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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 Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications. SIP is used for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, in instant messaging over Internet Protocol (IP) networks as well as mobile phone calling over LTE (VoLTE).
The Session Description Protocol (SDP) is a format for describing multimedia communication sessions for the purposes of session announcement and session invitation. Its predominant use is in support of streaming media applications, such as voice over IP (VoIP) and video conferencing. SDP does not deliver any media streams itself but is used between endpoints for negotiation of network metrics, media types, and other associated properties. The set of properties and parameters is called a session profile.
In computer networking, the User Datagram Protocol (UDP) is one of the core members of the Internet protocol suite. The protocol was designed by David P. Reed in 1980 and formally defined in RFC 768. With UDP, computer applications can send messages, in this case referred to as datagrams, to other hosts on an Internet Protocol (IP) network. Prior communications are not required in order to set up communication channels or data paths.
In the seven-layer OSI model of computer networking, the session layer is layer 5.
An application layer is an abstraction layer that specifies the shared communications protocols and interface methods used by hosts in a communications network. The application layer abstraction is used in both of the standard models of computer networking: the Internet Protocol Suite (TCP/IP) and the OSI model. Although both models use the same term for their respective highest level layer, the detailed definitions and purposes are different.
Universal Plug and Play (UPnP) is a set of networking protocols that permits networked devices, such as personal computers, printers, Internet gateways, Wi-Fi access points and mobile devices to seamlessly discover each other's presence on the network and establish functional network services for data sharing, communications, and entertainment. UPnP is intended primarily for residential networks without enterprise-class devices.
The Resource Reservation Protocol (RSVP) is a transport layer protocol designed to reserve resources across a network using the integrated services model. RSVP operates over an IPv4 or IPv6 and provides receiver-initiated setup of resource reservations for multicast or unicast data flows. It does not transport application data but is similar to a control protocol, like Internet Control Message Protocol (ICMP) or Internet Group Management Protocol (IGMP). RSVP is described in RFC 2205.
The RTP Control Protocol (RTCP) is a sister protocol of the Real-time Transport Protocol (RTP). Its basic functionality and packet structure is defined in RFC 3550. RTCP provides out-of-band statistics and control information for an RTP session. It partners with RTP in the delivery and packaging of multimedia data, but does not transport any media data itself.
IP multicast is a method of sending Internet Protocol (IP) datagrams to a group of interested receivers in a single transmission. It is the IP-specific form of multicast and is used for streaming media and other network applications. It uses specially reserved multicast address blocks in IPv4 and IPv6.
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. It was developed by a small team of Internet Protocol and cryptographic experts from Cisco and Ericsson. It was first published by the IETF in March 2004 as RFC 3711.
Multimedia Internet KEYing (MIKEY) is a key management protocol that is intended for use with real-time applications. It can specifically be used to set up encryption keys for multimedia sessions that are secured using SRTP, the security protocol commonly used for securing real-time communications such as VoIP.
SDES for Media Streams is a way to negotiate the key for Secure Real-time Transport Protocol. It has been proposed for standardization to the IETF in July 2006
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 format parameters of the RTP payload are typically communicated between transmission endpoints with the Session Description Protocol (SDP), but other protocols, such as the Extensible Messaging and Presence Protocol (XMPP) may be used.
In computer networking, the Message Session Relay Protocol (MSRP) is a protocol for transmitting a series of related instant messages in the context of a communications session. An application instantiates the session with the Session Description Protocol (SDP) over Session Initiation Protocol (SIP) or other rendezvous methods.
Ravenna is a technology for real-time transport of audio and other media data over IP networks. Ravenna was introduced on September 10, 2010 at the International Broadcasting Convention in Amsterdam. Ravenna can operate on most existing network infrastructures using standard networking technology. Performance and capacity scale with network performance. Ravenna is designed to match broadcasters' requirements for low latency, full signal transparency and high reliability. Fields of application include in-house signal distribution for broadcasting houses and other fixed installations, flexible setups at venues and live events, outside broadcasting support, and inter-studio links across wide area network links and production facilities.
RTP-MIDI is a protocol to transport MIDI messages within RTP packets over Ethernet and WiFi networks. It is completely open and free, and is compatible both with LAN and WAN application fields. Compared to MIDI 1.0, RTP-MIDI includes new features like session management, device synchronization and detection of lost packets, with automatic regeneration of lost data. RTP-MIDI is compatible with real-time applications, and supports sample-accurate synchronization for each MIDI message.
AES67 is a technical standard for audio over IP and audio over Ethernet (AoE) interoperability. The standard was developed by the Audio Engineering Society and first published in September 2013. It is a layer 3 protocol suite based on existing standards and is designed to allow interoperability between various IP-based audio networking systems such as RAVENNA, Livewire, Q-LAN and Dante.
The Session Initiation Protocol (SIP) is the signaling protocol selected by the 3rd Generation Partnership Project (3GPP) to create and control multimedia sessions with two or more participants in the IP Multimedia Subsystem (IMS), and therefore is a key element in the IMS framework.
Audio Video Bridging (AVB) is a common name for the set of technical standards which provide improved synchronization, low-latency, and reliability for switched Ethernet networks. AVB embodies the following technologies and standards:
References
- ↑ "SAP (v1 & v2): Session Announcement Protocol". Archived from the original on 2013-01-26. Retrieved 2012-04-06.
- ↑ M. Handley; C. Perkins; E. Whelan (October 2000). Session Announcement Protocol. RFC 2974 .
- ↑ Viewing with Session Announcement Protocol (SAP) , retrieved 2019-03-02
- ↑ "AES67-2013: AES standard for audio applications of networks - High-performance streaming audio-over-IP interoperability". Audio Engineering Society. 2013-09-11. Retrieved 2014-02-11.