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 ).
13-408: Not to be confused with RTSP . RSTP : Rstp, the abbreviation for the orchid genus Restrepia Rapid Spanning Tree Protocol , a computer network protocol Topics referred to by the same term [REDACTED] This disambiguation page lists articles associated with the title RSTP . If an internal link led you here, you may wish to change
26-527: Is stateless , RTSP has a state; an identifier is used when needed to track concurrent sessions. Like HTTP, RTSP uses TCP to maintain an end-to-end connection and, while most RTSP control messages are sent by the client to the server, some commands travel in the other direction (i.e. from server to client). Presented here are the basic RTSP requests. Some typical HTTP requests , like the OPTIONS request, are also available. The default transport layer port number
39-595: Is 554 for both TCP and UDP , the latter being rarely used for the control requests. The ANNOUNCE method serves two purposes: RTSP over HTTP was defined by Apple in 1999 and [1] . It interleaves the RTP Video and Audio data into the RTSP Command Connection (as defined in RFC2326), and then sends the RTSP Command Connection via a pair of HTTP connections, one is a long running GET connection and
52-428: Is a global and open industry forum with the goal of facilitating the development and use of a global open standard for the interface of physical IP-based security products. ONVIF creates a standard for how IP products within video surveillance and other physical security areas can communicate with each other. ONVIF is an organization started in 2008 by Axis Communications , Bosch Security Systems and Sony . It
65-453: The ONVIF member base had grown to 103 members. This comprised 12 full members, 13 contributing members and 78 user members. In December 2010, the forum had more than 240 members and more than 440 conformant products on the market. By January 2015, this had grown to more than 3,700 ONVIF conformant products and 500 members. By August 2016, this had grown to more than 6,900 conformant products on
78-467: The link to point directly to the intended article. Retrieved from " https://en.wikipedia.org/w/index.php?title=RSTP&oldid=852247130 " Category : Disambiguation pages Hidden categories: Short description is different from Wikidata All article disambiguation pages All disambiguation pages Real Time Streaming Protocol RTSP was developed by RealNetworks , Netscape and Columbia University . The first draft
91-415: The market but shrunk to 461 members. In February 2020, ONVIF reached more than 14,000 conformant products. As of June 2022 , there are 496 members and more than 23000 conformant products. ONVIF originally was an acronym for Open Network Video Interface Forum . The longer name was dropped as the scope of the standard expanded beyond video applications. The ONVIF Core Specification aims to standardize
104-540: The network interface (on the network layer ) of network video products. It defines a network video communication framework based on relevant IETF and Web services standards including security and IP configuration requirements. The following areas are covered by the Core Specification version 1.0: ONVIF utilizes IT industry technologies including SOAP , RTP , and Motion JPEG , MPEG-4 , H.264 video codecs and H.265 video codecs . Later releases of
117-550: The other is a long running POST connection. This method is also used in the ONVIF IP Camera standard and can be combined with HTTPS for secure and encrypted video and audio. There are several different methods for encrypting RTSP command messages and the RTP Video and Audio data. RTSP 2.0 (RFC7826) defines several methods for encryption and introduces a new rtsps:// URL and many of these have been incorporated into RFC2326 RTSP 1.0 Clients and Servers. IANA have reserved
130-645: The rtsps:// URL prefix and Port 322 for RTSPS. As of September 2024, RTSP over HTTPS has been implemented in several ONVIF IP Cameras and RTSPS (using the rtsps:// URL) has been implemented by Axis and Bosch CCTV Cameras, FFmpeg , GStreamer , MediaMTX and SharpRTSP. RTSP using RTP and RTCP allows for the implementation of rate adaptation. Many CCTV / Security cameras, often called IP cameras , support RTSP streaming too, especially those with ONVIF profiles G, S, T. ONVIF ONVIF (the Open Network Video Interface Forum )
143-608: Was officially incorporated as a non-profit, 501(c)6 Delaware corporation on November 25, 2008. ONVIF membership is open to manufacturers, software developers, consultants, system integrators, end users and other interest groups that wish to participate in the activities of ONVIF. The ONVIF specification aims to achieve interoperability between network video products regardless of manufacturer. ONVIF concerns itself with standardization of communication between IP-based physical security products to achieve open interoperability between equipment from different manufacturers. In December 2009,
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#1732787329278156-868: Was published as RFC 2326 in 1998. RTSP 2.0 published as RFC 7826 in 2016 as a replacement of RTSP 1.0. RTSP 2.0 is based on RTSP 1.0 but is not backwards compatible other than in the basic version negotiation mechanism, and remains a Proposed Standard. The transmission of streaming data itself is not a task of RTSP. Most RTSP servers use the Real-time Transport Protocol (RTP) in conjunction with Real-time Control Protocol (RTCP) for media stream delivery. However, some vendors implement proprietary transport protocols. The RTSP server software from RealNetworks , for example, also used RealNetworks' proprietary Real Data Transport (RDT). While similar in some ways to HTTP , RTSP defines control sequences useful in controlling multimedia playback. While HTTP
169-760: Was submitted to IETF in October 1996 by Netscape and Progressive Networks , after which Henning Schulzrinne from Columbia University submitted "RTSPÕš" ("RTSP prime") in December 1996. The two drafts were merged for standardization by the Multiparty Multimedia Session Control Working Group (MMUSIC WG) of the Internet Engineering Task Force (IETF) and further drafts were published by the working group. The Proposed Standard for RTSP
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