Discover the new easier way to develop Kurento video applications

Glossary

This is a glossary of terms that often appear in discussion about multimedia transmissions. Some of the terms are specific to GStreamer or Kurento, and most of them are described and linked to their RFC, W3C or Wikipedia documents.

Agnostic media
One of the big problems of media is that the number of variants of video and audio codecs, formats and variants quickly creates high complexity in heterogeneous applications. So Kurento developed the concept of an automatic converter of media formats that enables development of agnostic elements. Whenever a media element’s source is connected to another media element’s sink, the Kurento framework verifies if media adaption and transcoding is necessary and, if needed, it transparently incorporates the appropriate transformations making possible the chaining of the two elements into the resulting Pipeline.
AVI

Audio Video Interleaved, known by its initials AVI, is a multimedia container format introduced by Microsoft in November 1992 as part of its Video for Windows technology. AVI files can contain both audio and video data in a file container that allows synchronous audio-with-video playback. AVI is a derivative of the Resource Interchange File Format (RIFF).

Bower
Bower is a package manager for the web. It offers a generic solution to the problem of front-end package management, while exposing the package dependency model via an API that can be consumed by a build stack.
Builder Pattern

The builder pattern is an object creation software design pattern whose intention is to find a solution to the telescoping constructor anti-pattern. The telescoping constructor anti-pattern occurs when the increase of object constructor parameter combination leads to an exponential list of constructors. Instead of using numerous constructors, the builder pattern uses another object, a builder, that receives each initialization parameter step by step and then returns the resulting constructed object at once.

See also

Builder pattern

CORS

Cross-origin resource sharing is a mechanism that allows JavaScript code on a web page to make XMLHttpRequests to different domains than the one the JavaScript originated from. It works by adding new HTTP headers that allow servers to serve resources to permitted origin domains. Browsers support these headers and enforce the restrictions they establish.

See also

Cross-origin resource sharing

enable-cors.org
Information on the relevance of CORS and how and when to enable it.
DOM
Document Object Model is a cross-platform and language-independent convention for representing and interacting with objects in HTML, XHTML and XML documents.
EOS
End Of Stream is an event that occurs when playback of some media source has finished. In Kurento, some elements will raise an EndOfStream event.
GStreamer
GStreamer is a pipeline-based multimedia framework written in the C programming language.
H.264

A Video Compression Format. The H.264 standard can be viewed as a “family of standards” composed of a number of profiles. Each specific decoder deals with at least one such profiles, but not necessarily all.

See also

H.264/MPEG-4 AVC

RFC 6184
RTP Payload Format for H.264 Video (This RFC obsoletes RFC 3984)
HTTP

The Hypertext Transfer Protocol (HTTP) is an application protocol for distributed, collaborative, hypermedia information systems. HTTP is the foundation of data communication for the World Wide Web.

See also

Hypertext Transfer Protocol

RFC 2616
Hypertext Transfer Protocol – HTTP/1.1
ICE

Interactive Connectivity Establishment (ICE) is a technique used to achieve NAT Traversal. ICE makes use of the STUN protocol and its extension, TURN. ICE can be used by any aplication that makes use of the SDP Offer/Answer model..

See also

Interactive Connectivity Establishment

RFC 5245
Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols
IMS

IP Multimedia Subsystem (IMS) is the 3GPP’s Mobile Architectural Framework for delivering IP Multimedia Services in 3G (and beyond) Mobile Networks.

See also

IP Multimedia Subsystem

3GPP

RFC 3574
Transition Scenarios for 3GPP Networks
jQuery
jQuery is a cross-platform JavaScript library designed to simplify the client-side scripting of HTML.
JSON
JSON (JavaScript Object Notation) is a lightweight data-interchange format. It is designed to be easy to understand and write for humans and easy to parse for machines.
JSON-RPC
JSON-RPC is a simple remote procedure call protocol encoded in JSON. JSON-RPC allows for notifications and for multiple calls to be sent to the server which may be answered out of order.
Kurento
Kurento is a platform for the development of multimedia-enabled applications. Kurento is the Esperanto term for the English word ‘stream’. We chose this name because we believe the Esperanto principles are inspiring for what the multimedia community needs: simplicity, openness and universality. Some components of Kurento are the Kurento Media Server, the Kurento API, the Kurento Protocol, and the Kurento Client.
Kurento API
An object oriented API to create media pipelines to control media. It can be seen as and interface to Kurento Media Server. It can be used from the Kurento Protocol or from Kurento Clients.
Kurento Client
A programming library (Java or JavaScript) used to control an instance of Kurento Media Server from an application. For example, with this library, any developer can create a web application that uses Kurento Media Server to receive audio and video from the user web browser, process it and send it back again over Internet. The Kurento Client libraries expose the Kurento API to application developers.
Kurento Protocol
Communication between KMS and clients by means of JSON-RPC messages. It is based on WebSocket that uses JSON-RPC v2.0 messages for making requests and sending responses.
KMS
Kurento Media Server
Kurento Media Server is the core element of Kurento since it responsible for media transmission, processing, loading and recording.
Maven
Maven is a build automation tool used primarily for Java projects.
Media Element
A Media Element is a module that encapsulates a specific media capability. For example RecorderEndpoint, PlayerEndpoint, etc.
Media Pipeline
A Media Pipeline is a chain of media elements, where the output stream generated by one element (source) is fed into one or more other elements input streams (sinks). Hence, the pipeline represents a “machine” capable of performing a sequence of operations over a stream.
Media Plane

In a traditional IP Multimedia Subsystem, the handling of media is conceptually splitted in two layers. The layer that handles the media itself -with functionalities such as media transport, encoding/decoding, and processing- is called Media Plane.

MP4

MPEG-4 Part 14 or MP4 is a digital multimedia format most commonly used to store video and audio, but can also be used to store other data such as subtitles and still images.

See also

MPEG-4 Part 14

Multimedia

Multimedia is concerned with the computer controlled integration of text, graphics, video, animation, audio, and any other media where information can be represented, stored, transmitted and processed digitally. There is a temporal relationship between many forms of media, for instance audio, video and animations. There 2 are forms of problems involved in

  • Sequencing within the media, i.e. playing frames in correct order or time frame.
  • Synchronization, i.e. inter-media scheduling. For example, keeping video and audio synchronized or displaying captions or subtitles in the required intervals.

See also

Multimedia

Multimedia container format

Container or wrapper formats are meta-file formats whose specification describes how different data elements and metadata coexist in a computer file. Simpler multimedia container formats can contain different types of audio formats, while more advanced container formats can support multiple audio and video streams, subtitles, chapter-information, and meta-data, along with the synchronization information needed to play back the various streams together. In most cases, the file header, most of the metadata and the synchro chunks are specified by the container format.

NAT
Network Address Translation

Network address translation (NAT) is the technique of modifying network address information in Internet Protocol (IP) datagram packet headers while they are in transit across a traffic routing device for the purpose of remapping one IP address space into another.

NAT-T
NAT Traversal

NAT traversal (sometimes abbreviated as NAT-T) is a general term for techniques that establish and maintain Internet protocol connections traversing network address translation (NAT) gateways, which break end-to-end connectivity. Intercepting and modifying traffic can only be performed transparently in the absence of secure encryption and authentication.

See also

NAT Types and NAT Traversal
Entry in our Knowledge Base.
Node.js
Node.js is a cross-platform runtime environment for server-side and networking applications. Node.js applications are written in JavaScript, and can be run within the Node.js runtime on OS X, Microsoft Windows and Linux with no changes.
npm
npm is the official package manager for Node.js.
OpenCV
OpenCV (Open Source Computer Vision Library) is a BSD-licensed open source computer vision and machine learning software library. OpenCV aims to provide a common infrastructure for computer vision applications and to accelerate the use of machine perception.
Pad, Media

A Media Pad is is an element’s interface with the outside world. Data streams from the MediaSource pad to another element’s MediaSink pad.

See also

GStreamer Pad
Definition of the Pad structure in GStreamer
QR

QR code (Quick Response Code) is a type of two-dimensional barcode. that became popular in the mobile phone industry due to its fast readability and greater storage capacity compared to standard UPC barcodes.

See also

QR code

REMB

Receiver Estimated Maximum Bitrate (REMB) is a type of RTCP feedback message that a RTP receiver can use to inform the sender about what is the estimated reception bandwidth currently available for the stream itself. Upon reception of this message, the RTP sender will be able to adjust its own video bitrate to the conditions of the network. This message is a crucial part of the Google Congestion Control (GCC) algorithm, which provides any RTP session with the ability to adapt in cases of network congestion.

The GCC algorithm is one of several proposed algorithms that have been proposed by an IETF Working Group named RTP Media Congestion Avoidance Techniques (RMCAT).

See also

What is RMCAT congestion control, and how will it affect WebRTC?

draft-alvestrand-rmcat-remb
RTCP message for Receiver Estimated Maximum Bitrate
draft-ietf-rmcat-gcc
A Google Congestion Control Algorithm for Real-Time Communication
REST

Representational state transfer (REST) is an architectural style consisting of a coordinated set of constraints applied to components, connectors, and data elements, within a distributed hypermedia system. The term representational state transfer was introduced and defined in 2000 by Roy Fielding in his doctoral dissertation.

RTCP

The RTP Control Protocol (RTCP) is a sister protocol of the RTP, that provides out-of-band statistics and control information for an RTP flow.

See also

RTP Control Protocol

RFC 3605
Real Time Control Protocol (RTCP) attribute in Session Description Protocol (SDP)
RTP

Real-time Transport Protocol (RTP) is a standard packet format designed for transmitting audio and video streams on IP networks. It is used in conjunction with the RTP Control Protocol. Transmissions using the RTP audio/video profile (RTP/AVP) typically use SDP to describe the technical parameters of the media streams.

See also

Real-time Transport Protocol

RTP audio video profile

RFC 3550
RTP: A Transport Protocol for Real-Time Applications
Same-origin policy

The “same-origin policy” is a web application security model. The policy permits scripts running on pages originating from the same domain to access each other’s DOM with no specific restrictions, but prevents access to DOM on different domains.

SDP
Session Description Protocol
SDP Offer/Answer

The Session Description Protocol (SDP) is a text document that describes the parameters of a streaming media session. It is commonly used to describe the characteristics of RTP streams (and related protocols such as RTSP).

The SDP Offer/Answer model is a negotiation between two peers of a unicast stream, by which the sender and the receiver share the set of media streams and codecs they wish to use, along with the IP addresses and ports they would like to use to receive the media.

This is an example SDP Offer/Answer negotiation. First, there must be a peer that wishes to initiate the negotiation; we’ll call it the offerer. It composes the following SDP Offer and sends it to the other peer -which we’ll call the answerer-:

v=0
o=- 0 0 IN IP4 127.0.0.1
s=Example sender
c=IN IP4 127.0.0.1
t=0 0
m=audio 5006 RTP/AVP 96
a=rtpmap:96 opus/48000/2
a=sendonly
m=video 5004 RTP/AVP 103
a=rtpmap:103 H264/90000
a=sendonly

Upon receiving that Offer, the answerer studies the parameters requested by the offerer, decides if they can be satisfied, and composes an appropriate SDP Answer that is sent back to the offerer:

v=0
o=- 3696336115 3696336115 IN IP4 192.168.56.1
s=Example receiver
c=IN IP4 192.168.56.1
t=0 0
m=audio 0 RTP/AVP 96
a=rtpmap:96 opus/48000/2
a=recvonly
m=video 31278 RTP/AVP 103
a=rtpmap:103 H264/90000
a=recvonly

The SDP Answer is the final step of the SDP Offer/Answer Model. With it, the answerer agrees to some of the parameter requested by the offerer, but not all.

In this example, audio 0 means that the answerer rejects the audio stream that the offerer intended to send; also, it accepts the video stream, and the a=recvonly acknowledges that the answerer will exclusively act as a receiver, and won’t send any stream back to the other peer.

See also

Session Description Protocol

Anatomy of a WebRTC SDP

RFC 4566
SDP: Session Description Protocol
RFC 4568
Session Description Protocol (SDP) Security Descriptions for Media Streams
Semantic Versioning
Semantic Versioning is a formal convention for specifying compatibility using a three-part version number: major version; minor version; and patch.
Signaling Plane

It is the layer of a media system in charge of the information exchanges concerning the establishment and control of the different media circuits and the management of the network, in contrast to the transfer of media, done by the Signaling Plane. Functions such as media negotiation, QoS parametrization, call establishment, user registration, user presence, etc. as managed in this plane.

See also

Media Plane

Sink, Media
A Media Sink is a MediaPad that outputs a Media Stream. Data streams from a MediaSource pad to another element’s MediaSink pad.
SIP

Session Initiation Protocol (SIP) is a signaling plane protocol widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. SIP works in conjunction with several other application layer protocols:

  • SDP for media identification and negotiation.
  • RTP, SRTP or WebRTC for the transmission of media streams.
  • A TLS layer may be used for secure transmission of SIP messages.
Source, Media
A Media Source is a Media Pad that generates a Media Stream.
SPA
Single-Page Application
A single-page application is a web application that fits on a single web page with the goal of providing a more fluid user experience akin to a desktop application.
Sphinx

Sphinx is a documentation generation system. Text is first written using reStructuredText markup language, which then is transformed by Sphinx into different formats such as PDF or HTML. This is the documentation tool of choice for the Kurento project.

Spring Boot
Spring Boot is Spring’s convention-over-configuration solution for creating stand-alone, production-grade Spring based applications that can you can “just run”. It embeds Tomcat or Jetty directly and so there is no need to deploy WAR files in order to run web applications.
SRTCP

SRTCP provides the same security-related features to RTCP, as the ones provided by SRTP to RTP. Encryption, message authentication and integrity, and replay protection are the features added by SRTCP to RTCP.

See also

SRTP

SRTP

Secure RTP is a profile of RTP (Real-time Transport Protocol), intended to provide encryption, message authentication and integrity, and replay protection to the RTP data in both unicast and multicast applications. Similarly to how RTP has a sister RTCP protocol, SRTP also has a sister protocol, called Secure RTCP (or SRTCP).

See also

Secure Real-time Transport Protocol

RFC 3711
The Secure Real-time Transport Protocol (SRTP)
SSL
Secure Socket Layer. See TLS.
STUN
STUN stands for Session Traversal Utilities for NAT. It is a standard protocol (IETF RFC 5389) used by NAT traversal algorithms to assist hosts in the discovery of their public network information. If the routers between peers use full cone, address-restricted, or port-restricted NAT, then a direct link can be discovered with STUN alone. If either one of the routers use symmetric NAT, then a link can be discovered with STUN packets only if the other router does not use symmetric or port-restricted NAT. In this later case, the only alternative left is to discover a relayed path through the use of TURN.
Trickle ICE

Extension to the ICE protocol that allows ICE agents to send and receive candidates incrementally rather than exchanging complete lists. With such incremental provisioning, ICE agents can begin connectivity checks while they are still gathering candidates and considerably shorten the time necessary for ICE processing to complete.

See also

draft-ietf-ice-trickle
Trickle ICE: Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) Protocol
TLS

Transport Layer Security (TLS) and its predecessor Secure Socket Layer (SSL).

See also

Transport Layer Security

RFC 5246
The Transport Layer Security (TLS) Protocol Version 1.2
TURN
TURN stands for Traversal Using Relays around NAT. Like STUN, it is a network protocol (IETF RFC 5766) used to assist in the discovery of paths between peers on the Internet. It differs from STUN in that it uses a public intermediary relay to act as a proxy for packets between peers. It is used when no other option is available since it consumes server resources and has an increased latency. The only time when TURN is necessary is when one of the peers is behind a symmetric NAT and the other peer is behind either a symmetric NAT or a port-restricted NAT.
VP8

VP8 is a video compression format created by On2 Technologies as a successor to VP7. Its patents rights are owned by Google, who made an irrevocable patent promise on its patents for implementing it and released a specification under the Creative Commons Attribution 3.0 license.

See also

VP8

RFC 6386
VP8 Data Format and Decoding Guide
WebM
WebM is an open media file format designed for the web. WebM files consist of video streams compressed with the VP8 video codec and audio streams compressed with the Vorbis audio codec. The WebM file structure is based on the Matroska media container.
WebRTC

WebRTC is a set of protocols, mechanisms and APIs that provide browsers and mobile applications with Real-Time Communications (RTC) capabilities over peer-to-peer connections.

WebSocket
WebSocket specification (developed as part of the HTML5 initiative) defines a full-duplex single socket connection over which messages can be sent between client and server.