Which protocol does Webex Calling primarily use for voice traffic?

Webex Calling primarily utilizes the Session Initiation Protocol (SIP) for managing voice traffic. SIP is a signaling protocol widely used for initiating, maintaining, and terminating real-time sessions that include voice, video, and messaging applications. It serves as the foundation for controlling multimedia communication sessions over Internet Protocol (IP) networks.

In the context of Webex Calling, SIP allows for the establishment of connections between different endpoints, such as phones or soft clients, facilitating voice calls over the internet. SIP is pivotal in signaling the setup and teardown of communications, negotiating media capabilities, and managing call states effectively.

The other protocols listed serve different purposes; for instance, HTTP is primarily used for transferring hypertext, and FTP is aimed at file transfers. RTP (Real-Time Protocol), while essential for delivering audio and video over IP networks, is mainly concerned with the transport of the media streams, not the signaling aspect of call management. Therefore, while RTP is an important part of the voice communication process in conjunction with SIP, it is SIP that fundamentally manages the call flow in Webex Calling.