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Feng is a multimedia streaming server compliant with IETF standards for real-time streaming of multimedia content over IP networks. Feng implements Real Time Streaming Protocol (RTSP, RFC 2326) and Real-time Transport Protocol / RTP Control Protocol (RTP/RTCP, RFC 3550). It supports the RTP Profile for Audio and Video Conferences with Minimal Control ( RFC 3551). [1]
Feng is part of the LScube project, [2] supported by the Internet Media Group at the Politecnico di Torino. It is released under the LGPL v2.1.
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.
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RealAudio, or also spelled as Real Audio is a proprietary audio format developed by RealNetworks and first released in April 1995. It uses a variety of audio codecs, ranging from low-bitrate formats that can be used over dialup modems, to high-fidelity formats for music. It can also be used as a streaming audio format, that is played at the same time as it is downloaded. In the past, many internet radio stations used RealAudio to stream their programming over the internet in real time. In recent years, however, the format has become less common and has given way to more popular audio formats. RealAudio was heavily used by the BBC websites until 2009, though it was discontinued due to its declining use. BBC World Service, the last of the BBC websites to use RealAudio, discontinued its use in March 2011.
Helix DNA was a project to produce computer software that can play audio and video media in various formats and aid in creating such media. It is intended as a largely free and open-source digital media framework that runs on numerous operating systems and processors and it was started by RealNetworks, which contributed much of the code. The Helix Community was an open collaborative effort to develop and extend the Helix DNA platform. The Helix Project has been discontinued.
Digital storage media command and control (DSM-CC) is a toolkit for developing control channels associated with MPEG-1 and MPEG-2 streams. It is defined in part 6 of the MPEG-2 standard and uses a client/server model connected via an underlying network.
Media Resource Control Protocol (MRCP) is a communication protocol used by speech servers to provide various services to their clients. MRCP relies on another protocol, such as Real Time Streaming Protocol (RTSP) or Session Initiation Protocol (SIP) for establishing a control session and audio streams between the client and the server.
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.
QuickTime Broadcaster is an audio and video RTP/RTSP server by Apple Inc. for Mac OS X. It is separate from Apple's QuickTime Streaming Server, as it is not a service daemon but a desktop application. It is able to stream live video and audio over a network in any QuickTime supported streaming codec.
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.
GPAC Project on Advanced Content is an implementation of the MPEG-4 Systems standard written in ANSI C. GPAC provides tools for media playback, vector graphics and 3D rendering, MPEG-4 authoring and distribution.
LIVE555 Streaming Media is a set of open source (LGPL) C++ libraries developed by Live Networks, Inc. for multimedia streaming. The libraries support open standards such as RTP/RTCP and RTSP for streaming, and can also manage video RTP payload formats such as H.264, H.265, MPEG, VP8, and DV, and audio RTP payload formats such as MPEG, AAC, AMR, AC-3 and Vorbis. It is used internally by well-known software such as VLC and mplayer.
HTTP Live Streaming is an HTTP-based adaptive bitrate streaming communications protocol developed by Apple Inc. and released in 2009. Support for the protocol is widespread in media players, web browsers, mobile devices, and streaming media servers. As of 2022, an annual video industry survey has consistently found it to be the most popular streaming format.
The Helix Universal Media Server was a product developed by RealNetworks and originates from the first streaming media server originally developed by Progressive Networks in 1994. It supported a variety of streaming media delivery transports including MPEG-DASH RTMP (flash), RTSP (standard), HTTP Live Streaming (HLS), Microsoft Silverlight and HTTP Progressive Download enabling mobile phone OS and PC OS media client delivery.
Sirannon is a free, open-source, media server and client. The goal is to aid in video research and experimental streaming. Sirannon allows the programmer to create a wide variety of media-handling components such as streaming, reading, writing, packetizing. By organizing these components in a workflow the programmer can create many applications such as a media server, media proxy or video tool. Sirannon was introduced at the ACM multimedia conference in October 2009 under its former name xStreamer.
Wowza Streaming Engine is a unified streaming media server software developed by Wowza. The server is used for streaming of live and on-demand video, audio, and rich Internet applications over IP networks to desktop, laptop, and tablet computers, mobile devices, IPTV set-top boxes, internet-connected TV sets, game consoles, and other network-connected devices. The server is a Java application deployable on most operating systems.
InSoft Inc. was an American developer of network-based communications software and multimedia software founded in 1992 in Mechanicsburg, Pennsylvania. The company’s applications included Communique, which is used for desktop collaboration and videoconferencing.
Web Call Server is unified intermedia server software developed by Flashphoner. It is a server-side platform, implemented in Java, dedicated for streaming video over wide range of communication protocols, including:
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