What types of Phone Trunks are available in Genesys Cloud?

Prepare for the Genesys Cloud Implementation Exam. Study with flashcards and multiple-choice questions, each question comes with hints and explanations. Ace your exam preparation now!

Multiple Choice

What types of Phone Trunks are available in Genesys Cloud?

In Genesys Cloud, the types of phone trunks available are primarily SIP and WebRTC. SIP (Session Initiation Protocol) is a widely used signaling protocol that enables voice and video calls over the internet. It allows for the establishment, modification, and termination of communication sessions, making it a fundamental choice for integrating telephony with various applications and systems. SIP trunks replace traditional phone lines, enabling organizations to make and receive calls using their internet connection, thus offering more flexibility and often reducing costs. WebRTC (Web Real-Time Communication) is a technology that facilitates direct peer-to-peer communication between web browsers, allowing users to communicate via video and audio calls without the need for additional plugins or software installations. This is particularly useful within the Genesys Cloud platform, as it enables real-time communication capabilities directly from the browser, enhancing user experience and accessibility. These two types of trunks provide different approaches to telephony, catering to the diverse needs of organizations. SIP offers robustness and widespread compatibility with various systems, while WebRTC enables modern, browser-based communication, making it suitable for customer engagement scenarios. Other choices represent different concepts not applicable to the trunk types in this context. For instance, analog and digital refer more to traditional telephony and signal formats rather than

In Genesys Cloud, the types of phone trunks available are primarily SIP and WebRTC.

SIP (Session Initiation Protocol) is a widely used signaling protocol that enables voice and video calls over the internet. It allows for the establishment, modification, and termination of communication sessions, making it a fundamental choice for integrating telephony with various applications and systems. SIP trunks replace traditional phone lines, enabling organizations to make and receive calls using their internet connection, thus offering more flexibility and often reducing costs.

WebRTC (Web Real-Time Communication) is a technology that facilitates direct peer-to-peer communication between web browsers, allowing users to communicate via video and audio calls without the need for additional plugins or software installations. This is particularly useful within the Genesys Cloud platform, as it enables real-time communication capabilities directly from the browser, enhancing user experience and accessibility.

These two types of trunks provide different approaches to telephony, catering to the diverse needs of organizations. SIP offers robustness and widespread compatibility with various systems, while WebRTC enables modern, browser-based communication, making it suitable for customer engagement scenarios.

Other choices represent different concepts not applicable to the trunk types in this context. For instance, analog and digital refer more to traditional telephony and signal formats rather than

More practice questions