Several new releases of open media servers are available, designed for organizing streaming over the network:
- first release
, a toolkit for real-time audio streaming over a network with guaranteed latency and CD-quality assurance. The transmission takes into account the clock skew of the sender and receiver. It supports the recovery of lost packets using in implementation. (in minimal latency mode, Reed-Solomon coding is applied, and in maximum performance mode, the ). The RTP protocol (AVP L16, 44100Hz PCM 16-bit) is used for transmission. Currently, only audio transmission is supported, but plans are in place for video streaming and other content types.There is the ability to multiplex streams from multiple senders to a single receiver. Different sampling rate setting profiles can be connected depending on the CPU type and latency requirements during transmission. Streaming over various types of networks is supported, including local area networks, the internet, and wireless networks. Depending on the settings, bandwidth, and packet loss, Roc automatically selects the necessary encoding parameters for the stream and adjusts its intensity during transmission.
The project consists of a C library, a command line interface, and a set of modules for using Roc as a transport in . In its simplest case, the available toolkit allows directing audio from a file or sound device on one computer to a file or sound device of another computer. Various audio backends are supported, including ALSA, PulseAudio, and CoreAudio. The code is written in C++ and is licensed under MPL-2.0. It works on GNU/Linux and macOS.
- new release of the multimedia server , allowing for live streaming through RTMP, RTSP, and WebRTC protocols with adaptive bitrate streaming support. Ant can also be used for network video recording in MP4, HLS, and FLV formats. Notable features include a WebRTC to RTMP converter, support for IP cameras and IPTV, distribution and recording of live streams, streaming to social networks, scalability through cluster deployment, and the ability to broadcast to multiple recipients from a single point with delays of up to 500ms.
The product is developed under the Open Core model, which involves developing the core part under the Apache 2.0 license and providing extended features (such as streaming to YouTube) in a paid edition. The new version has increased WebRTC streaming performance by 40%, added a log viewer, improved web panel, introduced a REST API for statistics output, optimized memory consumption, enhanced error handling, and added the capability to send statistics to Apache Kafka.
- release of the streaming server , enabling video transmission in FLV, F4V, MP4, and 3GP formats, as well as audio in MP3, F4A, M4A, and AAC formats. Available modes include Live streaming and operation as a recording station for receiving streams from clients (FLV and AVC+AAC in FLV container). The project was initially created in 2005 to provide an alternative to Flash Communication Server using the RTMP protocol. Later, Red5 supported streaming using HLS, WebSockets, RTSP, and WebRTC through plugins.
Red5 is used as a streaming server in the project for organizing video and audio conferences. The code is written in Java and is licensed under Apache 2.0. A proprietary product is built on Red5, , providing scalability up to millions of viewers with delivery delays of up to 500ms and the capability for deployment in AWS, Google Cloud, and Azure clouds.
Source: opennet.ru
