Could use a hand troubleshooting this persistent issue with the Genesys Cloud JavaScript SDK (v3.12.1) regarding media control. I am building a custom softphone interface for internal agents using the @genesyscloud/genesyscloud-embed and @genesyscloud/genesyscloud-sdk packages. The goal is to programmatically toggle the microphone mute state via the client-side API during an active voice conversation.
I have successfully established the connection and can initiate outbound calls using conversation.createConversation. The audio streams are active, and I can hear the remote party. However, when attempting to mute the local audio stream by invoking the method conversation.setMute(true), the function returns a Promise that resolves without error, but the microphone remains unmuted from the perspective of the remote participant. The UI reflects the mute state correctly, but the actual WebRTC track is not being stopped.
Here is the relevant code snippet:
const sdk = new GenesysCloudSdk();
const conversation = sdk.Conversations();
// Inside the call active state handler
async function toggleMicMute(isMuted) {
try {
const result = await conversation.setMute(isMuted);
console.log('Mute state updated:', result);
// Returns { id: 'call-id', muted: true }
} catch (error) {
console.error('Failed to mute:', error);
}
}
I have verified that the call-id passed to the context is correct by cross-referencing it with the conversation.getConversation response. I also checked the browser’s DevTools WebRTC tab, and the local audio track’s enabled property does not change to false after the method call. Is there a specific permission scope or additional configuration required in the init options for media control? Or is setMute deprecated in favor of a direct WebRTC track manipulation approach?
The silent failure usually stems from attempting to modify the MediaStream track constraints after the WebRTC connection has stabilized, or relying on the SDK’s high-level wrapper without verifying the underlying getUserMedia state. In my k6 load testing scenarios, I’ve seen this pattern fail when the client tries to toggle mute before the track object is fully attached to the peer connection. You need to directly access the active audio track and toggle its enabled property, rather than calling a generic setMute method that might be deprecated or poorly implemented in your specific SDK version. Here is the robust pattern for handling this in the browser environment:
// Retrieve the active audio track from the local stream
const audioTrack = localStream.getAudioTracks()[0];
if (audioTrack) {
// Toggle the enabled state directly on the track
audioTrack.enabled = !audioTrack.enabled;
// Optional: Log the state for debugging in high-concurrency tests
console.log(`Mic state changed to: ${audioTrack.enabled ? 'Unmuted' : 'Muted'}`);
} else {
console.error('No active audio track found to mute/unmute.');
}
This approach bypasses the SDK’s internal state machine, which often gets out of sync during rapid state changes or network jitter. When running load tests or simulating high-concurrency agent sessions, ensure you are not triggering this toggle during the initial handshake phase. The enabled property on the MediaStreamTrack is the standard WebRTC mechanism and is significantly more reliable than SDK-specific wrappers. If you are still seeing issues, check your browser console for NotAllowedError or AbortError messages, which indicate that the browser’s media permissions were revoked or the stream was already closed. Also, verify that your OAuth scope includes conversation:write if you are attempting to update conversation attributes alongside the media control, as missing permissions can sometimes cause silent failures in the SDK’s event loop.