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The Real-Time Streaming Protocol (RTSP) is an application-level network protocol designed for multiplexing and packetizing multimedia transport streams over a suitable transport protocol. RTSP is used in entertainment and communications systems to control streaming media servers. The protocol is used for establishing and controlling media sessions between endpoints. Clients of media servers issue commands such as play, record and pause, to facilitate real-time control of the media streaming from the server to a client or from a client to the server.
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 Transmission Control Protocol (TCP) is one of the main protocols of the Internet protocol suite. It originated in the initial network implementation in which it complemented the Internet Protocol (IP). Therefore, the entire suite is commonly referred to as TCP/IP. TCP provides reliable, ordered, and error-checked delivery of a stream of octets (bytes) between applications running on hosts communicating via an IP network. Major internet applications such as the World Wide Web, email, remote administration, and file transfer rely on TCP, which is part of the transport layer of the TCP/IP suite. SSL/TLS often runs on top of TCP.
In computer networking, the User Datagram Protocol (UDP) is one of the core communication protocols of the Internet protocol suite used to send messages to other hosts on an Internet Protocol (IP) network. Within an IP network, UDP does not require prior communication to set up communication channels or data paths.
Voice over Internet Protocol (VoIP), also known as IP telephony, refers to a set of technologies used for voice communication sessions over Internet Protocol (IP) networks, such as the Internet. VoIP enables voice calls to be transmitted as data packets, facilitating various methods of voice communication, including traditional applications like Skype, Microsoft Teams, Google Voice, and VoIP phones. Regular telephones can also be used for VoIP by connecting them to the Internet via analog telephone adapters (ATAs), which convert traditional telephone signals into digital data packets that can be transmitted over IP networks.
In computer networking, the transport layer is a conceptual division of methods in the layered architecture of protocols in the network stack in the Internet protocol suite and the OSI model. The protocols of this layer provide end-to-end communication services for applications. It provides services such as connection-oriented communication, reliability, flow control, and multiplexing.
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.
Retransmission, essentially identical with automatic repeat request (ARQ), is the resending of packets which have been either damaged or lost. Retransmission is one of the basic mechanisms used by protocols operating over a packet switched computer network to provide reliable communication.
Packet loss occurs when one or more packets of data travelling across a computer network fail to reach their destination. Packet loss is either caused by errors in data transmission, typically across wireless networks, or network congestion. Packet loss is measured as a percentage of packets lost with respect to packets sent.
ZRTP is a cryptographic key-agreement protocol to negotiate the keys for encryption between two end points in a Voice over IP (VoIP) phone telephony call based on the Real-time Transport Protocol. It uses Diffie–Hellman key exchange and the Secure Real-time Transport Protocol (SRTP) for encryption. ZRTP was developed by Phil Zimmermann, with help from Bryce Wilcox-O'Hearn, Colin Plumb, Jon Callas and Alan Johnston and was submitted to the Internet Engineering Task Force (IETF) by Zimmermann, Callas and Johnston on March 5, 2006 and published on April 11, 2011 as RFC 6189.
Video Share is an IP Multimedia System (IMS) enabled service for mobile networks that allows users engaged in a circuit switch voice call to add a unidirectional video streaming session over the packet network during the voice call. Any of the parties on the voice call can initiate a video streaming session. There can be multiple video streaming sessions during a voice call, and each of these streaming sessions can be initiated by any of the parties on the voice call. The video source can either be the camera on the phone or a pre-recorded video clip.
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.
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.
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.
Secure Reliable Transport (SRT) is an open source video transport protocol that utilises the UDP transport protocol. The SRT Protocol specification is available as an Internet Draft from the IETF.
SMPTE 2110 is a suite of standards from the Society of Motion Picture and Television Engineers (SMPTE) that describes how to send digital media over an IP network.
Audio Video Bridging (AVB) is a common name for a set of technical standards that provide improved synchronization, low latency, and reliability for switched Ethernet networks. AVB embodies the following technologies and standards:
References
- ↑ "RIST Activity Group". Video Services Forum. Retrieved 29 March 2019.
- ↑ Noronha, Ciro. "A Study of Protocols for Low-Latency Video Transport over the Internet" (PDF). Retrieved April 20, 2020.
- ↑ "Video Services Forum Technical Recommendation TR-06-1" (PDF). Video Services Forum . Retrieved 4 April 2019.
- ↑ "Video Services Forum Technical Recommendation TR-06-2" (PDF). Video Services Forum . Retrieved April 20, 2020.