ElevenLabs JavaScript Client Library
npm install @elevenlabs/client!hero
Build multimodal agents with the ElevenLabs Agents platform.
An SDK library for using ElevenLabs Agents. If you're looking for a Node.js library for other audio APIs, please refer to the ElevenLabs Node.js Library.
!LOGO
Install the package in your project through package manager.
``shell`
npm install @elevenlabs/clientor
yarn add @elevenlabs/clientor
pnpm install @elevenlabs/client
This library is primarily meant for development in vanilla JavaScript projects, or as a base for libraries tailored to specific frameworks.
It is recommended to check whether your specific framework has it's own library.
However, you can use this library in any JavaScript-based project.
A conversation can be started via one of two connection types: WebSockets (the default) or WebRTC.
First, initialize the Conversation instance:
`js`
const conversation = await Conversation.startSession(options);
This will kick off the websocket connection and start using microphone to communicate with the ElevenLabs agent. Consider explaining and allowing microphone access in your apps UI before the Conversation kicks off. The microphone may also be blocked for the current page by default, resulting in the allow prompt not showing up at all. You should handle such use case in your application and display appropriate message to the user:
`js`
// call after explaning to the user why the microphone access is needed
// handle errors and show appropriate message to the user
try {
await navigator.mediaDevices.getUserMedia();
} catch {
// handle error
}
#### Session configuration
The options passed to startSession specifiy how the session is established. There are three ways to start a session:
##### Public agents
Agents that don't require any authentication can be used to start a conversation by using the agent ID and the connection type. The agent ID can be acquired through the ElevenLabs UI.
For public agents, you can use the ID directly:
`js`
const conversation = await Conversation.startSession({
agentId: "
connectionType: "webrtc", // 'websocket' is also accepted
});
##### Private agents
If the conversation requires authorization, you will need to add a dedicated endpoint to your server that will either request a signed url (if using the WebSockets connection type) or a conversation token (if using WebRTC) using the ElevenLabs API and pass it back to the client.
Here's an example for a WebSocket connection:
`js
// Node.js server
app.get("/signed-url", yourAuthMiddleware, async (req, res) => {
const response = await fetch(
https://api.elevenlabs.io/v1/convai/conversation/get-signed-url?agent_id=${process.env.AGENT_ID},
{
headers: {
// Requesting a signed url requires your ElevenLabs API key
// Do NOT expose your API key to the client!
"xi-api-key": process.env.ELEVENLABS_API_KEY,
},
}
);
if (!response.ok) {
return res.status(500).send("Failed to get signed URL");
}
const body = await response.json();
res.send(body.signed_url);
});
`
`js
// Client
const response = await fetch("/signed-url", yourAuthHeaders);
const signedUrl = await response.text();
const conversation = await Conversation.startSession({
signedUrl,
connectionType: "websocket",
});
`
Here's an example for WebRTC:
`js
// Node.js server
app.get("/conversation-token", yourAuthMiddleware, async (req, res) => {
const response = await fetch(
https://api.elevenlabs.io/v1/convai/conversation/token?agent_id=${process.env.AGENT_ID},
{
headers: {
// Requesting a conversation token requires your ElevenLabs API key
// Do NOT expose your API key to the client!
'xi-api-key': process.env.ELEVENLABS_API_KEY,
}
}
);
if (!response.ok) {
return res.status(500).send("Failed to get conversation token");
}
const body = await response.json();
res.send(body.token);
);
`
Once you have the token, providing it to startSession will initiate the conversation using WebRTC.
`js
// Client
const response = await fetch("/conversation-token", yourAuthHeaders);
const conversationToken = await response.text();
const conversation = await Conversation.startSession({
conversationToken,
connectionType: "webrtc",
});
`
#### Optional callbacks
The options passed to startSession can also be used to register optional callbacks:
- onConnect - handler called when the conversation websocket connection is established.
- onDisconnect - handler called when the conversation websocket connection is ended.
- onMessage - handler called when a new text message is received. These can be tentative or final transcriptions of user voice, replies produced by LLM. Primarily used for handling conversation transcription.
- onError - handler called when an error is encountered.
- onStatusChange - handler called whenever connection status changes. Can be connected, connecting and disconnected (initial).speaking
- onModeChange - handler called when a status changes, eg. agent switches from to listening, or the other way around.
- onCanSendFeedbackChange - handler called when sending feedback becomes available or unavailable.
- onUnhandledClientToolCall - handler called when a client tool is invoked but no corresponding client tool was defined
- onDebug - handler called for debugging events, including tentative agent responses and internal events. Useful for development and troubleshooting.
- onAudio - handler called when audio data is received from the agent. Provides access to raw audio events for custom processing.
- onInterruption - handler called when the conversation is interrupted, typically when the user starts speaking while the agent is talking.
- onVadScore - handler called with voice activity detection scores, indicating the likelihood of speech in the audio input.
- onMCPToolCall - handler called when an MCP (Model Context Protocol) tool is invoked by the agent.
- onMCPConnectionStatus - handler called when the MCP connection status changes, useful for monitoring MCP server connectivity.
- onAgentToolRequest - handler called when the agent begins tool execution.
- onAgentToolResponse - handler called when the agent receives a response from a tool execution.
- onConversationMetadata - handler called with conversation initiation metadata, providing information about the conversation setup.
- onAsrInitiationMetadata - handler called with ASR (Automatic Speech Recognition) initiation metadata, containing configuration details for speech recognition.
- onAgentChatResponsePart - handler called with streaming text chunks during text-only conversations. Provides start, delta, and stop events for real-time text streaming.
- onAudioAlignment - handler called with character-level timing data for synthesized audio. Provides arrays of characters, start times, and durations for text-to-speech synchronization.
#### Setting input/output devices
You can provide a device ID to start the conversation using the input/output device of your choice. If the device ID is invalid, the default input and output devices will be used.
`js`
const conversation = await Conversation.startSession({
agentId: "
inputDeviceId: "
outputDeviceId: "
});
Note: Device switching only works for voice conversations. You can enumerate available devices using the MediaDevices.enumerateDevices() API.
#### Client Tools
Client tools are a way to enabled agent to invoke client-side functionality. This can be used to trigger actions in the client, such as opening a modal or doing an API call on behalf of the user.
Client tools definition is an object of functions, and needs to be identical with your configuration within the ElevenLabs UI, where you can name and describe different tools, as well as set up the parameters passed by the agent.
`ts
const conversation = await Conversation.startSession({
clientTools: {
displayMessage: async (parameters: { text: string }) => {
alert(text);
return "Message displayed";
},
},
});
`
In case function returns a value, it will be passed back to the agent as a response.
Note that the tool needs to be explicitly set to be blocking conversation in ElevenLabs UI for the agent to await and react to the response, otherwise agent assumes success and continues the conversation.
#### Conversation overrides
You may choose to override various settings of the conversation and set them dynamically based other user interactions.
We support overriding various settings.
These settings are optional and can be used to customize the conversation experience.
The following settings are available:
`ts`
const conversation = await Conversation.startSession({
overrides: {
agent: {
prompt: {
prompt: "My custom prompt",
llm: "gemini-2.5-flash",
},
firstMessage: "My custom first message",
language: "en",
},
tts: {
voiceId: "custom voice id",
speed: 1.0,
stability: 0.5,
similarityBoost: 0.8,
},
conversation: {
textOnly: true,
},
},
});
#### User identification
You can optionally pass a user ID to identify the user in the conversation. This can be your own customer identifier. This will be included in the conversation initiation data sent to the server:
Tracking this ID can be helpful for filtering conversations, tracking analytics on a user level, etc.
`ts`
const conversation = await Conversation.startSession({
agentId: "
userId: "user-123", // Optional user identifier
connectionType: "webrtc",
});
#### Text only
If your agent is configured to run in text-only mode, i.e. it does not send or receive audio messages,
you can use this flag to use a lighter version of the conversation. In that case, the
user will not be asked for microphone permissions and no audio context will be created.
`ts`
const conversation = await Conversation.startSession({
textOnly: true,
});
#### Prefer Headphones for iOS Devices
While this SDK leaves the choice of audio input/output device to the browser/system, iOS Safari seem to prefer the built-in speaker over headphones even when bluetooth device is in use. If you want to "force" the use of headphones on iOS devices when available, you can use the following option. Please, keep in mind that this is not guaranteed, since this functionality is not provided by the browser. System audio should be the default choice.
`ts`
const conversation = await Conversation.startSession({
preferHeadphonesForIosDevices: true,
});
#### Connection delay
You can configure additional delay between when the microphone is activated and when the connection is established.
On Android, the delay is set to 3 seconds by default to make sure the device has time to switch to the correct audio mode.
Without it, you may experience issues with the beginning of the first message being cut off.
`ts`
const conversation = await Conversation.startSession({
connectionDelay: {
android: 3_000,
ios: 0,
default: 0,
},
});
#### Acquiring a Wake Lock
By default, the conversation will attempt to acquire a wake lock to prevent the device from going to sleep during the conversation.
This can be disabled by setting the useWakeLock option to false:
`ts`
const conversation = await Conversation.startSession({
useWakeLock: false,
});
#### Return value
startSession returns a Conversation instance that can be used to control the session. The method will throw an error if the session cannot be established. This can happen if the user denies microphone access, or if the websocket connection
fails.
##### endSession
A method to manually end the conversation. The method will end the conversation and disconnect from websocket.
Afterwards the conversation instance will be unusable and can be safely discarded.
`js`
await conversation.endSession();
##### sendFeedback
A method for sending binary feedback to the agent.
The method accepts a boolean value, where true represents positive feedback and false negative feedback.onCanSendFeedbackChange
Feedback is always correlated to the most recent agent response and can be sent only once per response.
You can listen to to know if feedback can be sent at the given moment.
`js`
conversation.sendFeedback(true);
##### sendContextualUpdate
A method to send contextual updates to the agent.
This can be used to inform the agent about user actions that are not directly related to the conversation, but may influence the agent's responses.
`js`
conversation.sendContextualUpdate(
"User navigated to another page. Consider it for next response, but don't react to this contextual update."
);
##### sendUserMessage
Sends a text messages to the agent.
Can be used to let the user type in the message instead of using the microphone.
Unlike sendContextualUpdate, this will be treated as a user message and will prompt the agent to take its turn in the conversation.
`js`
sendButton.addEventListener("click", e => {
conversation.sendUserMessage(textInput.value);
textInput.value = "";
});
##### sendUserActivity
Notifies the agent about user activity.
The agent will not attempt to speak for at least 2 seconds after the user activity is detected.
This can be used to prevent the agent from interrupting the user when they are typing.
`js`
textInput.addEventListener("input", () => {
conversation.sendUserActivity();
});
##### getId
A method returning the conversation ID.
`js`
const id = conversation.getId();
##### setVolume
A method to set the output volume of the conversation. Accepts object with volume field between 0 and 1.
`js`
await conversation.setVolume({ volume: 0.5 });
##### muteMic
A method to mute/unmute the microphone.
`js
// Mute the microphone
conversation.setMicMuted(true);
// Unmute the microphone
conversation.setMicMuted(false);
`
##### getInputVolume / getOutputVolume
Methods that return the current input/output volume on a scale from 0 to 1 where 0 is -100 dB and 1 is -30 dB.
`js`
const inputVolume = await conversation.getInputVolume();
const outputVolume = await conversation.getOutputVolume();
##### getInputByteFrequencyData / getOutputByteFrequencyData
Methods that return Uint8Arrays containing the current input/output frequency data. See AnalyserNode.getByteFrequencyData for more information.
Note: These methods are only available for voice conversations. In WebRTC mode the audio is hardcoded to use pcm_48000, meaning any visualization using the returned data might show different patterns to WebSocket connections.
##### changeInputDevice
Allows you to change the audio input device during an active voice conversation. This method is only available for voice conversations.
Note: In WebRTC mode the input format and sample rate are hardcoded to pcm and 48000 respectively. Changing those values when changing the input device is a no-op.
`js
const conversation = await Conversation.startSession({
agentId: "
});
// Change to a specific input device
await conversation.changeInputDevice({
sampleRate: 16000,
format: "pcm",
preferHeadphonesForIosDevices: true,
inputDeviceId: "your-device-id",
});
`
##### changeOutputDevice
Allows you to change the audio output device during an active voice conversation. This method is only available for voice conversations.
Note: In WebRTC mode the output format and sample rate are hardcoded to pcm and 48000 respectively. Changing those values when changing the output device is a no-op.
`js`
// Change to a specific output device
await conversation.changeOutputDevice({
sampleRate: 16000,
format: "pcm",
outputDeviceId: "your-device-id",
});
Note: Device switching only works for voice conversations. If no specific deviceId is provided, the browser will use its default device selection. You can enumerate available devices using the MediaDevices.enumerateDevices() API.
Scribe is ElevenLabs' real-time speech-to-text API that provides low-latency transcription with support for both streaming microphone input and pre-recorded audio files.
`js
import { Scribe, RealtimeEvents } from "@elevenlabs/client";
// Connect with microphone streaming
const connection = Scribe.connect({
token: "your-token",
modelId: "scribe_v2_realtime",
microphone: {
echoCancellation: true,
noiseSuppression: true,
},
});
// Listen for transcripts
connection.on(RealtimeEvents.PARTIAL_TRANSCRIPT, (data) => {
console.log("Partial:", data.text);
});
connection.on(RealtimeEvents.COMMITTED_TRANSCRIPT, (data) => {
console.log("Committed:", data.text);
});
// Close connection when done
connection.close();
`
Scribe requires a single-use token for authentication. These tokens are generated via the ElevenLabs API on the server.
You should create an API endpoint on your server to generate these tokens:
`js
// Node.js server
app.get("/scribe-token", yourAuthMiddleware, async (req, res) => {
const response = await fetch(
"https://api.elevenlabs.io/v1/single-use-token/realtime_scribe",
{
method: "POST",
headers: {
"xi-api-key": process.env.ELEVENLABS_API_KEY,
},
}
);
const data = await response.json();
res.json({ token: data.token });
});
`
`js`
// Client
const response = await fetch("/scribe-token");
const { token } = await response.json();
Warning: Your ElevenLabs API key is sensitive, do not leak it to the client. Always generate the token on the server.
Automatically stream audio from the user's microphone:
`js
import { Scribe, RealtimeEvents } from "@elevenlabs/client";
const connection = Scribe.connect({
token: "your-token",
modelId: "scribe_v2_realtime",
microphone: {
deviceId: "optional-device-id", // Optional: specific microphone
echoCancellation: true,
noiseSuppression: true,
autoGainControl: true,
channelCount: 1,
},
});
`
The microphone stream is automatically converted to PCM16 format required by the API. In this mode audio is automatically committed.
For transcribing pre-recorded audio files or custom audio sources:
`js
import { Scribe, AudioFormat, RealtimeEvents } from "@elevenlabs/client";
const connection = Scribe.connect({
token: "your-token",
modelId: "scribe_v2_realtime",
audioFormat: AudioFormat.PCM_16000,
sampleRate: 16000,
});
// Send audio chunks as base64
connection.send({ audioBase64: base64AudioChunk });
// Signal end of audio segment
connection.commit();
`
#### Example: Transcribing an Audio File
`js
// Get file from input element
const fileInput = document.querySelector('input[type="file"]');
const audioFile = fileInput.files[0];
// Read file as ArrayBuffer
const arrayBuffer = await audioFile.arrayBuffer();
const audioData = new Uint8Array(arrayBuffer);
// Convert to base64 and send in chunks
const chunkSize = 8192; // 8KB chunks
for (let i = 0; i < audioData.length; i += chunkSize) {
const chunk = audioData.slice(i, i + chunkSize);
const base64 = btoa(String.fromCharCode(...chunk));
connection.send({ audioBase64: base64 });
// Optional: Add delay to simulate real-time streaming
await new Promise((resolve) => setTimeout(resolve, 100));
}
// Signal end of audio
connection.commit();
`
Subscribe to events using the connection instance:
`js
import { RealtimeEvents } from "@elevenlabs/client";
// Session started
connection.on(RealtimeEvents.SESSION_STARTED, () => {
console.log("Session started");
});
// Partial transcripts (interim results)
connection.on(RealtimeEvents.PARTIAL_TRANSCRIPT, (data) => {
console.log("Partial:", data.text);
});
// Committed transcripts
connection.on(RealtimeEvents.COMMITTED_TRANSCRIPT, (data) => {
console.log("Committed:", data.text);
});
// Committed transcripts with word-level timestamps
// Only received when includeTimestamps = true
connection.on(RealtimeEvents.COMMITTED_TRANSCRIPT_WITH_TIMESTAMPS, (data) => {
console.log("Committed:", data.text);
console.log("Timestamps:", data.words);
});
// Errors
connection.on(RealtimeEvents.ERROR, (error) => {
console.error("Error:", error);
});
// Authentication errors
connection.on(RealtimeEvents.AUTH_ERROR, (data) => {
console.error("Auth error:", data.error);
});
// Connection opened
connection.on(RealtimeEvents.OPEN, () => {
console.log("Connection opened");
});
// Connection closed
connection.on(RealtimeEvents.CLOSE, () => {
console.log("Connection closed");
});
// Quota exceeded
connection.on(RealtimeEvents.QUOTA_EXCEEDED, (data) => {
console.log("Quota exceeded:", data.error)
})
`
#### Common Options
All connection modes support these options:
`js
const connection = await scribe.connect({
token: "your-token", // Required: Single-use token
modelId: "scribe_v2_realtime", // Required: Model ID
baseUri: "wss://api.elevenlabs.io", // Optional: Custom endpoint
// Voice Activity Detection (VAD) settings
commitStrategy: CommitStrategy.MANUAL, // or CommitStrategy.VAD
vadSilenceThresholdSecs: 0.5, // Seconds of silence before committing
vadThreshold: 0.5, // VAD sensitivity (0-1)
minSpeechDurationMs: 100, // Minimum speech duration to process
minSilenceDurationMs: 500, // Minimum silence to detect pause
languageCode: "en", // ISO 639-1 language code
includeTimestamps: true // Whether to receive the committed_transcript_with_timestamps event after committing
});
`
#### Microphone-Specific Options
`js`
const connection = await scribe.connect({
// ... common options
microphone: {
deviceId: "optional-device-id",
echoCancellation: true,
noiseSuppression: true,
autoGainControl: true,
channelCount: 1,
},
});
#### Manual Audio Options
`js
import { AudioFormat } from "@elevenlabs/client";
const connection = Scribe.connect({
// ... common options
audioFormat: AudioFormat.PCM_16000, // or AudioFormat.PCM_24000
sampleRate: 16000, // Must match audioFormat
});
`
Scribe supports two commit strategies when in manual audio mode:
#### Manual
You explicitly control when to commit transcriptions:
`js
import { Scribe, CommitStrategy, RealtimeEvents } from "@elevenlabs/client";
const connection = Scribe.connect({
token: "your-token",
modelId: "scribe_v2_realtime",
commitStrategy: CommitStrategy.MANUAL,
audioFormat: AudioFormat.PCM_16000,
sampleRate: 16000,
});
connection.send({ audioBase64: base64Audio });
// Later, when you want to commit the segment
connection.commit();
`
#### Voice Activity Detection (VAD)
The API automatically detects when speech ends and commits the transcription:
`js
import { Scribe, CommitStrategy, RealtimeEvents } from "@elevenlabs/client";
const connection = Scribe.connect({
token: "your-token",
modelId: "scribe_v2_realtime",
commitStrategy: CommitStrategy.VAD,
audioFormat: AudioFormat.PCM_16000,
sampleRate: 16000,
});
`
#### close()
Close the connection and clean up resources:
`js`
connection.close();
#### send(options)
Send audio data (manual mode only):
`js`
connection.send({
audioBase64: base64AudioData,
commit: false, // Optional: commit immediately
sampleRate: 16000, // Optional: override sample rate
previousText: "Previous transcription text", // Optional: include text from a previous transcription or base64 encoded audio data. Will be used to provide context to the model. Can only be sent in the first audio chunk.
});
Warning: The previousTextfield can only be sent in the first audio chunk of a session. If sent in any other chunk an error will be returned.
#### commit()
Manually commit the current segment:
`js`
connection.commit();
Full TypeScript types are included:
`typescript
import {
Scribe,
RealtimeConnection,
AudioFormat,
CommitStrategy,
RealtimeEvents,
type AudioOptions,
type MicrophoneOptions,
type PartialTranscriptMessage,
type CommittedTranscriptMessage,
} from "@elevenlabs/client";
const connection: RealtimeConnection = await scribe.connect({
token: "your-token",
modelId: "scribe_v2_realtime",
microphone: {
echoCancellation: true,
},
});
`
Always handle errors appropriately:
`js
import { Scribe, RealtimeEvents } from "@elevenlabs/client";
try {
const connection = Scribe.connect({
token: "your-token",
modelId: "scribe_v2_realtime",
microphone: {},
});
// Generic event that fires on all errors, including auth and quota exceeded
connection.on(RealtimeEvents.ERROR, (error) => {
console.error("Connection error:", error);
});
connection.on(RealtimeEvents.AUTH_ERROR, (data) => {
console.error("Authentication failed:", data.error);
});
connection.on(RealtimeEvents.QUOTA_EXCEEDED, (data) => {
console.error("Quota exceeded:", data.error);
});
} catch (error) {
console.error("Failed to connect:", error);
}
`
If your application has a tight Content Security Policy and does not allow data: or blob: in the script-src (w3.org/TR/CSP2#source-list-guid-matching), you self-host the needed files in the public folder.
Whitelisting these values is not recommended w3.org/TR/CSP2#source-list-guid-matching.
Add the worklet files to your public folder eg public/elevenlabs.
`
@elevenlabs/client/scripts/
Then call start with values in options as workletPaths.
It is recommended to update the scripts with a build script like
`js
import { viteStaticCopy } from 'vite-plugin-static-copy'
import { createRequire } from 'node:module';
import path from 'path';
const require = createRequire(import.meta.url);
export default {
plugins: [
viteStaticCopy({
targets: [
{
src: require.resolve('@elevenlabs/client')/dist/worklets/audioConcatProcessor.js',
dest: 'dist',
},
{
src: require.resolve('@elevenlabs/client')/dist/worklets/rawAudioProcessor.js',
dest: 'dist',
},
],
}),
],
}
``
Please, refer to the README.md file in the root of this repository.
Please, create an issue first to discuss the proposed changes. Any contributions are welcome!
Remember, if merged, your code will be used as part of a MIT licensed project. By submitting a Pull Request, you are giving your consent for your code to be integrated into this library.