How to Mute/Unmute an Agent Microphone via the Genesys Cloud WebRTC API

What You Will Build

• This tutorial demonstrates how to programmatically mute and unmute an agent’s microphone during an active Genesys Cloud interaction using the WebRTC Client SDK.
• The solution utilizes the Genesys Cloud WebRTC Client SDK (JavaScript) which is the standard for building custom agent desktops or integrating into existing web applications.
• The primary programming language is JavaScript (ES6+), suitable for Node.js backend triggers or browser-based frontend logic.

Prerequisites

• OAuth Client: A Genesys Cloud OAuth client with the webchat:manage or voice:manage scope depending on the interaction type, though WebRTC client operations typically require the user to be authenticated via the standard login flow. For direct API control of media, the voice:manage scope is critical if using REST, but the WebRTC SDK handles internal signaling.
• SDK Version: Genesys Cloud WebRTC Client SDK (@genesyscloud/webchat-client-sdk or the pure WebRTC client @genesyscloud/webrtc-client). Note: For voice calls, the @genesyscloud/webrtc-client is the specific library.
• Runtime: Node.js 16+ or a modern browser environment.
• Dependencies:
  • @genesyscloud/webrtc-client (Latest stable version)
  • @genesyscloud/auth (For handling token refresh if not using a persistent session)

Authentication Setup

The Genesys Cloud WebRTC Client SDK does not use standard REST OAuth tokens for the media stream itself. It uses a WebSocket connection authenticated by a temporary access token. However, to initialize the client, you need a valid access token for the Genesys Cloud user.

The following code snippet shows how to initialize the WebRTC client with a valid access token. In a production environment, this token is obtained via the standard OAuth 2.0 Authorization Code Grant or Client Credentials Grant (for server-to-server, though media control usually requires a user context).

import { WebRtcClient } from '@genesyscloud/webrtc-client';
import { AuthClient } from '@genesyscloud/auth';

// Configuration
const REGION = 'mypurecloud.com'; // e.g., 'mypurecloud.com' or 'us-east-1.mypurecloud.com'
const CLIENT_ID = 'your-client-id';
const CLIENT_SECRET = 'your-client-secret';
const GRANT_TYPE = 'client_credentials'; // Or 'authorization_code' for user context

// Initialize Auth Client to get a token
const authClient = new AuthClient({
  clientId: CLIENT_ID,
  clientSecret: CLIENT_SECRET,
  grantType: GRANT_TYPE,
  region: REGION
});

async function getToken() {
  try {
    const tokenResponse = await authClient.auth();
    return tokenResponse.accessToken;
  } catch (error) {
    console.error('Failed to authenticate:', error);
    throw new Error('Authentication failed');
  }
}

Important: The WebRTC client requires a user with a valid license and a configured skill group or queue to accept calls. Ensure the authenticated user has the necessary permissions to manage voice interactions.

Implementation

Step 1: Initialize the WebRTC Client and Connect

Before controlling media, you must establish a connection to the Genesys Cloud WebRTC service. The WebRtcClient manages the underlying WebSocket connection and SDP negotiation.

async function initializeWebRtcClient(accessToken) {
  const client = new WebRtcClient({
    region: REGION,
    accessToken: accessToken,
    // Optional: Log level for debugging
    logLevel: 'info' 
  });

  try {
    // Connect to the Genesys Cloud WebRTC service
    await client.connect();
    console.log('Connected to Genesys Cloud WebRTC service');
    return client;
  } catch (error) {
    console.error('Failed to connect to WebRTC service:', error);
    throw new Error('WebRTC connection failed');
  }
}

Step 2: Handle Active Interaction and Media Control

The core logic involves listening for an active interaction (call) and then invoking the mute and unmute methods on the WebRtcClient instance. The Genesys Cloud WebRTC Client SDK exposes mute() and unmute() methods that control the local media tracks.

Key Concept: Muting in WebRTC does not stop the audio from being captured by the microphone. Instead, it replaces the audio track with a silence track or sets the track’s enabled property to false. The Genesys Cloud platform interprets this as a mute state and updates the agent’s UI accordingly.

/**
 * Mutes the agent's microphone for the current active interaction.
 * @param {WebRtcClient} client - The initialized WebRTC client instance.
 */
async function muteMicrophone(client) {
  try {
    // Check if there is an active interaction
    const activeInteraction = client.getActiveInteraction();
    
    if (!activeInteraction) {
      console.warn('No active interaction found to mute.');
      return;
    }

    // Mute the local audio track
    // The SDK handles the signaling to Genesys Cloud to update the call state
    await client.mute();
    
    console.log('Agent microphone muted successfully.');
    
    // Optional: Update local UI state
    updateLocalUiState('muted');

  } catch (error) {
    console.error('Failed to mute microphone:', error);
    // Handle specific errors like 401 (Unauthorized) or 503 (Service Unavailable)
    if (error.code === '401') {
      console.error('Session expired. Please re-authenticate.');
    }
  }
}

/**
 * Unmutes the agent's microphone for the current active interaction.
 * @param {WebRtcClient} client - The initialized WebRTC client instance.
 */
async function unmuteMicrophone(client) {
  try {
    const activeInteraction = client.getActiveInteraction();
    
    if (!activeInteraction) {
      console.warn('No active interaction found to unmute.');
      return;
    }

    // Unmute the local audio track
    await client.unmute();
    
    console.log('Agent microphone unmuted successfully.');
    
    // Optional: Update local UI state
    updateLocalUiState('unmuted');

  } catch (error) {
    console.error('Failed to unmute microphone:', error);
  }
}

// Helper function to simulate UI updates
function updateLocalUiState(state) {
  // In a real application, this would update the DOM or React/Vue state
  console.log(`UI State updated to: ${state}`);
}

Step 3: Listen for Mute State Changes

It is critical to listen for state changes from the Genesys Cloud platform. For example, if the supervisor mutes the agent, or if the agent mutes via the Genesys Cloud desktop, your custom application needs to reflect that change. The WebRtcClient emits events for these changes.

function setupMuteStateListeners(client) {
  // Listen for local mute state changes
  client.on('mute', () => {
    console.log('Event: Agent microphone muted locally.');
    updateLocalUiState('muted');
  });

  client.on('unmute', () => {
    console.log('Event: Agent microphone unmuted locally.');
    updateLocalUiState('unmuted');
  });

  // Listen for remote mute state changes (e.g., supervisor mute)
  // Note: The event name may vary slightly depending on the SDK version.
  // Check the official documentation for the exact event name.
  client.on('remoteMute', () => {
    console.log('Event: Agent microphone muted by remote party.');
    updateLocalUiState('muted_by_supervisor');
  });

  client.on('remoteUnmute', () => {
    console.log('Event: Agent microphone unmuted by remote party.');
    updateLocalUiState('unmuted');
  });
}

Complete Working Example

The following is a complete, runnable Node.js script that demonstrates the full lifecycle: authentication, connection, handling a mock interaction (for demonstration purposes, as a real call requires a live dial-in), and toggling mute state.

Note: This example assumes you have a valid Genesys Cloud organization and OAuth client configured. For a real call, you would need to initiate an outbound call or accept an inbound call using the SDK’s startCall or acceptCall methods. This example focuses on the mute/unmute logic once an interaction is active.

import { WebRtcClient } from '@genesyscloud/webrtc-client';
import { AuthClient } from '@genesyscloud/auth';
import dotenv from 'dotenv';

// Load environment variables
dotenv.config();

const REGION = process.env.GENESYS_REGION || 'mypurecloud.com';
const CLIENT_ID = process.env.GENESYS_CLIENT_ID;
const CLIENT_SECRET = process.env.GENESYS_CLIENT_SECRET;

if (!CLIENT_ID || !CLIENT_SECRET) {
  console.error('Missing GENESYS_CLIENT_ID or GENESYS_CLIENT_SECRET in environment variables.');
  process.exit(1);
}

let webRtcClient = null;

async function main() {
  try {
    console.log('Starting Genesys Cloud WebRTC Mute/Unmute Demo...');

    // Step 1: Authenticate
    const authClient = new AuthClient({
      clientId: CLIENT_ID,
      clientSecret: CLIENT_SECRET,
      grantType: 'client_credentials',
      region: REGION
    });

    console.log('Authenticating...');
    const tokenResponse = await authClient.auth();
    const accessToken = tokenResponse.accessToken;
    console.log('Authenticated successfully.');

    // Step 2: Initialize WebRTC Client
    console.log('Initializing WebRTC Client...');
    webRtcClient = new WebRtcClient({
      region: REGION,
      accessToken: accessToken,
      logLevel: 'info'
    });

    // Setup listeners
    setupMuteStateListeners(webRtcClient);

    // Step 3: Connect
    console.log('Connecting to WebRTC service...');
    await webRtcClient.connect();
    console.log('Connected to WebRTC service.');

    // Step 4: Simulate Mute/Unmute Actions
    // In a real scenario, you would wait for an active call here.
    // For demonstration, we will attempt to mute/unmute immediately.
    // Note: This may fail if there is no active interaction.
    
    console.log('\n--- Attempting to Mute ---');
    await muteMicrophone(webRtcClient);

    console.log('\n--- Attempting to Unmute ---');
    await unmuteMicrophone(webRtcClient);

    // Keep the process running for a few seconds to observe logs
    console.log('\nDemo completed. Shutting down in 5 seconds...');
    setTimeout(() => {
      cleanup();
    }, 5000);

  } catch (error) {
    console.error('Fatal error:', error);
    cleanup();
    process.exit(1);
  }
}

/**
 * Mutes the agent's microphone for the current active interaction.
 */
async function muteMicrophone(client) {
  try {
    const activeInteraction = client.getActiveInteraction();
    
    if (!activeInteraction) {
      console.warn('No active interaction found. Mute action skipped.');
      return;
    }

    await client.mute();
    console.log('Agent microphone muted successfully.');
    updateLocalUiState('muted');

  } catch (error) {
    console.error('Failed to mute microphone:', error);
  }
}

/**
 * Unmutes the agent's microphone for the current active interaction.
 */
async function unmuteMicrophone(client) {
  try {
    const activeInteraction = client.getActiveInteraction();
    
    if (!activeInteraction) {
      console.warn('No active interaction found. Unmute action skipped.');
      return;
    }

    await client.unmute();
    console.log('Agent microphone unmuted successfully.');
    updateLocalUiState('unmuted');

  } catch (error) {
    console.error('Failed to unmute microphone:', error);
  }
}

function setupMuteStateListeners(client) {
  client.on('mute', () => {
    console.log('Event: Agent microphone muted locally.');
    updateLocalUiState('muted');
  });

  client.on('unmute', () => {
    console.log('Event: Agent microphone unmuted locally.');
    updateLocalUiState('unmuted');
  });
}

function updateLocalUiState(state) {
  console.log(`[UI] State updated to: ${state}`);
}

function cleanup() {
  if (webRtcClient) {
    webRtcClient.disconnect();
    console.log('WebRTC Client disconnected.');
  }
  process.exit(0);
}

// Run the main function
main();

Common Errors & Debugging

Error: No active interaction found

• Cause: You are calling mute() or unmute() when the agent is not in an active call.
• Fix: Ensure that the getActiveInteraction() method returns a valid object before attempting to mute. In a production application, wrap the mute/unmute buttons in a conditional check that only enables them when a call is active.

Error: 401 Unauthorized

• Cause: The access token has expired or is invalid.
• Fix: Implement token refresh logic. The AuthClient from @genesyscloud/auth can be configured to automatically refresh tokens. If using a custom token flow, ensure you are passing a fresh token when initializing the WebRtcClient.

Error: 503 Service Unavailable

• Cause: The Genesys Cloud WebRTC service is temporarily unavailable or the WebSocket connection was dropped.
• Fix: Implement reconnection logic. The WebRtcClient emits a disconnected event. Listen for this event and attempt to reconnect after a short delay.

webRtcClient.on('disconnected', () => {
  console.warn('WebRTC connection lost. Attempting to reconnect...');
  setTimeout(async () => {
    try {
      await webRtcClient.connect();
      console.log('Reconnected successfully.');
    } catch (error) {
      console.error('Reconnection failed:', error);
    }
  }, 5000);
});

Error: TypeError: client.mute is not a function

• Cause: You are using an outdated version of the @genesyscloud/webrtc-client SDK or have imported the wrong class.
• Fix: Ensure you are using the latest version of the SDK. Check the package.json for the @genesyscloud/webrtc-client version. Verify that you are instantiating WebRtcClient and not a different class.

Official References

• Genesys Cloud WebRTC Client SDK Documentation
• Genesys Cloud API Explorer: WebRTC
• Genesys Cloud OAuth Authentication Guide