Does Genesys Cloud continue collecting audio in between WebRTC calls when using the Modern WebRTC feature and persistent connection?

Genesys Cloud does not collect, route, or store audio in between WebRTC calls when using the Modern WebRTC feature and Persistent Connection. Genesys Cloud only collects audio when you are actively on a call. Modern WebRTC behavior differs by configuration:

• With Modern WebRTC enabled: The WebRTC client retains microphone access so calls can connect quickly. However, between calls, the client is put on hold and does not send RTP media traffic. The client only exchanges lightweight health/keep-alive messages. When another call comes in, the client resumes streaming audio to a media session where it can be routed, recorded, and transcribed per policy.
  
  
• Without Modern WebRTC enabled: The WebRTC client retains microphone permission between calls and continues sending RTP media traffic toward the cloud. However, because no active media session exists in Genesys Cloud during this time, that traffic is not processed, routed, or stored. Audio is only handled when a call begins and a media session is established.