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.
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.
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.
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.
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.
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.
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
Object Real-time Communications is an object-centric application programming interface for WebRTC
A PeerConnection allows media to flow directly between browsers without any intervening servers.
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 communications (RTC) is a term used to refer to any live telecommunications that occur without transmission delays. RTC is nearly instant with minimal latency.
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.
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.
Real-time Transport Protocol (RTP) is a network protocol for delivering low-delay, gap-minimized (real-time) audio and video over IP networks.
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.
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.
In the context of RTC, signaling channel refers to the application-specific mechanism used to communicate signaling information between browsers in a Peer Connection.
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.
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.
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.
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.
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.
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.
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 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.
Voice over Internet Protocol (also voice over IP, VoIP 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.
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.
An open and royalty free video coding format developed by Google.
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.
World Wide Web Consortium (W3C) is the main international standards organization for the World Wide Web (abbreviated WWW or W3).
Web Real-Time Communication is a collection of communications protocols and application programming interfaces that enable real-time communication over peer-to-peer connections.