What is Real-time Transport Protocol?
{{IPstack}} The '''Real-time Transport Protocol''' ('''RTP''') is a [[network protocol]] used to send [[Sound recording and reproduction|audio]] and [[video]] over [[IP|IP networks]]. RTP is used in [[communication]] and [[entertainment]] systems that use [[streaming media]], such as [[telephony]], [[video teleconference]] applications, [[IPTV|television services]] and web-based [[push-to-talk]] features. RTP normally runs over [[User Datagram Protocol]] (UDP). It is used with the [[RTP Control Protocol]] (RTCP). While RTP carries the audio and video,, RTCP monitors [[transmission statistics]] and [[quality of service]] (QoS). RTP is one of the basic [[Technology|technologies]] in the use of [[Voice over IP]]. It is often used with a [[signaling protocol]] such as the [[Session Initiation Protocol]] (SIP) which creates connections across the network. RTP was developed by the Audio-Video Transport Working Group of the [[Internet Engineering Task Force]] (IETF). It was first [[Publishing|published]] in 1996 as {{IETF RFC|1889}}. It was replaced by {{IETF RFC|3550}} in 2003.<ref>{{Cite web |last=Wright |first=Gavin |title=What is the Real-time Transport Protocol (RTP)? |url=https://www.techtarget.com/searchnetworking/definition/Real-Time-Transport-Protocol |access-date=2022-11-10 |website=TechTarget |language=en}}</ref> ==References== {{reflist}} [[Category:internet protocols]]