Difference between revisions of "WebRTC/Current/WebRTCPEGuide/Overview"
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| − | |structuredtext=Web Real-Time Communication (WebRTC) is a real time communication over the internet that enables agent to connect | + | |structuredtext=Web Real-Time Communication (WebRTC) is a real-time communication over the internet that enables an agent to connect with the Genesys contact center environment to perform their business operations. |
| − | WebRTC is a shared ( | + | WebRTC is a shared (multitenant) service that acts as the signaling and media gateway. The signaling gateway is used to interwork WebRTC with Session Initiation Protocol (SIP), and the media gateway is used to terminate the Interactive Connectivity Establishment (ICE) and Secure Real-time Transport Protocol (SRTP). |
| − | + | WebRTC bridges the calls that are initiated/received by the browser. The SIP Server handles these calls 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. | |
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| + | *Azure Kubernetes Service (AKS) | ||
*Google Kubernetes Engine (GKE) | *Google Kubernetes Engine (GKE) | ||
*OpenShift Container Platform (OpenShift) | *OpenShift Container Platform (OpenShift) | ||
Revision as of 05:00, September 30, 2022
Contents
Learn about WebRTC and how it works in Genesys Multicloud CX private edition.
Web Real-Time Communication (WebRTC) is a real-time communication over the internet that enables an agent to connect with the Genesys contact center environment to perform their business operations.
WebRTC is a shared (multitenant) service that acts as the signaling and media gateway. The signaling gateway is used to interwork WebRTC with Session Initiation Protocol (SIP), and the media gateway is used to terminate the Interactive Connectivity Establishment (ICE) and Secure Real-time Transport Protocol (SRTP).
WebRTC bridges the calls that are initiated/received by the browser. The SIP Server handles these calls 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:
- Azure Kubernetes Service (AKS)
- Google Kubernetes Engine (GKE)
- OpenShift Container Platform (OpenShift)
See the WebRTC Release Notes for information about when support was introduced.