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The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks.RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.
The Real-time Transport Protocol (RTP) specifies a general-purpose data format and network protocol for transmitting digital media streams on Internet Protocol (IP) networks. The details of media encoding, such as signal sampling rate, frame size and timing, are specified in an RTP payload format .
The Secure Real-time Transport Protocol (SRTP) is a profile for Real-time Transport Protocol (RTP) intended to provide encryption, message authentication and integrity, and replay attack protection to the RTP data in both unicast and multicast applications.
The Real-Time Streaming Protocol (RTSP), Real-time Transport Protocol (RTP) and the Real-time Transport Control Protocol (RTCP) were specifically designed to stream media over networks. RTSP runs over a variety of transport protocols, [78] while the latter two are built on top of UDP.
This is a list of the IP protocol numbers found in the field Protocol of the IPv4 header and the Next Header field of the IPv6 header. It is an identifier for the encapsulated protocol and determines the layout of the data that immediately follows the header. Both fields are eight bits wide.
Secure Electronic Transaction (SET) Supplement for the v1.0 Internet Open Trading Protocol (IOTP) June 2003 Internet Open Trading Protocol: RFC 3550 : RTP: A Transport Protocol for Real-Time Applications: July 2003: RTP: RFC 3711 : The Secure Real-time Transport Protocol (SRTP) March 2004: SRTP: RFC 3720 : Internet Small Computer Systems ...
Long packet times increase latencies and require more buffering. A range from 125 μs to 4 ms is defined, though it is recommended that devices shall adapt to packet time changes and/or determine packet time by analyzing RTP timestamps. Packet time determines RTP payload size according to a supported sample rate. 1 ms is required for all devices.
ZRTP ("Z" is a reference to its inventor, Zimmermann; "RTP" stands for Real-time Transport Protocol) [1] is described in the Internet Draft as a "key agreement protocol which performs Diffie–Hellman key exchange during call setup in-band in the Real-time Transport Protocol (RTP) media stream which has been established using some other signaling protocol such as Session Initiation Protocol (SIP).