai (JavaScript/TypeScript).
---

# 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** — affective dialog, proactive audio, thinking
- **Function calling** — synchronous and asynchronous 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-2.5-flash-native-audio-preview-12-2025` — Native audio output, affective dialog, proactive audio, thinking. 128k context window. **This is the recommended model for all Live API use cases.**

> [!WARNING]
> The following Live API models are **deprecated** and will be shut down. Migrate to `gemini-2.5-flash-native-audio-preview-12-2025`.
> - `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**). Use `send_client_content` / `sendClientContent` **only** for incremental conversation history updates (appending prior turns to context), not for sending new user messages.

> [!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

```python
from google import genai

client = genai.Client(api_key="YOUR_API_KEY")
```

#### JavaScript

```js
import { GoogleGenAI } from '@google/genai';

const ai = new GoogleGenAI({ apiKey: 'YOUR_API_KEY' });
```

### Connecting to the Live API

#### Python
```python
from google.genai import types

config = types.LiveConnectConfig(
    response_modalities=[types.Modality.AUDIO],
    system_instruction=types.Content(
        parts=[types.Part(text="You are a helpful assistant.")]
    )
)

async with client.aio.live.connect(model="gemini-2.5-flash-native-audio-preview-12-2025", config=config) as session:
    pass  # Session is now active
```

#### JavaScript
```js
const session = await ai.live.connect({
  model: 'gemini-2.5-flash-native-audio-preview-12-2025',
  config: {
    responseModalities: ['audio'],
    systemInstruction: { parts: [{ text: 'You are a helpful assistant.' }] }
  },
  callbacks: {
    onopen: () => console.log('Connected'),
    onmessage: (response) => console.log('Message:', response),
    onerror: (error) => console.error('Error:', error),
    onclose: () => console.log('Closed')
  }
});
```

### Sending Text

#### Python
```python
await session.send_realtime_input(text="Hello, how are you?")
```

#### JavaScript
```js
session.sendRealtimeInput({ text: 'Hello, how are you?' });
```

### Sending Audio

#### Python
```python
await session.send_realtime_input(
    audio=types.Blob(data=chunk, mime_type="audio/pcm;rate=16000")
)
```

#### JavaScript
```js
session.sendRealtimeInput({
  audio: { data: chunk.toString('base64'), mimeType: 'audio/pcm;rate=16000' }
});
```

### Sending Video

#### Python
```python
# frame: raw JPEG-encoded bytes
await session.send_realtime_input(
    video=types.Blob(data=frame, mime_type="image/jpeg")
)
```

#### JavaScript
```js
session.sendRealtimeInput({
  video: { data: frame.toString('base64'), mimeType: 'image/jpeg' }
});
```

### Receiving Audio and Text

#### Python
```python
async for response in session.receive():
    content = response.server_content
    if content:
        # Audio
        if content.model_turn:
            for part in content.model_turn.parts:
                if part.inline_data:
                    audio_data = part.inline_data.data
        # Transcription
        if content.input_transcription:
            print(f"User: {content.input_transcription.text}")
        if content.output_transcription:
            print(f"Gemini: {content.output_transcription.text}")
        # Interruption
        if content.interrupted is True:
            pass  # Stop playback, clear audio queue
```

#### JavaScript
```js
// Inside the onmessage callback
const content = response.serverContent;
if (content?.modelTurn?.parts) {
  for (const part of content.modelTurn.parts) {
    if (part.inlineData) {
      const audioData = part.inlineData.data; // Base64 encoded
    }
  }
}
if (content?.inputTranscription) console.log('User:', content.inputTranscription.text);
if (content?.outputTranscription) console.log('Gemini:', content.outputTranscription.text);
if (content?.interrupted) { /* Stop playback, clear audio queue */ }
```

---

## Limitations

- **Response modality** — Only `TEXT` **or** `AUDIO` per session, not both
- **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)
- **Code execution** — Not supported
- **URL context** — Not supported

## 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 injecting conversation history
6. **Send `audioStreamEnd`** when the mic is paused to flush cached audio
7. **Clear audio playback queues** on interruption signals

## How to use the Gemini API

For detailed API documentation, 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 API Capabilities Guide](https://ai.google.dev/gemini-api/docs/live-guide.md.txt) — voice config, transcription config, native audio (affective dialog, proactive 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 | SkillHub