What protocol is commonly used by TSCs for signaling in VoIP applications?

The Session Initiation Protocol (SIP) is the correct answer because it serves as a signaling protocol specifically designed for initiating, maintaining, and terminating real-time sessions that involve video, voice, messaging, and other communications. In VoIP applications, SIP is essential for establishing connections between call participants, negotiating the parameters of the session, and ensuring that all necessary information is exchanged for effective communication.

SIP allows devices to discover and communicate with each other, set up calls, and manage the call state throughout its lifecycle. It handles critical functions such as user registration, call setup, session preferences, and call termination, making it the backbone of VoIP signaling.

In contrast, Real-Time Transport Protocol (RTP) is primarily used for delivering audio and video over IP networks but does not handle the signaling aspect. Media Gateway Control Protocol (MGCP) focuses on controlling media gateways on VoIP networks, primarily dealing with call control at the media gateway rather than user signaling. Hypertext Transfer Protocol (HTTP) is used for transferring web pages and is not relevant in the context of VoIP signaling or session management. Thus, SIP's specific focus on signaling makes it the most appropriate choice for VoIP applications.