Genesys Cloud WebRTC phone enables users to make and receive calls directly through the Genesys Cloud app using WebRTC technology, eliminating the need for physical phones or separate softphone applications. The feature operates on desktop and web apps with compatible browsers and requires specific permissions including conversation call access, telephony self-station association, and station management capabilities. A telephony administrator must create and assign a WebRTC phone to each user, which users then access through the Calls panel by selecting Phone Details and searching for their assigned phone name. Configuration is managed through the Settings option in the Calls panel, where users can enable third-party application calling, select microphone and speaker devices through Audio Controls, and adjust output volume via Device Volumes settings. Custom audio profiles can be created to support different calling scenarios. The platform incorporates comprehensive diagnostic functionality through a "Run diagnostics" tool that tests microphone performance, connectivity, and throughput for troubleshooting purposes. Call management features include number entry during calls, non-ACD call transfers, and softphone configuration options. Technical configuration requires access to Genesys Cloud CX Utility for WebRTC settings and trunk creation, with migration pathways available from legacy PureCloud Softphone solutions. Genesys recommends using high-quality phone headsets and standard computer audio settings for optimal performance. Users should close other browser instances when enabling third-party application calling to prevent conflicts. The WebRTC client alerting feature allows configuration of priority when multiple Genesys Cloud clients are open with the same WebRTC phone selected. Users without an assigned WebRTC phone or those experiencing unresolved issues should contact their IT department or administrator for setup assistance. Additional support resources include community forums and troubleshooting documentation for WebRTC phone-related issues.