Glossary of Terms

CODEC

The word codec is a portmanteau of “coder” and “decoder”, appropriate since a codec is a combination of a piece of software or hardware that encodes information, usually streamed data such as audio or video, and a piece of software or hardare that decodes that information.  A codec always includes both pieces because the decoding algorithm is designed to decode what the encoder produces.  Examples of modern codecs for real-time communications include VP8, VP9, H.264, H.265, AV1, Opus, and G.711.

Congestion control

Modulates traffic entry into a telecommunications network in order to avoid congestive collapse resulting from oversubscription. 

Classification of congestion control algorithms

Among the ways to classify congestion control algorithms are:

  • By type and amount of feedback received from the network: Loss; delay; single-bit or multi-bit explicit signals
  • By incremental deployability: Only sender needs modification; sender and receiver need modification; only router needs modification; sender, receiver and routers need modification.
  • By performance aspect: high bandwidth-delay product networks; lossy links; fairness; advantage to short flows; variable-rate links
  • By fairness criterion: max-min, proportional, “minimum potential delay

DSCP – Differentiated Services Code Point

DSCP provides for a set of markings in Internet data packet headers that can be used by supporting hardware to prioritize some packets over others, thus guaranteeing specific levels of Quality of Service (QoS) for those packets.

DTLS – Datagram Transport Layer Security (DTLS)

TLS, but for datagram packets.  This means that UDP traffic can be encrypted in the same way that TLS encrypts HTTP traffic in HTTPS.  In WebRTC, DTLS is used to encrypt all media and data channel traffic.

DTMF

Dual Tone Multi Frequency signaling is the way landline phone keypresses are signaled to the phone company.  When a key is pressed, two tones of different frequency are played simultaneously over the phone line.  This unique pairing indicates which key was pressed.

H.264

MPEG-4 Part 10, Advanced Video Coding (MPEG-4 AVC) is a block-oriented motion-compensation-based video compression standard. As of 2014 it is one of the most commonly used formats for the recording, compression, and distribution of video content. It supports resolutions up to 4096×2304, including 4K UHD.

H.265

High Efficiency Video Coding (HEVC), or H.265, is a video compression standard designed to substantially improve coding efficiency when compared to its precedent, the Advanced Video Coding (AVC), or H.264.

ICE

Interactive Connectivity Establishment (ICE) is a technique used in computer networking to find ways for two computers to talk to each other as directly as possible in peer-to-peer networking. This is most commonly used for interactive media such as Voice over Internet Protocol (VoIP), peer-to-peer communications, video, and instant messaging.

IETF

Internet Engineering Task Force (IETF) develops and promotes voluntary Internet standards, in particular the standards that comprise the Internet protocol suite (TCP/IP).  Standards defined at the IETF include HTTP, HTTPS, FTP, RTP, ICE, STUN, TURN

ORTC

Object Real-time Communications is an object-centric application programming interface for WebRTC

Peer Connection

A PeerConnection allows media to flow directly between browsers without any intervening servers.

Peer-to-peer

Peer-to-peer (P2P) computing or networking is a distributed application architecture that partitions tasks or workloads between peers. Peers are equally privileged, equipotent participants in the application. They are said to form a peer-to-peer network of nodes.

Real-time communication

Real-time communications (RTC) is a term used to refer to any live telecommunications that occur without transmission delays. RTC is nearly instant with minimal latency.

SDP

is a format for describing streaming media communications parameters.  The current definition is IETF RFC 4566.  This format is used by WebRTC to configure and exchange low-level media parameters between the two browsers in a Peer Connection.

RFC

Request for Comments (RFC) is a type of publication from the Internet Engineering Task Force (IETF) and the Internet Society (ISOC), the principal technical development and standards-setting bodies for the Internet.

RTP

Real-time Transport Protocol (RTP) is a network protocol for delivering low-delay, gap-minimized (real-time) audio and video over IP networks.

SCTP

Stream Control Transmission Protocol (SCTP) is a transport-layer protocol, serving in a similar role to the popular protocols TCP and UDP.  WebRTC uses SCTP in its implementation of the data channel.

Signaling

In telecommunications, signaling refers to the use of signals to set up, control, and close down the communication channel(s).  Some well-known signaling approaches are switchhook signaling, SS7, and SIP.

Signaling channel

In the context of RTC, signaling channel refers to the application-specific mechanism used to communicate signaling information between browsers in a Peer Connection.

Simulcast

A portmanteau of simultaneous and broadcast, simulcast in the context of real-time communications refers to the simultaneous sending of media in multiple formats or resolutions.  It is one way to address the needs of multiple display devices that differ in their resolutions, the other being scalable codecs.

SIP

Session Initiation Protocol (SIP) is a communications protocol for signaling and controlling multimedia communication sessions, in applications of Internet telephony for voice and video calls, in private IP telephone systems, as well as in instant messaging over Internet Protocol (IP) networks.

SRTP

Secure Real-time Transport Protocol (or SRTP) is the encrypted version of RTP.  In addition to encryption, SRTP provides message authentication and integrity and replay protection.  WebRTC mandates the use of SRTP rather than RTP in order to ensure that communications can only be accessed by the communications peer.

STUN

Session Traversal Utilities for NAT (STUN) is a standardized set of methods, including a network protocol, for traversal of network address translator (NAT) gateways in applications of real-time voice, video, messaging, and other interactive communications.

SVC 

A switched virtual circuit (SVC) is a type of virtual circuit in telecommunication and computer networks that is used to establish a temporary connection between two different network nodes until completion of a data transfer session, after which the connection is terminated.

SVC

Scalable Video Coding (SVC) refers both to the name for the Annex G extension of the H.264/MPEG-4 AVC video compression standard and to a general approach to efficiently encoding video bitstreams at multiple resolutions and/or frame rates.

TLS

Transport Layer Security (TLS) and its predecessor, Secure Sockets Layer (SSL), both frequently referred to as “SSL”, are cryptographic protocols that provide communications security over a computer network.

TURN

TURN is a service that helps clients behind NAT routers and firewalls to discover the most efficient way to communicate with other clients and to relay the media streams if no direct media path can be found.

VoIP

Voice over Internet Protocol (also voice over IPVoIP or IP telephony) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.

VP8

An open and royalty free video compression format owned and released by Google.  It is one of the two required video codecs for WebRTC browser implementations.

VP9

An open and royalty free video coding format developed by Google.

VPN

virtual private network (VPN) extends a private network across a public network, and enables users to send and receive data across shared or public networks as if their computing devices were directly connected to the private network.

W3C

World Wide Web Consortium (W3C) is the main international standards organization for the World Wide Web (abbreviated WWW or W3).

WebRTC

Web Real-Time Communication is a collection of communications protocols and application programming interfaces that enable real-time communication over peer-to-peer connections.