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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.
Supports HLS and Low Latency HLS in standalone and cluster modes. It can ingest WebRTC, RTMP, RTSP and can create HLS and Low Latency HLS playback endpoints: 2.11 [58] Ant Media No HLS is a out of the box feature in Community and Enterprise Editions. Low Latency HLS is a plugin that is compatible in Community and Enterprise Editions.
Real-Time Messaging Protocol (RTMP) is a communication protocol for streaming audio, video, and data over the Internet. Originally developed as a proprietary protocol by Macromedia for streaming between Flash Player and the Flash Communication Server, Adobe (which acquired Macromedia) has released an incomplete version of the specification of ...
Audio/Video OpenBroadcaster: OpenBroadcaster 2003 5.0.0 (2018-01-27) 2018-01-27 Free AGPLv3: Audio/Video Open Broadcaster Software: OBS Project 2012-9-01 30.0.2 (2023-12-10) 2023-12-10 Free GPL v2: Audio/Video Plex (software) Plex Media Server 2008-07-08 1.15.6.1079 (2019-05-15) 2019-05-15 Free proprietary: Audio/Video Red5 (open source ...
Ampache – GPL/LGPL Audio streaming atmosph3re – responsive web-based streaming audio server for personal music collection; Ant Media Server Real-time streaming engine delivers content with sub-0.5secs latency. Ant Media Server supports WebRTC, CMAF, HLS, RTMP, RTSP, SRT, Zixi and more for your needs.
Adaptive streaming overview Adaptive streaming in action. Adaptive bitrate streaming is a technique used in streaming multimedia over computer networks.. While in the past most video or audio streaming technologies utilized streaming protocols such as RTP with RTSP, today's adaptive streaming technologies are based almost exclusively on HTTP, [1] and are designed to work efficiently over large ...
RFC 3551, entitled RTP Profile for Audio and Video (RTP/AVP), specifies the technical parameters of payload formats for audio and video streams. The standard also describes the process of registering new payload types with IANA; additional payload formats and payload types are defined in the following specifications:
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.