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Fault-Tolerant Real-Time Streaming with FEC thanks to Capillary Multi-Path Routing

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Streaming media

Streaming media is multimedia that is delivered and consumed in a continuous manner from a source, with little or no intermediate storage in network elements. Streaming refers to the delivery method of content, rather than the content itself. Distinguishing delivery method from the media applies specifically to telecommunications networks, as most of the traditional media delivery systems are either inherently streaming (e.g. radio, television) or inherently non-streaming (e.g. books, videotapes, audio CDs).

Real Time Streaming Protocol

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. RTSP is used in entertainment and communications systems to control streaming media servers. The protocol is used for establishing and controlling media sessions between endpoints.

Erasure code

In coding theory, an erasure code is a forward error correction (FEC) code under the assumption of bit erasures (rather than bit errors), which transforms a message of k symbols into a longer message (code word) with n symbols such that the original message can be recovered from a subset of the n symbols. The fraction r = k/n is called the code rate. The fraction k’/k, where k’ denotes the number of symbols required for recovery, is called reception efficiency.

Error correction code

In computing, telecommunication, information theory, and coding theory, forward error correction (FEC) or channel coding is a technique used for controlling errors in data transmission over unreliable or noisy communication channels. The central idea is that the sender encodes the message in a redundant way, most often by using an error correction code or error correcting code (ECC). The redundancy allows the receiver not only to detect errors that may occur anywhere in the message, but often to correct a limited number of errors.

Real-Time Messaging Protocol

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 the protocol for public use. 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.

Opus (audio format)

Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed to efficiently code speech and general audio in a single format, while remaining low-latency enough for real-time interactive communication and low-complexity enough for low-end embedded processors. Opus replaces both Vorbis and Speex for new applications, and several blind listening tests have ranked it higher-quality than any other standard audio format at any given bitrate until transparency is reached, including MP3, AAC, and HE-AAC.

Link aggregation

In computer networking, link aggregation is the combining (aggregating) of multiple network connections in parallel by any of several methods. Link aggregation increases total throughput beyond what a single connection could sustain, and provides redundancy where all but one of the physical links may fail without losing connectivity. A link aggregation group (LAG) is the combined collection of physical ports. Other umbrella terms used to describe the concept include trunking, bundling, bonding, channeling or teaming.

Real-time Transport Protocol

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. RTP typically runs over User Datagram Protocol (UDP). RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g.

Gapless playback

Gapless playback is the uninterrupted playback of consecutive audio tracks, such that relative time distances in the original audio source are preserved over track boundaries on playback. For this to be useful, other artifacts (than timing-related ones) at track boundaries should not be severed either. Gapless playback is common with compact discs, gramophone records, or tapes, but is not always available with other formats that employ compressed digital audio.

Routing table

In computer networking, a routing table, or routing information base (RIB), is a data table stored in a router or a network host that lists the routes to particular network destinations, and in some cases, metrics (distances) associated with those routes. The routing table contains information about the topology of the network immediately around it. The construction of routing tables is the primary goal of routing protocols. Static routes are entries that are fixed, rather than resulting from routing protocols and network topology discovery procedures.