|
RTSP
The Real Time Streaming Protocol (RTSP) is an application-level network protocol designed for multiplexing and packetizing multimedia transport streams (such as interactive media, video and audio) 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 (video on demand) or from a client to the server (voice recording). History RTSP was developed by RealNetworks, Netscape and Columbia University. The first draft was submitted to IETF in October 1996 by Netscape and Progressive Networks, after which Henning Schulzrinne from Columbia University submitted "RTSP'" ("RTSP prime") in December 1996. The two drafts were merged for standardization by the Multipar ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Streaming Media
Streaming media is multimedia that is delivered and consumed in a continuous manner from a source, with little or no intermediate storage in network elements. ''Streaming'' refers to the delivery method of content, rather than the content itself. Distinguishing delivery method from the media applies specifically to telecommunications networks, as most of the traditional media delivery systems are either inherently ''streaming'' (e.g. radio, television) or inherently ''non-streaming'' (e.g. books, videotape, audio CDs). There are challenges with streaming content on the Internet. For example, users whose Internet connection lacks sufficient bandwidth may experience stops, lags, or poor buffering of the content, and users lacking compatible hardware or software systems may be unable to stream certain content. With the use of buffering of the content for just a few seconds in advance of playback, the quality can be much improved. Livestreaming is the real-time delivery of co ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Real-time Transport Protocol
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. RTP typically runs over User Datagram Protocol (UDP). RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. RTP is one of the technical foundations of Voice over IP and in this context is often used in conjunction with a signaling protocol such as the Session Initiation Protocol (SIP) which establishes connections across the network. RTP was developed by the Audio-Video Transport Working Group of the Internet Engineering Task Force (IETF) and first published in 1996 a ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Darwin Streaming Server
Darwin Streaming Server (DSS) was the first open sourced RTP/RTSP streaming server. It was released March 16, 1999 and is a fully featured RTSP/ RTP media streaming server capable of streaming a variety of media types including H.264/MPEG-4 AVC, MPEG-4 Part 2 and 3GP. Development Developed by Apple, it is the open source equivalent of QuickTime Streaming Server, and is based on its code. Ports The initial DSS source code release compiled only on OS X, but external developers quickly ported the code to Linux, FreeBSD, Solaris, Tru64 Unix, Mac OS 9 and Windows. Source code is available as a release download or as development code via CVS. See also *HTTP Live Streaming – Apple's video/audio streaming server protocol *Helix Universal Server – Multiformat streaming server from RealNetworks *Wowza Media Server Wowza Streaming Engine (known as Wowza Media Server prior to version 4) is a unified streaming media server software developed by Wowza. The server is u ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Session Description Protocol
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''. SDP is extensible for the support of new media types and formats. SDP was originally a component of the Session Announcement Protocol (SAP), but found other uses in conjunction with the Real-time Transport Protocol (RTP), the Real-time Streaming Protocol (RTSP), Session Initiation Protocol (SIP), and as a standalone protocol for describing multicast sessions. The IETF published the original specification as a Proposed Standard in April 1998. Revised specifications were released in 2006 (RFC ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Application Layer
An application layer is an abstraction layer that specifies the shared communications protocols and Interface (computing), interface methods used by Host (network), 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. Internet protocol suite In the Internet protocol suite, the application layer contains the communications protocols and interface methods used in process-to-process communications across an Internet Protocol (IP) computer network. The application layer only standardizes communication and depends upon the underlying transport layer protocols to establish host-to-host data transfer channels and manage the data exchange in a client–server model, client–server or peer-to-peer networking model. Though the TCP/IP application layer does not d ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Real Data Transport
Real Data Transport (RDT) is a proprietary transport protocol for the actual audio-video data, developed by RealNetworks in the 1990s. It is commonly used in companion with a control protocol for streaming media like the IETF's Real Time Streaming Protocol (RTSP). A non-proprietary alternative for RDT is IETF's Real-time Transport Protocol (RTP), which is also implemented in RealNetworks players. As reported in a 2002 book about firewalls, RDT used two unidirectional UDP connections, one for the data sent from the server to the client and another in the opposite direction for retransmission requests. The same book reported that RealNetworks' G2 server used RDT in this configuration by default. Another 2003 book reported that RDT was also seen carried over Transmission Control Protocol (TCP), as a fall-back mechanism. RDT is now included as part of the Helix Community project. References External links Helix Community RDT implementation C++ source code repository. RDT v2protoco ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Interactive Media
Interactive media normally refers to products and services on digital computer-based systems which respond to the user's actions by presenting content such as text, moving image, animation, video and audio. Since its early conception, various forms of interactive media have emerged with impacts on educational and commercial markets. With the rise of decision-driven media, concerns surround the impacts of cybersecurity and societal distraction. Definition Interactive media is a method of communication in which the output from the media comes from the input of the users. Interactive media works with the user's participation. The media still has the same purpose but the user's input adds interaction and brings interesting features to the system for better enjoyment. Development The analogue videodisc developed by NV Philips was the pioneering technology for interactive media. Additionally, there are several elements that encouraged the development of interactive media including the ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Proposed Standard
In computer network engineering, an Internet Standard is a normative specification of a technology or methodology applicable to the Internet. Internet Standards are created and published by the Internet Engineering Task Force (IETF). They allow interoperation of hardware and software from different sources which allows internets to function. As the Internet became global, Internet Standards became the lingua franca of worldwide communications. Engineering contributions to the IETF start as an Internet Draft, may be promoted to a Request for Comments, and may eventually become an Internet Standard. An Internet Standard is characterized by technical maturity and usefulness. The IETF also defines a Proposed Standard as a less mature but stable and well-reviewed specification. A Draft Standard was an intermediate level, discontinued in 2011. A Draft Standard was an intermediary step that occurred after a Proposed Standard but prior to an Internet Standard. As put in RFC 2026: In ge ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Multiplexing
In telecommunications and computer networking, multiplexing (sometimes contracted to muxing) is a method by which multiple analog or digital signals are combined into one signal over a shared medium. The aim is to share a scarce resource - a physical transmission medium. For example, in telecommunications, several telephone calls may be carried using one wire. Multiplexing originated in telegraphy in the 1870s, and is now widely applied in communications. In telephony, George Owen Squier is credited with the development of telephone carrier multiplexing in 1910. The multiplexed signal is transmitted over a communication channel such as a cable. The multiplexing divides the capacity of the communication channel into several logical channels, one for each message signal or data stream to be transferred. A reverse process, known as demultiplexing, extracts the original channels on the receiver end. A device that performs the multiplexing is called a multiplexer (MUX), and a dev ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Uniform Resource Locator
A Uniform Resource Locator (URL), colloquially termed as a web address, is a reference to a web resource that specifies its location on a computer network and a mechanism for retrieving it. A URL is a specific type of Uniform Resource Identifier (URI), although many people use the two terms interchangeably. URLs occur most commonly to reference web pages (HTTP) but are also used for file transfer (FTP), email (mailto), database access (JDBC), and many other applications. Most web browsers display the URL of a web page above the page in an address bar. A typical URL could have the form http://www.example.com/index.html, which indicates a protocol (http), a hostname (www.example.com), and a file name (index.html). History Uniform Resource Locators were defined in in 1994 by Tim Berners-Lee, the inventor of the World Wide Web, and the URI working group of the Internet Engineering Task Force (IETF), as an outcome of collaboration started at the IETF Living Documents birds of a ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
User Datagram Protocol
In computer networking, the User Datagram Protocol (UDP) is one of the core communication protocols of the Internet protocol suite used to send messages (transported as datagrams in packets) 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. UDP uses a simple connectionless communication model with a minimum of protocol mechanisms. UDP provides checksums for data integrity, and port numbers for addressing different functions at the source and destination of the datagram. It has no handshaking dialogues, and thus exposes the user's program to any unreliability of the underlying network; there is no guarantee of delivery, ordering, or duplicate protection. If error-correction facilities are needed at the network interface level, an application may instead use Transmission Control Protocol (TCP) or Stream Control Transmission Protocol (SCTP) which are designed for this ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Transmission Control Protocol
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. TCP is connection-oriented, and a connection between client and server is established before data can be sent. The server must be listening (passive open) for connection requests from clients before a connection is established. Three-way handshake (active open), retransmission, and error detection adds to reliability but lengthens latency. Applica ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
.jpg "Click for more on -> Interactive Media")
