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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 communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony, in private IP telephone systems, 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 announcement and 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.
Simple Network Management Protocol (SNMP) is an Internet Standard protocol for collecting and organizing information about managed devices on IP networks and for modifying that information to change device behavior. Devices that typically support SNMP include cable modems, routers, switches, servers, workstations, printers, and more.
An application layer is an abstraction layer that specifies the shared communication protocols and interface methods used by hosts in a communications network. An application layer abstraction is specified in both 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 on the Internet Protocol (IP) 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. UPnP is intended primarily for residential networks without enterprise-class devices.
Extensible Messaging and Presence Protocol is an open communication protocol designed for instant messaging (IM), presence information, and contact list maintenance. Based on XML, it enables the near-real-time exchange of structured data between two or more network entities. Designed to be extensible, the protocol offers a multitude of applications beyond traditional IM in the broader realm of message-oriented middleware, including signalling for VoIP, video, file transfer, gaming and other uses.
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 binary-encoded out-of-band signaling protocol that functions alongside the Real-time Transport Protocol (RTP). Its basic functionality and packet structure is defined in RFC 3550. RTCP provides 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.
Architecture for Control Networks (ACN) is a suite of network protocols for control of entertainment technology equipment, particularly as used in live performance or large-scale installations. For example, lighting, audio or special effects equipment. ACN is maintained by Entertainment Services and Technology Association and its first official release was ANSI Standard E1.17-2006 - Entertainment Technology - Architecture for Control Networks. The standard was subsequently revised and released as ANSI E1.17-2010.
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.
The Microsoft Open Specification Promise is a promise by Microsoft, published in September 2006, to not assert its patents, in certain conditions, against implementations of a certain list of specifications.
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.
RTP-MIDI is a protocol to transport MIDI messages within Real-time Transport Protocol (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.
NACK-Oriented Reliable Multicast (NORM) is a transport layer Internet protocol designed to provide reliable transport in multicast groups in data networks. It is formally defined by the Internet Engineering Task Force (IETF) in Request for Comments (RFC) 5740, which was published in November 2009.
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, Wheatnet, 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 multiple participants in the IP Multimedia Subsystem (IMS). It is therefore a key element in the IMS framework.
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.