About WebRTC

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This topic is part of the manual WebRTC Private Edition Guide for version Current of WebRTC.

Learn about WebRTC and how it works in Genesys Multicloud CX private edition.

Related documentation:

Web Real-Time Communication (WebRTC) is a real time communication over the internet that enables agent to connect into Genesys contact center environment to perform their business operations.

WebRTC is a shared (multi-tenant) service that acts as the signalling and media gateway. The signalling gateway is used to interwork WebRTC with Session Initiation Protocol (SIP), and the media gateway is used to terminate Interactive Connectivity Establishment (ICE) and Secure Real-time Transport Protocol (SRTP).

Calls that are initiated/received by the browser is bridged through the WebRTC. These calls are handled by the SIP server as a SIP call to provide core Genesys features such as routing and IVR. These features are handled by Genesys for browser endpoints with the help of MCP in the call flow. Third party component CoTURN is used to implement TURN and STUN servers.

Supported Kubernetes platforms

WebRTC is supported on the following cloud platforms:

  • Google Kubernetes Engine (GKE)
  • OpenShift Container Platform (OpenShift)

See the WebRTC Release Notes for information about when support was introduced.