The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications. SIP is used for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, in instant messaging over Internet Protocol (IP) networks as well as mobile phone calling over LTE (VoLTE).
IBM Sametime is a client–server application and middleware platform that provides real-time, unified communications and collaboration for enterprises. Those capabilities include presence information, enterprise instant messaging, web conferencing, community collaboration, and telephony capabilities and integration. It is sold by the Lotus Software division of IBM.
Real-Time Messaging Protocol (RTMP) was initially a proprietary protocol developed by Macromedia for streaming audio, video and data over the Internet, between a Flash player and a server. Macromedia is now owned by Adobe, which has released an incomplete version of the specification of the protocol for public use.
A media server refers either to a dedicated computer appliance or to a specialized application software, ranging from an enterprise class machine providing video on demand, to, more commonly, a small personal computer or NAS for the home, dedicated for storing various digital media. This can also mean that these servers are specialized for media for streaming
Adobe Media Server (AMS) is a proprietary data and media server from Adobe Systems. This server works with the Flash Player runtime to create media driven, multiuser RIAs. The server uses ActionScript 1, an ECMAScript based scripting language, for server-side logic. Prior to version 2, it was known as Flash Communication Server. Prior to version 5, it was known as Flash Media Server. In February 2019, Adobe Systems Incorporated granted Veriskope LLC rights to further develop, resell, and extend distribution of the software product.
FreeSWITCH is a free and open-source application server for real-time communication, WebRTC, telecommunications, video and Voice over Internet Protocol (VoIP). Multiplatform, it runs on Linux, Windows, macOS and FreeBSD. It is used to build PBX systems, IVR services, videoconferencing with chat and screen sharing, wholesale least-cost routing, Session Border Controller (SBC) and embedded communication appliances. It has full support for encryption, ZRTP, DTLS, SIPS. It can act as a gateway between PSTN, SIP, WebRTC, and many other communication protocols. Its core library, libfreeswitch, can be embedded into other projects. It is licensed under the Mozilla Public License (MPL), a free software license.
Jitsi is a collection of free and open-source multiplatform voice (VoIP), videoconferencing and instant messaging applications for the web platform, Windows, Linux, Mac OS X and Android. The Jitsi project began with the Jitsi Desktop. With the growth of WebRTC, the project team focus shifted to the Jitsi Video Bridge for allowing web-based multi-party video calling. Later the team added Jitsi Meet, a full video conferencing application that includes a web, Android, and iOS clients. Jitsi also operates meet.jit.si, a version of Jitsi Meet its hosts for free community use. Other projects include; Jigasi, lib-jitsi-meet, Jidesha, and Jitsi.
The Secure Real-Time Media Flow Protocol (RTMFP) is a protocol suite developed by Adobe Systems for encrypted, efficient multimedia delivery through both client-server and peer-to-peer models over the Internet. The protocol was originally proprietary, but was later opened up and is now published as RFC 7016
HTTP Live Streaming is an HTTP-based adaptive bitrate streaming communications protocol implemented by Apple Inc. as part of its QuickTime, Safari, OS X, and iOS software. Client implementations are also available in Microsoft Edge, Firefox and some versions of Google Chrome. Support is widespread in streaming media servers.
The HTML5 specification introduced the video element for the purpose of playing videos, partially replacing the object element. HTML5 video is intended by its creators to become the new standard way to show video on the web, instead of the previous de facto standard of using the proprietary Adobe Flash plugin, though early adoption was hampered by lack of agreement as to which video coding formats and audio coding formats should be supported in web browsers.
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 Media Systems. 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.
Unreal Media Server is a streaming server software created by Unreal Streaming Technologies.
Media Source Extensions (MSE) is a W3C specification that allows JavaScript to send byte streams to media codecs within Web browsers that support HTML 5 video and audio. Among other possible uses, this allows the implementation of client-side prefetching and buffering code for streaming media entirely in JavaScript. It is compatible with, but should not be confused with, the Encrypted Media Extensions specification, and neither requires the use of the other.
JsSIP is a library for the programming language JavaScript. It takes advantage of SIP and WebRTC to provide a fully featured SIP endpoint in any website. JsSIP allows any website to get real-time communication features using audio and video. It makes it possible to build SIP user agents that send and receive audio and video calls as well as and text messages.
WebRTC Gateway connects between WebRTC and an established VoIP technology such as SIP. WebRTC is an API definition drafted by the World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat, and messaging without the need of either internal or external plugins.
