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.

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.

[!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 — Use the Gemini Live API with LiveKit Agents.
• Pipecat by Daily — Create a real-time AI chatbot using Gemini Live and Pipecat.
• Fishjam by Software Mansion — Create live video and audio streaming applications with Fishjam.
• Vision Agents by Stream — Build real-time voice and video AI applications with Vision Agents.
• Voximplant — Connect inbound and outbound calls to Live API with Voximplant.
• Firebase AI SDK — 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

from google import genai

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

JavaScript

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

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

Connecting to the Live API

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-3.1-flash-live-preview", config=config) as session:
    pass  # Session is active

JavaScript

const session = await ai.live.connect({
  model: 'gemini-3.1-flash-live-preview',
  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

await session.send_realtime_input(text="Hello, how are you?")

JavaScript

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

Sending Audio

Python

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

JavaScript

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

Sending Video

Python

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

JavaScript

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

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

async for response in session.receive():
    content = response.server_content
    if content:
        # Audio — process ALL parts in each event
        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

// 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)
• 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_documentation

tool (from the Google MCP server) is available, use it as your only documentation source:

1. Call

   search_documentation

   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 — getting started, raw WebSocket usage
• Live API Capabilities Guide — voice config, transcription config, native audio (thinking), VAD configuration, media resolution
• Live API Tool Use — function calling (sync and async), Google Search grounding
• Session Management — context window compression, session resumption, GoAway signals
• Ephemeral Tokens — secure client-side authentication for browser/mobile
• WebSockets API Reference — 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.