Several new releases of open media servers are available for streaming on the web:
- first edition
, a toolkit for real-time network audio streaming with guaranteed latency and CD-quality quality. The transmission takes into account the time deviation of the sender's and receiver's system clocks. Supports recovery of lost packets using codes in implementation (in the minimum delay mode, the Reed-Solomon code is used, and in the maximum performance mode, the scheme ). The transmission uses the RTP protocol (AVP L16, 44100Hz PCM 16-bit). Currently only audio streaming is supported, but there are plans to support streaming video and other types of content.It is possible to multiplex a stream from several senders for delivery to one recipient. It is possible to connect different sampling profiles, depending on the type of CPU and requirements for transmission delays. It supports broadcasting over various types of networks, including LAN, Internet, and wireless network. Depending on the settings, bandwidth and packet loss, Roc automatically selects the necessary stream encoding parameters and adjusts its intensity during transmission.
The project consists of a C library, command line and a set of modules for using Roc as a transport in . At its simplest, the tools available allow you to route audio from a file or sound device on one computer to a file or sound device on another computer. Various audio backends are supported, including ALSA, PulseAudio, and CoreAudio. The code is written in C++ and licensed under MPL-2.0. Work is supported in GNU/Linux and macOS.
- new media server release , which allows organizing streaming via RTMP, RTSP and WebRTC protocols with support for adaptive bitrate change mode. Ant can also be used to organize network video recording in MP4, HLS and FLV formats. Among the features, we can note the presence of a WebRTC to RTMP converter, support for IP cameras and IPTV, distribution and recording of live streams, organizing streaming to social networks, providing scaling through the deployment of a cluster, the possibility of mass broadcasting from one point to many recipients with delays at the level of 500ms.
The product is developed within the framework of the Open Core model, which implies the development of the main part under the Apache 2.0 license and the delivery of advanced features (for example, streaming to Youtube) in a paid edition. In the new version, the performance of broadcasting via WebRTC has been increased by 40%, a log viewer has been added, the web panel has been improved, a REST API has been added for displaying statistics, memory consumption has been optimized, error handling has been improved, and the ability to send statistics to Apache Kafka has been added.
- streaming server release , which allows you to transfer video in FLV, F4V, MP4 and 3GP formats, as well as audio in MP3, F4A, M4A, AAC formats. Live broadcasting and work in the form of a recording station for receiving streams from clients (FLV and AVC + AAC in an FLV container) are available. The project was originally created in 2005 to create an alternative to Flash Communication Server using the RTMP protocol. Later in Red5, through plug-ins, broadcasting support was provided using HLS, WebSockets, RTSP and WebRTC.
Red5 is used as a streaming server in a project for organizing video and audio conferences. The code is written in Java and licensed under Apache 2.0. A proprietary product is built on the basis of Red5 , which scales to millions of viewers with 500ms delivery delays and can be deployed on AWS, Google Cloud, and Azure.
Source: opennet.ru