RTP
RTP (Real-time Transport Protocol) is a network protocol designed for delivering audio and video over IP networks in real-time, such as in video conferencing, streaming media, and VoIP applications. It provides end-to-end delivery services for data with real-time characteristics, including payload type identification, sequence numbering, timestamping, and delivery monitoring, but does not guarantee delivery or quality of service—those are handled by companion protocols like RTCP.
Developers should learn RTP when building applications that require real-time multimedia transmission, such as video chat apps (e.g., Zoom, Skype), live streaming platforms, or telephony systems, as it is the standard protocol for such use cases. It is essential for ensuring synchronized playback and handling jitter and packet loss in time-sensitive communications, often paired with signaling protocols like SIP or WebRTC.