Gemini Live API
GoogleOfficialPublished by Google
- Skill
- Tool integration
- Software development
- Apache-2.0
- Low risk
AutoGineer summary
Written by AutoGineer from the skill’s own files, not by the author
This skill provides a structured approach to generating a client service class for the Gemini Enterprise LiveAPI. It facilitates bidirectional streaming over WebSockets, handling session setup, authentication via Application Default Credentials, and the exchange of ClientMessage and ServerMessage protocol buffers. The skill also generates a demo application to validate the client, including support for audio, video, and text modalities, as well as handling interrupt signals and transcription streams.
What it does
- Generates a client service class for the Gemini LiveAPI in the user's preferred language.
- Implements WebSocket connection management with session setup and resumption logic.
- Handles bearer token authentication using Application Default Credentials.
- Provides async methods for sending realtime data and client content.
- Creates a demo frontend and backend service to test multimodal interactions.
- Includes documentation on how to run and test the generated client and UI.
When to use it
- When building a client that requires bidirectional, low-latency communication with the Gemini LiveAPI.
- When you need to handle continuous audio or video input streams.
- When you need to implement session management, including resumption and token refreshing.
- When you need a reference implementation for processing ServerMessage and ClientMessage protocol buffers.
Advantages
- Automates the scaffolding of complex WebSocket-based client services.
- Ensures adherence to the required LiveAPI wire-level protocol.
- Provides a ready-to-use demo for validating multimodal input and output.
- Encapsulates authentication and session lifecycle management logic.
Usage examples
- “Generate a Python client for the Gemini LiveAPI.”
- The agent creates a Python class with async methods for WebSocket communication, sets up a virtual environment, and provides a demo service to test the connection.
- “How do I handle model interruptions in my LiveAPI client?”
- The agent provides the implementation logic for monitoring the serverContent message for the interrupted flag and clearing stale audio buffers.
Install
Download the skill as a ZIP:
https://autogineer.com/agent-skills/google/gemini-live-api/downloadUnzip it into the cross-client skills folder of your project:
mkdir -p .agents/skills && unzip gemini-live-api.zip -d .agents/skills
Documentation by Google
LiveAPI Service Skill
This skill provides instructions for generating a LiveAPI client service
class that connects to the Gemini Enterprise Live API over WebSockets. The
generated client handles bidirectional streaming, bearer-token authentication
via Application Default Credentials (ADC), transparent session resumption, and
ClientMessage / ServerMessage proto exchange.
The skill also produces a demo frontend + backend service so the user can interactively validate the generated client (text, audio, video, transcription, and interrupt handling).
Attribution
MUST READ: Before executing any gcloud, REST API, or MCP command, you
MUST read Usage Attribution. For commands executed
directly in the terminal (never in generated scripts, Terraform, or user code),
prefix gcloud with
CLOUDSDK_METRICS_REQUEST_ATTRIBUTION="agent-skills/1.0 (skill:gemini-live-api)"
and append agent-skills/1.0 (skill:gemini-live-api) to User-Agent headers.
Prerequisites
Before running the generation flow, ensure the following are available on the host:
A Google Cloud project with the Vertex AI / Gemini Enterprise Agent Platform APIs enabled.
Application Default Credentials configured on the host running the generated client:
gcloud auth application-default loginA destination output folder supplied by the user (e.g.
/tmp/liveapi_out) where the generated code, environment, and demo will be written. Never mutate the host's system Python environment.The user's chosen implementation language (Python is the default and reference language for this skill).
Reference Files
Provided files in references/ (do not treat these as standalone skills —
they are loaded on demand):
client_server_messages.md: Public reference for theClientMessage/ServerMessageschemas used by the Live API.client_server_messages.proto: The proto definition generated fromclient_server_messages.md.session_manager.md: Describes how to correctly handle sessions, buffering, and resumption on disconnection.
Steps
Step 1: Copy the reference files
Copy client_server_messages.md, client_server_messages.proto, and
session_manager.md from this skill's references/ folder into the user's
destination output folder. These files become the source of truth for the
generated client.
Step 2: Reconcile with the public documentation
Examine the public documents linked from client_server_messages.md. If there
are any discrepancies between the public documents and the copied
client_server_messages.md / client_server_messages.proto, update the copies
in the destination folder so the generated client compiles and runs against the
current server contract.
Step 3: Implement the client class
Implement a class in the user's chosen language that:
Imports the local
client_server_messages.prototypes (ClientMessage,ServerMessage).Opens a WebSocket connection to the Live API endpoint.
Exposes async methods so the user can send and receive data to/from the model.
For languages that require an isolated runtime (e.g. Python), create an isolated
environment (e.g. venv) inside the destination folder and generate a bash
script (e.g. setup.sh) that recreates the environment and installs
dependencies. Never install into the system interpreter or the user's global
site-packages, and never instruct the user to run sudo pip install.
Initialization parameters
The user provides the following at construction time:
project_idlocationmodel_idconfig: aClientMessagewith thesetupfield populated.
Authentication
Obtain a bearer token via Application Default Credentials, attach it to the
WebSocket connect request as Authorization: Bearer <token>, refresh the token
before or upon expiry, and reuse the refreshed token on every reconnection
(including go_away and unexpected disconnects). Do not hard-code a
long-lived API key as the only auth mechanism.
Public async API
The class MUST expose the following async methods, gated on receipt of a
setup_complete ServerMessage before sending:
send_realtime_data(data): send realtime input.datais aClientMessagecarrying arealtime_inputfield.send_client_content(data): send non-realtime, turn-based content that contributes to history.datais aClientMessagecarrying aclient_contentfield.receive(): yieldServerMessageinstances parsed from the WebSocket stream.
Do not expose synchronous blocking variants as the primary API surface.
Step 4: Write a test file
Once the client is implemented, generate a test file that initializes the
connection and exercises sending text, audio, and video data and receiving
the responses. Ask the user for any information required to run the test
(project, model, media samples).
Step 5: Generate how_to_run.md
Provide a how_to_run.md in the destination folder that documents the generated
class. Include full examples showing how to build ClientMessage payloads for
every supported modality, how to send them, and how to receive data from the
model.
Step 6: Generate a demo frontend + backend service
Create scripts that deploy the implementation as a service with both a frontend
UI and a backend service (any language). The service MUST reuse the
ClientMessage / ServerMessage protos from Step 1 for wire traffic. Through
the UI the user should be able to:
Start a new connection / close the current connection.
Select the model to use.
Select input sources (audio and/or video from camera or screenshot) and stream them to the model.
Send a text message to the model.
Hear model audio and see the interleaved model and user transcription / conversation history.
While implementing audio and transcription playback, follow the guidance in Live API best practices.
Handling the interrupt signal
When a ServerMessage's server_content arrives with interrupted: true, the
UI MUST:
Ensure played audio and its corresponding transcription remain time-aligned.
Immediately stop the currently playing model audio and stop appending to the in-progress transcription bubble.
Clear the unplayed audio buffer and any pending unrendered transcription so stale content does not bleed into the next turn.
Start new chat bubbles for the next user and model turns.
Handling the transcription finished signal
For streamed input_transcription / output_transcription chunks, append to
the currently active bubble while finished is unset, and close that bubble and
start a fresh one when finished is observed. Route input_transcription text
to user-role bubbles and output_transcription text to model-role bubbles.
Step 7: Generate how_to_test_with_ui.md
Write how_to_test_with_ui.md describing how to launch and use the demo
service. It MUST include:
The exact shell command(s) or script invocation(s) to start the backend service.
The exact shell command(s) or script invocation(s) to start the frontend UI.
The host and port (e.g.
http://localhost:PORT) the user should open in their browser.How to start a session, select a model, choose input sources (mic, camera, screen), send a text message, and observe model audio and transcription in the UI.
Validation Checklist
Before considering the generation complete, verify each item:
[ ]
client_server_messages.md,client_server_messages.proto, andsession_manager.mdwere copied into the destination folder.[ ] The generated client imports the local proto-generated
ClientMessageandServerMessagetypes.[ ] The client connects to the Live API WebSocket at
wss://{location}-aiplatform.googleapis.com/ws/google.cloud.aiplatform.v1beta1.LlmBidiService/BidiGenerateContent(regional endpoint required; Live API models do not support theglobalendpoint), and formats the setupmodelfield asprojects/{project_id}/locations/{location}/publishers/google/models/{model_id}.[ ] Authentication uses ADC-provided bearer tokens sent as
Authorization: Bearer <token>, is refreshed before expiry, and reattached on every reconnect.[ ] Public async methods
send_realtime_data,send_client_content, andreceiveare present, correctly typed, and gated onsetup_complete.[ ] Transparent session resumption is enabled (
session_resumption.transparent = true), the latestnew_handleis tracked, sent-message indexing starts at 1, the buffer is pruned vialast_consumed_client_message_index, and buffered messages are replayed on reconnect (including ongo_awayand WebSocket close codes 1000 / 1006).[ ] If using python, an isolated environment (e.g.
venv) plus asetup.shandrequirements.txt(or equivalent) exist inside the destination folder; no changes were made to system or user-global Python.[ ]
how_to_run.mdandhow_to_test_with_ui.mdare present, and the demo UI reuses the sameClientMessage/ServerMessageprotos.[ ] Interrupt handling and transcription
finishedhandling behave as described above.[ ] The client does not target
generativelanguage.googleapis.comand does not authenticate via API key in a query string.
Files
- ATTRIBUTION.mdAdded by AutoGineer848 B
- LICENSELicence file added by AutoGineer11.1 KiB
- SKILL.md10.1 KiB
- references/attribution.md1.8 KiB
- references/client_server_messages.md29.5 KiB
- references/client_server_messages.proto10.6 KiB
- references/session_manager.md4.1 KiB
references/client_server_messages.md
Live API ClientMessage and ServerMessage Reference
This document describes the wire-level WebSocket protocol for the Gemini
Enterprise Agent Platform Live API (BidiGenerateContent*).
Docs: https://docs.cloud.google.com/gemini-enterprise-agent-platform/reference/models/multimodal-live
The service exposes the RPC BidiGenerateContent. Search the corresponding
website for more information.
1. Connection
Endpoints
| Backend | WebSocket URI |
|---|---|
| Gemini Enterprise Agent Platform | wss://{LOCATION}-aiplatform.googleapis.com/ws/google.cloud.aiplatform.v1beta1.LlmBidiService/BidiGenerateContent |
| Gemini Enterprise Agent Platform (global) | wss://aiplatform.googleapis.com/ws/google.cloud.aiplatform.v1beta1.LlmBidiService/BidiGenerateContent |
Auth
Authorization: Bearer <ADC token>(or API key in Express mode).
Frames
All frames are JSON-serialized protobuf messages of either ClientMessage
(client → server) or ServerMessage (server → client). Each frame is exactly
one WebSocket text message.
2. Top-level structure (oneof)
ClientMessage and ServerMessage are both oneof envelopes — each frame
sets exactly one of the listed fields.
ClientMessage (you → server)
| Field | Type | When to use |
|---|---|---|
setup | BidiGenerateContentSetup | First frame only. Configures the session. |
clientContent | BidiGenerateContentClientContent | Append to conversation history; turn-based input. Interrupts the model. |
realtimeInput | BidiGenerateContentRealtimeInput | Continuous, low-latency audio/video/text input. Not added to history. |
toolResponse | BidiGenerateContentToolResponse | Reply to a server-issued toolCall. |
ServerMessage (server → you)
| Field | Type | Meaning |
|---|---|---|
setupComplete | BidiGenerateContentSetupComplete | Sent once after setup is accepted. Gate further sends on this. |
serverContent | BidiGenerateContentServerContent | Streamed model output (audio/text), turn lifecycle. |
toolCall | BidiGenerateContentToolCall | Model is requesting tool execution. |
toolCallCancellation | BidiGenerateContentToolCallCancellation | Cancels previously-issued tool calls (e.g. on user interruption). |
usageMetadata | UsageMetadata | Token / duration accounting. |
goAway | GoAway | Connection will be terminated soon. |
sessionResumptionUpdate | SessionResumptionUpdate | Resume handle for reconnects. |
Audio transcriptions are delivered inside
serverContent(see § 9), not as separate top-level frames.
3. Lifecycle
Client Server
│ │
├── ClientMessage{ setup } ──────────► │
│ │
│ ◄────── ServerMessage{ setupComplete }
│ │
├── realtimeInput / clientContent ────► │
│ (audio frames, text, etc.) │
│ │
│ ◄────── serverContent (audio chunks, modelTurn parts ...)
│ ◄────── serverContent { generationComplete: true }
│ ◄────── serverContent { turnComplete: true }
│ │
│ ◄────── toolCall { functionCalls[] }
├── toolResponse { functionResponses[] } ► │
│ │
│ ◄────── serverContent ...
│ ◄────── goAway { timeLeft } (eventually)
│ │
│ (close & reconnect using sessionResumptionUpdate.newHandle)Rules:
The first frame must be
setup. Do not send anything else until you receivesetupComplete.Use
clientContentfor turn-based, history-affecting messages (e.g. a typed user message). Sending it interrupts any current model generation.Use
realtimeInputfor continuous audio/video. It does not go into history. Turn boundaries come from VAD (or from explicitactivityStart/activityEndif VAD is disabled).Reply to
toolCallwithtoolResponse(never withclientContent).On
goAway, reconnect using the most recentsessionResumptionUpdate.newHandle.
4. BidiGenerateContentSetup (setup)
Initial-and-only-once configuration for the session.
| Field | Type | Notes |
|---|---|---|
model | string (required) | projects/{p}/locations/{l}/publishers/google/models/{m}. |
generationConfig | GenerationConfig | Unsupported sub-fields here: responseLogprobs, responseMimeType, logprobs, responseSchema, stopSequence, routingConfig, audioTimestamp. |
systemInstruction | Content | Text-only parts. |
tools[] | repeated Tool | Function declarations and built-ins (Search, code execution). |
sessionResumption | SessionResumptionConfig | { handle?: string, transparent?: bool }. Provide handle to resume; omit to start a new resumable session. |
contextWindowCompression | ContextWindowCompressionConfig | { triggerTokens?: int64, slidingWindow?: { targetTokens?: int64 } }. |
realtimeInputConfig | RealtimeInputConfig | See below. |
inputAudioTranscription | AudioTranscriptionConfig | {}. |
outputAudioTranscription | AudioTranscriptionConfig | {}. |
RealtimeInputConfig
| Field | Type | Notes |
|---|---|---|
automaticActivityDetection | AutomaticActivityDetection | Unset → server-side VAD enabled by default. |
activityHandling | enum | START_OF_ACTIVITY_INTERRUPTS (default) | NO_INTERRUPTION. |
turnCoverage | enum | Default TURN_INCLUDES_ALL_INPUT. Also TURN_INCLUDES_ONLY_ACTIVITY, TURN_INCLUDES_AUDIO_ACTIVITY_AND_ALL_VIDEO. |
AutomaticActivityDetection
| Field | Type | Notes |
|---|---|---|
disabled | bool | If true, you must send activityStart / activityEnd yourself. |
startOfSpeechSensitivity | enum | START_SENSITIVITY_HIGH | LOW. |
endOfSpeechSensitivity | enum | END_SENSITIVITY_HIGH | LOW. |
prefixPaddingMs | int32 | Min speech duration to commit start-of-speech. |
silenceDurationMs | int32 | Min silence to commit end-of-speech. |
GenerationConfig — Live-API-relevant subset
Fields most often used in setup.generationConfig:
| Field | Type | Notes |
|---|---|---|
responseModalities[] | repeated enum | TEXT | AUDIO. Pick one per session (mixing not supported). Default AUDIO. |
temperature | float | 0.0–2.0. |
topP, topK, maxOutputTokens | various | Standard sampling / length controls. |
speechConfig | SpeechConfig | Voice / language — only effective when responseModalities=[AUDIO]. See below. |
mediaResolution | enum | MEDIA_RESOLUTION_LOW | MEDIUM | HIGH. Controls token-cost vs. quality of input images / video frames. |
thinkingConfig | ThinkingConfig | { thinkingBudget?: int }. Only on models that support thinking (e.g. Gemini 2.5 Flash). Set thinkingBudget: 0 to disable. |
Unsupported in Live
setup.generationConfig:responseLogprobs,responseMimeType,logprobs,responseSchema,stopSequence,routingConfig,audioTimestamp.
SpeechConfig
Controls the voice the model speaks with. Only meaningful when
responseModalities includes AUDIO.
| Field | Type | Notes |
|---|---|---|
voiceConfig | VoiceConfig | Single-speaker voice. The Live API supports only single-speaker output. |
languageCode | string | BCP-47 (e.g. en-US, de-DE, ja-JP). Output language. Native-audio models auto-detect / switch on their own; for non-native-audio models, set this explicitly. See Languages supported. |
Note on multi-speaker output: A Live API session has exactly one
prebuiltVoiceConfig.voiceName. To approximate multiple speakers in a Live session, use prompt engineering insidesystemInstructionorclientContentto have the single voice play different roles.
VoiceConfig
| Field | Type | Notes |
|---|---|---|
prebuiltVoiceConfig | PrebuiltVoiceConfig | Selects a named built-in voice. |
PrebuiltVoiceConfig
| Field | Type | Notes |
|---|---|---|
voiceName | string | Name of the prebuilt voice. 30 voices supported. See Voices supported below for the full list and references. |
Voices supported
The Live API supports 30 prebuilt voices. Names are case-sensitive.
| Voice | Style | Voice | Style | Voice | Style |
|---|---|---|---|---|---|
| Zephyr | Bright | Puck | Upbeat | Charon | Informative |
| Kore | Firm | Fenrir | Excitable | Leda | Youthful |
| Orus | Firm | Aoede | Breezy | Callirrhoe | Easy-going |
| Autonoe | Bright | Enceladus | Breathy | Iapetus | Clear |
| Umbriel | Easy-going | Algieba | Smooth | Despina | Smooth |
| Erinome | Clear | Algenib | Gravelly | Rasalgethi | Informative |
| Laomedeia | Upbeat | Achernar | Soft | Alnilam | Firm |
| Schedar | Even | Gacrux | Mature | Pulcherrima | Forward |
| Achird | Friendly | Zubenelgenubi | Casual | Vindemiatrix | Gentle |
| Sadachbia | Lively | Sadaltager | Knowledgeable | Sulafat | Warm |
Reference (authoritative voice list):
The available set may vary per model (e.g. native-audio vs. half-cascade models). If a voice is rejected during
setup, the WebSocket closes instead of returningsetupComplete.
Languages supported
The Live API supports 24 BCP-47 languages for speechConfig.languageCode:
ar-EG, bn-BD, de-DE, en-IN (bundled with hi-IN), en-US, es-US,
fr-FR, hi-IN, id-ID, it-IT, ja-JP, ko-KR, mr-IN, nl-NL,
pl-PL, pt-BR, ro-RO, ru-RU, ta-IN, te-IN, th-TH, tr-TR,
uk-UA, vi-VN.
Native-audio models (e.g.
gemini-live-2.5-flash-native-audio) can switch languages mid-conversation; for those,languageCodeis optional and the model auto-detects. For non-native-audio models, set it explicitly.
Examples
Basic prebuilt voice:
{
"setup": {
"model": "...",
"generationConfig": {
"responseModalities": ["AUDIO"],
"speechConfig": {
"voiceConfig": {
"prebuiltVoiceConfig": { "voiceName": "Aoede" }
}
}
}
}
}Voice + output language pinned to German:
{
"setup": {
"model": "...",
"generationConfig": {
"responseModalities": ["AUDIO"],
"speechConfig": {
"voiceConfig": { "prebuiltVoiceConfig": { "voiceName": "Charon" } },
"languageCode": "de-DE"
}
}
}
}Notes -
speechConfigis silently ignored ifresponseModalitiesis["TEXT"]. -voiceNameis case-sensitive and model-specific. Sending an unknown name typically fails thesetup(you'll see the WebSocket close instead ofsetupComplete). -speechConfig.languageCodecontrols output TTS language. - The Live API is single-speaker only.
5. BidiGenerateContentClientContent (clientContent)
Turn-based input. Append to history and (optionally) trigger generation.
| Field | Type | Notes |
|---|---|---|
turns[] | repeated Content | Conversation history + the latest user request. |
turnComplete | bool | If true, server starts generating immediately. |
Sending
clientContentwhile the model is speaking interrupts it. Do not useclientContentto deliverFunctionResponses — usetoolResponse.
6. BidiGenerateContentRealtimeInput (realtimeInput)
Continuous, low-latency input. Does not populate history. End-of-turn is derived from VAD (or activity events).
| Field | Type | Notes |
|---|---|---|
audio | Blob | Typed realtime audio. PCM |
| : : : 16-bit, 16 kHz mono input : | ||
: : : (audio/pcm;rate=16000). : | ||
video | Blob | Typed realtime video frame |
: : : (image/jpeg/png/webp), : | ||
| : : : should be sent at 1 fps. : | ||
text | string | Realtime text input. |
mediaChunks[] | repeated Blob | Deprecated. Combined |
| : : : audio/video chunks. Prefer : | ||
: : : the typed audio / video : | ||
: : : / text fields instead. : | ||
audioStreamEnd | bool | Mic turned off. Only valid |
| : : : with auto-VAD enabled. : | ||
activityStart | ActivityStart (empty) | Only when auto-VAD is |
| : : : disabled. : | ||
activityEnd | ActivityEnd (empty) | Only when auto-VAD is |
| : : : disabled. : |
Audio formats
Input:
audio/pcm;rate=16000(16 kHz, 16-bit signed PCM, mono, little-endian).Output: 24 kHz, 16-bit signed PCM, mono.
7. BidiGenerateContentToolResponse (toolResponse)
| Field | Type | Notes |
|---|---|---|
functionResponses[] | repeated | Match FunctionCall.id from |
: : FunctionResponse : the server. id is optional. : |
8. BidiGenerateContentSetupComplete (setupComplete)
| Field | Type | Notes |
|---|---|---|
sessionId | string | Server-assigned session identifier. |
Receipt of this message is the gate for sending any other client frame.
9. BidiGenerateContentServerContent (serverContent)
Primary streaming-output channel.
| Field | Type | Notes |
|---|---|---|
modelTurn | Content | Streamed model output parts (text and/or inlineData audio). |
generationComplete | bool | Model has finished generating; playback may still be flushing. |
turnComplete | bool | Logical end of turn. |
interrupted | bool | Generation was interrupted by client input — drop any queued audio playback. |
groundingMetadata | GroundingMetadata | When grounding (e.g. Google Search) is used. |
inputTranscription | Transcription | { text?: string } — transcription of the user's spoken input. Requires inputAudioTranscription set in setup. |
outputTranscription | Transcription | { text?: string } — transcription of the model's spoken output. Requires outputAudioTranscription set in setup. |
Playback note: audio chunks arrive inside modelTurn.parts[].inlineData
with mimeType audio/pcm;rate=24000. Concatenate as they stream. On
interrupted: true, flush the playback queue.
10. BidiGenerateContentToolCall (toolCall)
| Field | Type | Notes |
|---|---|---|
functionCalls[] | repeated FunctionCall | Each has id, name, args. Reply with a matching FunctionResponse.id. |
11. BidiGenerateContentToolCallCancellation (toolCallCancellation)
| Field | Type | Notes |
|---|---|---|
ids[] | repeated string | IDs of previously-issued tool calls to cancel. Typically caused by user interruption. Stop the work; do not send toolResponse for these. |
12. GoAway, SessionResumptionUpdate, UsageMetadata
GoAway
| Field | Type | Notes |
|---|---|---|
timeLeft | Duration (string) | Time until the connection terminates with ABORTED. Reconnect using the latest SessionResumptionUpdate.newHandle. |
SessionResumptionUpdate
| Field | Type | Notes |
|---|---|---|
newHandle | string | Handle to resume; empty if resumable=false. |
resumable | bool | Whether the session is currently resumable. |
lastConsumedClientMessageIndex | int64 | Set only when SessionResumptionConfig.transparent=true — enables transparent reconnect. |
UsageMetadata
| Field | Type |
|---|---|
totalTokenCount | int32 |
textCount | int32 |
imageCount | int32 |
videoDurationSeconds | int32 |
audioDurationSeconds | int32 |
13. End-to-end JSON examples
Setup
{
"setup": {
"model": "projects/my-proj/locations/us-central1/publishers/google/models/gemini-3.8-live",
"generationConfig": {
"responseModalities": ["AUDIO"],
"speechConfig": {
"voiceConfig": { "prebuiltVoiceConfig": { "voiceName": "Aoede" } }
}
},
"systemInstruction": {
"parts": [{ "text": "You are a concise voice assistant." }]
},
"realtimeInputConfig": {
"automaticActivityDetection": { "disabled": false }
},
"sessionResumption": {},
"outputAudioTranscription": {}
}
}Setup complete (server)
{ "setupComplete": { "sessionId": "abc-123" } }Streaming user audio (realtime)
{
"realtimeInput": {
"audio": {
"mimeType": "audio/pcm;rate=16000",
"data": "<base64-pcm-bytes>"
}
}
}Typed user message (turn-based)
{
"clientContent": {
"turns": [{ "role": "user", "parts": [{ "text": "Hello!" }] }],
"turnComplete": true
}
}Streaming model output (server)
{
"serverContent": {
"modelTurn": {
"role": "model",
"parts": [{
"inlineData": {
"mimeType": "audio/pcm;rate=24000",
"data": "<base64-pcm-bytes>"
}
}]
}
}
}Tool call / response
// Server → client
{
"toolCall": {
"functionCalls": [
{ "id": "call_42", "name": "get_weather", "args": { "city": "Paris" } }
]
}
}
// Client → server
{
"toolResponse": {
"functionResponses": [
{
"id": "call_42",
"name": "get_weather",
"response": { "tempC": 18, "summary": "Partly cloudy" }
}
]
}
}GoAway + resumption
{ "goAway": { "timeLeft": "10s" } }
{ "sessionResumptionUpdate": { "newHandle": "ses_xyz", "resumable": true } }Reconnect with:
{ "setup": { "model": "...", "sessionResumption": { "handle": "ses_xyz" } } }14. Full input examples — audio, video, text
There are two distinct ways to send user input. Pick one based on intent:
| | Part 1 — Realtime input | Part 2 — Add-context |
: : (realtimeInput) : input (clientContent) :
| --------------------- | ------------------------ | ------------------------- |
| Goal | Stream live mic / camera | Append a discrete turn to |
: : continuously : the conversation :
| Latency | Lowest possible; | Normal request/response |
: : sub-second : :
| Added to history? | No (transient | Yes (persistent |
: : signals) : conversation) :
| Turn boundary | VAD (auto), or explicit | Explicit turnComplete: | : : activityStart/End : true :
| Effect on model | Streamed; auto-triggers | Setting turnComplete |
: : a turn when VAD fires : triggers generation; :
: : : interrupts any :
: : : ongoing model output :
| Field carriers | audio / video / | turns[].parts[].text / |
: : text : inlineData / fileData :
| Typical use | Live voice + screen | Typed chat, uploading an |
: : sharing, push-to-talk : image/clip, replaying :
: : : history on resume :
You may use both in the same session — e.g. send a
clientContentsystem nudge once, then continue streamingrealtimeInput. But within a single logical user turn, pick one.
The two parts below give every supported variant for audio, video, and text in each mode.
Part 1 — Realtime input (realtimeInput)
Continuous, low-latency input that does not populate conversation
history. End-of-turn comes from server-side VAD by default, or from explicit
activityStart / activityEnd events when auto-VAD is disabled in setup.
1.A Realtime audio
Required format: PCM, 16-bit signed, 16 kHz, mono, little-endian, base64
encoded. MIME type: audio/pcm;rate=16000. Send one frame per chunk
(~20–100 ms is typical).
1.A.1 Continuous mic
{
"realtimeInput": {
"audio": {
"mimeType": "audio/pcm;rate=16000",
"data": "<base64 PCM chunk>"
}
}
}When the mic turns off (and server-side VAD is enabled), commit end-of-stream:
{ "realtimeInput": { "audioStreamEnd": true } }1.A.2 Manual VAD (auto-VAD disabled)
Required setup:
{
"setup": {
"model": "...",
"realtimeInputConfig": { "automaticActivityDetection": { "disabled": true } }
}
}Then explicitly frame each utterance:
{ "realtimeInput": { "activityStart": {} } }
{ "realtimeInput": { "audio": { "mimeType": "audio/pcm;rate=16000", "data": "..." } } }
{ "realtimeInput": { "audio": { "mimeType": "audio/pcm;rate=16000", "data": "..." } } }
{ "realtimeInput": { "activityEnd": {} } }1.B Realtime video
Video = a stream of sampled image frames (should be sent at 1 fps). Each frame is JPEG / PNG / WebP. The model does not consume an encoded container (mp4/webm) — sample frames client-side and send each as an inline image.
1.B.1 Continuous camera
{
"realtimeInput": {
"video": {
"mimeType": "image/jpeg",
"data": "<base64 JPEG frame>"
}
}
}1.C Realtime text
{ "realtimeInput": { "text": "Switch to a calmer tone." } }1.D Combined realtime audio + video (+ text)
Typical "talk-to-the-screen" flow. Send each modality in its own frame:
{ "realtimeInput": { "video": { "mimeType": "image/jpeg", "data": "<frame_t0>" } } }
{ "realtimeInput": { "audio": { "mimeType": "audio/pcm;rate=16000", "data": "<pcm_t0>" } } }
{ "realtimeInput": { "video": { "mimeType": "image/jpeg", "data": "<frame_t1>" } } }
{ "realtimeInput": { "audio": { "mimeType": "audio/pcm;rate=16000", "data": "<pcm_t1>" } } }
{ "realtimeInput": { "text": "Focus on the chart in the upper-right." } }
{ "realtimeInput": { "audio": { "mimeType": "audio/pcm;rate=16000", "data": "<pcm_t2>" } } }
{ "realtimeInput": { "audioStreamEnd": true } }1.E Realtime quick reference
| Modality | Field |
|---|---|
| Audio chunk | realtimeInput.audio (audio/pcm;rate=16000) |
| Video frame | realtimeInput.video (image/jpeg/png/webp) |
| Text | realtimeInput.text |
| Combined audio+video | realtimeInput.mediaChunks[] (deprecated — |
: : prefer the typed audio / video fields) : | |
| Mic-off signal | realtimeInput.audioStreamEnd: true |
| Manual turn boundary | realtimeInput.activityStart / activityEnd |
| : : (auto-VAD disabled) : |
Part 2 — Add-context input (clientContent)
Discrete, history-bearing turns. Use this when you want the message to be
permanently part of the conversation context the model sees on subsequent
turns. Setting turnComplete: true triggers generation immediately and
interrupts any ongoing model output.
Do not use
clientContentto reply to atoolCall— usetoolResponse.
2.A Add-context text
2.A.1 Single user turn
{
"clientContent": {
"turns": [
{ "role": "user", "parts": [{ "text": "What's the capital of France?" }] }
],
"turnComplete": true
}
}2.A.2 Multi-turn history (e.g. on resume)
{
"clientContent": {
"turns": [
{ "role": "user", "parts": [{ "text": "Hi, my name is Sam." }] },
{ "role": "model", "parts": [{ "text": "Nice to meet you, Sam!" }] },
{ "role": "user", "parts": [{ "text": "What's my name?" }] }
],
"turnComplete": true
}
}2.A.3 Streamed turn (don't generate yet — more parts coming)
{ "clientContent": { "turns": [{ "role": "user", "parts": [{ "text": "Once upon" }] }] } }
{ "clientContent": { "turns": [{ "role": "user", "parts": [{ "text": " a time..." }] }], "turnComplete": true } }2.B Add-context audio
A pre-recorded clip delivered as a discrete history-bearing turn.
2.B.1 Audio clip alone
{
"clientContent": {
"turns": [{
"role": "user",
"parts": [{
"inlineData": {
"mimeType": "audio/pcm;rate=16000",
"data": "<base64 of full clip>"
}
}]
}],
"turnComplete": true
}
}2.B.2 Audio clip with accompanying text prompt
{
"clientContent": {
"turns": [{
"role": "user",
"parts": [
{ "text": "Transcribe this and summarize:" },
{ "inlineData": { "mimeType": "audio/pcm;rate=16000", "data": "..." } }
]
}],
"turnComplete": true
}
}2.C Add-context video / image
2.C.1 Single image
{
"clientContent": {
"turns": [{
"role": "user",
"parts": [
{ "text": "What's in this picture?" },
{ "inlineData": { "mimeType": "image/jpeg", "data": "<base64 image>" } }
]
}],
"turnComplete": true
}
}2.C.2 Multiple frames as a single turn (sampled clip)
{
"clientContent": {
"turns": [{
"role": "user",
"parts": [
{ "text": "Here are 3 frames from a video. What's happening?" },
{ "inlineData": { "mimeType": "image/jpeg", "data": "<frame_0>" } },
{ "inlineData": { "mimeType": "image/jpeg", "data": "<frame_1>" } },
{ "inlineData": { "mimeType": "image/jpeg", "data": "<frame_2>" } }
]
}],
"turnComplete": true
}
}2.C.3 Image by URI (fileData)
{
"clientContent": {
"turns": [{
"role": "user",
"parts": [
{ "text": "Describe this image." },
{ "fileData": { "mimeType": "image/jpeg", "fileUri": "gs://my-bucket/cat.jpg" } }
]
}],
"turnComplete": true
}
}2.D Combined add-context multimodal turn
Text + image + audio in a single user turn:
{
"clientContent": {
"turns": [{
"role": "user",
"parts": [
{ "text": "Compare what I'm saying with what I'm showing:" },
{ "inlineData": { "mimeType": "image/jpeg", "data": "<image>" } },
{ "inlineData": { "mimeType": "audio/pcm;rate=16000", "data": "<clip>" } }
]
}],
"turnComplete": true
}
}2.E Add-context quick reference
| Modality | Field | Notes |
|---|---|---|
| Text | turns[].parts[].text | One or more text |
| : : : parts per turn. : | ||
| Inline audio | turns[].parts[].inlineData | Full clip, base64. |
: : (audio/pcm;rate=16000) : : | ||
| Inline image / video | turns[].parts[].inlineData | Multiple parts allowed |
: frame : (image/jpeg/png/webp) : for sampled clips. : | ||
| Remote file | turns[].parts[].fileData | GCS URI |
: : (fileUri) : : | ||
| Trigger generation | turnComplete: true | Omit to keep streaming |
| : : : more parts. : | ||
| History order | turns[] ordered oldest → | role is "user" or |
: : newest : "model". : |
15. Common pitfalls
Sending data before
setupComplete— server will close the connection.Mixing
clientContentandrealtimeInputfor the same logical turn — pick one mode.clientContentinterrupts ongoing generation.Ignoring
interrupted: true— leftover queued audio will play over the user's next utterance.Replying to
toolCallwithclientContent— must betoolResponse.Forgetting
FunctionResponse.id— request rejected.Wrong audio format — input must be 16 kHz PCM, output is 24 kHz PCM.
No reconnect handling — sessions have a max duration; always honor
goAwayand persist the latestSessionResumptionUpdate.newHandle.
Requirements and compatibility
Works with: Claude Code, Claude apps, OpenAI Codex, GitHub Copilot, Cursor, Gemini CLI, Kiro, OpenCode, Goose, Amp, Roo Code, JetBrains Junie, Factory Droid, Devin (ex-Windsurf), Mistral Vibe, OpenHands, Letta
“Works with” means the tool reads the open Agent Skills format; a skill written for one tool may use features others lack.
Needs: Needs an API key, Needs a command-line tool, Needs Python
Security
What our scanner noticed· Low risk
- Instructions that use the network
A static scan is not a guarantee. Skills are third-party content: review a skill before you let an agent use it.
Attribution and licence
- Published by
- Publisher
- Google (Official)
- Source
- google/skills · skills/cloud/gemini-live-api · commit 2194d3b
- Licence
- Apache License 2.0 (redistribution allowed)
- LICENSE (added by AutoGineer from LICENSE)
- ATTRIBUTION.md (written by AutoGineer)
- Synced
AutoGineer is not affiliated with or endorsed by Google or Google.
Download ZIP
The ZIP holds the skill’s files, its licence files and an ATTRIBUTION.md. Download gemini-live-api.zip