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OBS Studio is a free and open-source app for screencasting and live streaming.Written in C/C++ and built with Qt, OBS Studio provides real-time capture, scene composition, recording, encoding, and broadcasting via Real-Time Messaging Protocol (RTMP), HLS, SRT, RIST or WebRTC.
The RTMP protocol has multiple variations: RTMP proper, the "plain" protocol which works on top of Transmission Control Protocol (TCP) and uses port number 1935 by default. RTMPS, which is RTMP over a Transport Layer Security (TLS/SSL) connection. RTMPE, which is RTMP encrypted using Adobe's own security mechanism.
HTTP Live Streaming (also known as HLS) 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.
Helix Universal Server – delivers MPEG-DASH, RTSP, HTTP Live Streaming (HLS), RTMP; developed by RealNetworks, discontinued since October 2014 [3] [4] HelixCommunity – RealNetworks Open Source development community; Jellyfin – GPL-licensed fully open-source fork of Emby; Icecast – GPL streaming media server
RTMFP is based on User Datagram Protocol (UDP), [1] whereas Real-Time Messaging Protocol (RTMP) is based on Transmission Control Protocol (TCP). UDP‐based protocols have some specific advantages over TCP‐based protocols when delivering live streaming media , such as decreased latency and overhead, and greater tolerance for dropped or ...
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.
This class of status code indicates the client must take additional action to complete the request. Many of these status codes are used in URL redirection. [2]A user agent may carry out the additional action with no user interaction only if the method used in the second request is GET or HEAD.
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.