FAQs: WebRTC

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.

The built-in WebRTC Diagnostics app may not load properly due to SSO vendor policies. If you encounter this situation, you need to run the standalone Genesys Cloud WebRTC Diagnostics app.

No. Genesys Cloud does not natively provide QoS for WebRTC phones. Instead, Genesys Cloud provides you with the Mean Opinion Score (MOS), which is a measurement of the voice quality of an interaction. To calculate MOS, Genesys Cloud uses an industry standard measurement methodology to rank audio quality from 1 (unacceptable) to 5 (excellent). You can view the MOS values in the Interactions view. For more information, see Mean Opinion Score (MOS) Overview and Interactions view.

WebRTC applications require a server to relay traffic between the clients over the internet. The server that relays that traffic is called a TURN (Traversal Using Relay NAT) server. TURN is a protocol for relaying network traffic.