Just noticed that the WebRTC softphone connection drops immediately after answering a call when using the custom Zendesk-to-Genesys integration widget. The migration from Zendesk Talk to Genesys Cloud Web Messaging and Voice is progressing, but the audio bridge seems unstable in the browser environment. We are using the Genesys Cloud Web SDK v2.65.0 embedded in a React frontend that mimics the old Zendesk agent desk layout.
When an inbound call arrives, the call.answer() method triggers successfully, and the UI updates to show the ‘In Progress’ state. However, the audio stream fails to initialize within 2 seconds. The browser console throws a MediaStreamError: DOMException: The source is already connected error, followed by a Genesys specific warning about rtc_session_timeout. This behavior is inconsistent with the standard Genesys Cloud desktop app, which works fine. Since we are replicating the Zendesk agent experience, we need this web-based softphone to be rock-solid for our support team in Paris.
We have verified that the browser permissions for microphone and camera are granted. The network logs show that the STUN/TURN servers are reachable, but the actual RTP packets never seem to flow once the answer signal is sent. The Architect flow is standard inbound routing with no complex data actions that would delay the answer signal.
- Tested across Chrome (v120) and Firefox (v121) with identical results, ruling out browser-specific codec issues. Disabled all extensions to prevent interference with the WebRTC API.
- Checked the Genesys Cloud API logs for the
POST /api/v2/interactions/{id}endpoint. The status changes toacceptedcorrectly, but the subsequentmediaevent never fires with a valid stream URL.
Is there a specific configuration in the Web SDK or the Architect flow that needs to be adjusted to handle the media stream handoff correctly during a migration scenario? We want to avoid falling back to CTI desktops if we can resolve this web-based issue.