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Real Time Streaming Protocol
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 "RTSPÕ" ("RTSP prime") in December 1996. The two drafts were merged for standardization by ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Internet Streaming
Streaming media refers to multimedia delivered through a network for playback using a media player. Media is transferred in a ''stream'' of packets from a server to a client and is rendered in real-time; this contrasts with file downloading, a process in which the end-user obtains an entire media file before consuming the content. Streaming is more commonly used for video on demand, streaming television, and music streaming services over the Internet. While streaming is most commonly associated with multimedia from a remote server over the Internet, it also includes offline multimedia between devices on a local area network. For example, using DLNA and a home server, or in a personal area network between two devices using Bluetooth (which uses radio waves rather than IP). Online streaming was initially popularized by RealNetworks and Microsoft in the 1990s and has since grown to become the globally most popular method for consuming music and videos, with nume ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Progressive Networks
RealNetworks LLC is an American technology company and provider of Internet streaming media delivery software and services based in Seattle, Washington. The company also provides subscription-based online entertainment services and mobile entertainment and messaging services. History RealNetworks (then known as Progressive Networks) was founded in 1994 by Rob Glaser, an ex-Microsoft executive, and a management team including Phil Barrett, Andy Sharpless, and Stephen Buerkle. The original goal of the company was to provide a distribution channel for politically progressive content. It evolved into a technology venture to leverage the Internet as an alternative distribution medium for audio broadcasts. Progressive Networks became RealNetworks in September 1997, in advance of the company's initial public offering (IPO) in October 1997 when shares of the company started trading on Nasdaq as "RNWK". RealNetworks were pioneers in the streaming media markets and broadcast one of th ... [...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 (RFC 2327). Revised specifications were rele ... [...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 Network packet, 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 is a connectionless protocol, meaning that messages are sent without negotiating a connection and that UDP does not keep track of what it has sent. UDP provides checksums for data integrity, and port numbers for addressing different functions at the source and destination of the datagram. It has no Handshake (computing), handshaking dialogues and thus exposes the user's program to any Reliability (computer networking), 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 applica ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Transmission Control Protocol
The Transmission Control Protocol (TCP) is one of the main communications protocol, 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 reliability (computer networking), reliable, ordered, and error detection and correction, error-checked delivery of a reliable byte stream, stream of octet (computing), 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. Transport Layer Security, SSL/TLS often runs on top of TCP. TCP is Connection-oriented communication, connection-oriented, meaning that sender and receiver firstly need to establish a connection based on agreed parameters; they do this through three-way Ha ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Port Number
In computer networking, a port is a communication endpoint. At the software level within an operating system, a port is a logical construct that identifies a specific process or a type of network service. A port is uniquely identified by a number, the port number, associated with the combination of a transport protocol and the network IP address. Port numbers are 16-bit unsigned integers. The most common transport protocols that use port numbers are the Transmission Control Protocol (TCP) and the User Datagram Protocol (UDP). The port completes the destination and origination addresses of a message within a host to point to an operating system process. Specific port numbers are reserved to identify specific services so that an arriving packet can be easily forwarded to a running application. For this purpose, port numbers lower than 1024 identify the historically most commonly used services and are called the well-known port numbers. Higher-numbered ports are available for g ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
HTTP Request
HTTP (Hypertext Transfer Protocol) is an application layer protocol in the Internet protocol suite model for distributed, collaborative, hypermedia information systems. HTTP is the foundation of data communication for the World Wide Web, where hypertext documents include hyperlinks to other resources that the user can easily access, for example by a mouse click or by tapping the screen in a web browser. Development of HTTP was initiated by Tim Berners-Lee at CERN in 1989 and summarized in a simple document describing the behavior of a client and a server using the first HTTP version, named 0.9. That version was subsequently developed, eventually becoming the public 1.0. Development of early HTTP Requests for Comments (RFCs) started a few years later in a coordinated effort by the Internet Engineering Task Force (IETF) and the World Wide Web Consortium (W3C), with work later moving to the IETF. HTTP/1 was finalized and fully documented (as version 1.0) in 1996. It evolved ( ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Stateless Server
A stateless protocol is a communication protocol in which the receiver must not retain session state from previous requests. The sender transfers relevant session state to the receiver in such a way that every request can be understood in isolation, that is without reference to session state from previous requests retained by the receiver. In contrast, a stateful protocol is a communication protocol in which the receiver may retain session state from previous requests. In computer networks, examples of stateless protocols include the Internet Protocol (IP), which is the foundation for the Internet, and the Hypertext Transfer Protocol (HTTP), which is the foundation of the World Wide Web. Examples of stateful protocols include the Transmission Control Protocol (TCP) and the File Transfer Protocol (FTP). Stateless protocols improve the properties of visibility, reliability, and scalability. Visibility is improved because a monitoring system does not have to look beyond a single req ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
HTTP
HTTP (Hypertext Transfer Protocol) is an application layer protocol in the Internet protocol suite model for distributed, collaborative, hypermedia information systems. HTTP is the foundation of data communication for the World Wide Web, where hypertext documents include hyperlinks to other resources that the user can easily access, for example by a Computer mouse, mouse click or by tapping the screen in a web browser. Development of HTTP was initiated by Tim Berners-Lee at CERN in 1989 and summarized in a simple document describing the behavior of a client and a server using the first HTTP version, named 0.9. That version was subsequently developed, eventually becoming the public 1.0. Development of early HTTP Requests for Comments (RFCs) started a few years later in a coordinated effort by the Internet Engineering Task Force (IETF) and the World Wide Web Consortium (W3C), with work later moving to the IETF. HTTP/1 was finalized and fully documented (as version 1.0) in 1996 ... [...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 The Transmission Control Protocol (TCP) is one of the main communications protocol, protocols of the Internet protocol suite. It originated in the initial network implementation in which it co ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
RTCP
The RTP Control Protocol (RTCP) is a binary-encoded out-of-band signaling protocol that functions alongside the Real-time Transport Protocol (RTP). 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. The primary function of RTCP is to provide feedback on the quality of service (QoS) in media distribution by periodically sending statistics information such as transmitted octet and packet counts, packet loss, packet delay variation, and round-trip delay time to participants in a streaming multimedia session. An application may use this information to control quality of service parameters, perhaps by limiting flow, or using a different codec. Protocol functions Typically RTP will be sent on an even-numbered UDP port, with RTCP messages being sent over the next higher odd-numbered port. RTCP itself does not provide any flow encryption or auth ... [...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 i ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
