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.

If you attempt to assign a WebRTC phone to a user who already has a WebRTC phone assigned to them, you will see the error message:

This person is already assigned  to a WebRTC phone.

appear below the Person box. You can avoid this scenario by using the WebRTC Person feature. When you add this feature to the Phone Management page, you get a searchable list that you can use to search for individual names to see if they already have a WebRTC phone assigned to them.

To enable and use the WebRTC Person feature on the Phone Management page:

1. Click Admin.
2. Under Telephony, click Phone Management.
3. Select the Phones tab.
4. On the top right of the column headers, click Select columns . 
5. In the Data to Display panel, click the WebRTC Person check box
6. Click Save and the WebRTC Person column appears on the Phones page.
7. To find a person, type a name in the Type to filter this page box and press [Enter].

Yes. While the Geo-Lookup TURN feature and the Relay/TURN Behavior feature are designed to work independently of one another, they definitely can work together. The Relay/TURN Behavior feature is designed to narrow down the number of open outbound ports on your firewall needed for Genesys Cloud’s WebRTC client implementation to access the TURN services. And, the Geo-Lookup TURN feature optimizes the network path for WebRTC calls routed through those cloud TURN services, such that they access the closest TURN service.

For more information, see Use the Relay/TURN Behavior feature and Configure your site for Global Media Fabric.

Yes, you can use a CSV file to import WebRTC phone configurations into Genesys Cloud. For detailed instructions on how to do so, see the Import phones article.

Yes. For more information, see Set Genesys Cloud sound preferences.

The Genesys Cloud WebRTC phone runs right from your browser. There is no additional software to download or install. You simply enable the Genesys Cloud WebRTC phone and you can immediately use it to make and receive calls. For more information, see Set up Genesys Cloud WebRTC phones.

The PureCloud Softphone is different from the Genesys Cloud WebRTC phone.  It is a Windows-only software-based phone. It requires a separate software package to download and install before you can begin to use the phone to make and receive calls.

Yes. Genesys Cloud sends alerts, such as incoming calls, to your default computer audio. The Genesys Cloud WebRTC phone sends call audio to the speakers of your choice. To choose default audio devices for your computer, use your computer’s controls. To change the audio devices used during a web-based phone call, use the phone settings found where you select the Genesys Cloud WebRTC phone as your phone.