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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 "RTSP'" ("RTSP prime") in December 1996. The two drafts were merged for standardization by the Multipar ... [...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] |
Henning Schulzrinne
Henning Schulzrinne is a German-American computer scientist, who led research and development of the voice over IP network protocols. Life Schulzrinne studied engineering management at the Department of Electrical Engineering and Information Technology of the German Technische Universität Darmstadt in Darmstadt, where he earned his ''Vordiplom'' (cf. Diplom), then went on to earn his M.Sc. at the University of Cincinnati and his Ph.D. at the University of Massachusetts Amherst. From 1992 to 1994 he worked for AT&T Bell Laboratories. From 1994 to 1996 he worked in Berlin at the ''Forschungs-Institut für Offene Kommunikationssysteme (GMD FOKUS)'', an institute of the now-defunct Gesellschaft für Mathematik und Datenverarbeitung (GMD) which became part of the Fraunhofer Society as Fraunhofer Institute for Open Communication Systems. He joined the faculty of the Computer Science department at Columbia University in 1998, and served as chair and Julian Clarence Levi Professor. H ... [...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] |
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] |
Port Number
In computer networking, a port is a number assigned to uniquely identify a connection endpoint and to direct data to a specific service. 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 identified for each transport protocol and address combination by a 16-bit unsigned number, known as the port number. The most common transport protocols that use port numbers are the Transmission Control Protocol (TCP) and the User Datagram Protocol (UDP). A port number is always associated with an IP address of a host and the type of transport protocol used for communication. It completes the destination or origination network address of a message. 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 a ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
HTTP Request
The Hypertext Transfer Protocol (HTTP) 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 protocol version that was named 0.9. That first version of HTTP protocol soon evolved into a more elaborated version that was the first draft toward a far future version 1.0. Development of early HTTP Requests for Comments (RFCs) started a few years later and it was a coordinated effort by the Internet Engineering Task Force (IETF) and the World Wide Web Consortium (W3C), with work later moving to t ... [...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
The Hypertext Transfer Protocol (HTTP) 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 protocol version that was named 0.9. That first version of HTTP protocol soon evolved into a more elaborated version that was the first draft toward a far future version 1.0. Development of early HTTP Requests for Comments (RFCs) started a few years later and it was a coordinated effort by the Internet Engineering Task Force (IETF) and the World Wide Web Consortium (W3C), with work later moving to ... [...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] |
RTCP
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. 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 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
