Which protocol does SIP commonly utilize for media transport?

SIP, or Session Initiation Protocol, is designed for initiating, maintaining, and terminating voice, video, and messaging sessions over IP networks. When it comes to media transport for these sessions, the protocol that SIP commonly utilizes is RTP, or Real-Time Transport Protocol.

RTP is specifically designed for delivering audio and video over IP networks and provides features such as payload type identification, sequencing, and timing reconstruction, which are vital for maintaining the quality of media streams. It supports transporting real-time audio and video data and allows for synchronization of various media types, such as voice and video during a call or conferencing session.

In contrast, other protocols mentioned, such as HTTP and FTP, are not suitable for real-time media streaming. HTTP (Hypertext Transfer Protocol) is mainly used for transferring hypertext documents on the web, and while it can handle file transfers, it is not efficient for real-time communication needs. HTTPS is the secure version of HTTP, providing encrypted communication over a computer network, but it still does not address the real-time requirements of media transport. FTP (File Transfer Protocol) is designed for transferring files over a network but lacks the capability to manage real-time data, making it ineffective for audio or video transport during active sessions.

Thus,