I was searching around for a simple WebRTC glossary but could not find one, so I decided to make one of my own. Please see below for my list of the top 50 WebRTC terms and acronyms with a description of each.

Category

Acronym/Term

What is it?

Signaling & APIs

createPeerConnection JavaScript API that makes a secure connection to another "peer" - another browser or a server side element
CU-RTC-Web Customizable, Ubiquitous Real Time Communication over the Web - Microsoft's alternative recommendation for WebRTC API's. (See ORTC for the evolution of this)
DataChannel JavaScript API that allows arbitrary data to be sent between peers
getUserMedia JavaScript API that allows access to the Camera and Microphone
H2S HTTP-to-SIP conversion - element that converts proprietary web-based signaling to SIP
JSON JavaScript Object Notation - a lightweight data exchange format that is native to JavaScript and easy for humans to read and write
JSONoWS A WebRTC signaling mechanism that utilizes proprietary signaling in JSON format with WebSockets as a transport protocol
orca.js Open Realtime Communications API - Alliance for Telecommunications Industry Solutions effort to provide a standardized API into Teclo Service Provider IMS networks
ORTC Object Real Time Communication - World Wide Web Consortium's (W3C) Community Group working and recommendation for augmenting the WebRTC APIs with increased controls to provide broader use-case applicability
RESTful Representational state transfer - an API architectural style that embodies common web systems
SIPoWS A WebRTC signaling mechanism where a SIP stack in the browser with WebSockets as a transport protocol to signal a SIP-based network
WebRTC Web Real Time Communications - the umbrella term for this technology and name of the World Wide Web Consortium's (W3C) working group to standardize the technology in that body
WebSocket A bi-directional, full-duplex protocol that was made to work in web-environments without requiring firewall changes

Infrastructure Software and Devices

eIMS-AGW IMS Access GateWay enhanced for WebRTC -  3GPP gateway element that can translate the WebRTC media plane into formats compatible with the IMS network
eP-CSCF P-CSCF enhanced for WebRTC - entry point for a WIC into the 3GPP IMS network through use of SIPoWS or other signaling protocols
MCU Multi-point Control Unit - bridging device and multi-party conferencing model for videoconferences that provides full mixing between several parties
Media Server Network device that handles media processing functions - including those for WebRTC - such as multi-party conferencing, transcoding, media plane interworking, stream processing, recording, machine interaction, and more
MRF Media Resource Function - 3GPP term for an IMS media server
Server-side Application logic and/or processing that occurs in the network or cloud
SFU Selective Forwarding Unit - multi-party conferencing device and architecture that is able to forward a single incoming media stream to one or more parties
STUN server A server-side infrastructure element that uses the STUN protocol to return an external IP address to a user behind a NAT
TURN server A server-side infrastructure element that uses the TURN protocl to relays media between two peers
WIC WebRTC IMS Client - 3GPP terminology for a JavaScript based application that can interface with the IMS network
WWSF WebRTC Web Server Function - 3GPP name for a web server in an IMS network

Media & Codecs

 
FEC Forward Error Correction - technique for improving media stream reliability by correcting for lost packets
Full-mesh Multi-party conferencing architecture where each peer must send and receive individual media streams to each and every other peer
G.711 Widely used audio compression format used widely in telephony and mandated by WebRTC
H.264 (aka MPEG-4 Part 10 & Advaced Video Coding) Widely used compression format used in existing  video telephony systems, web video streaming, and television governed by the MPEG-LA
MPEG-LA Patent holder responsible for pooling the licenses of various patent holders and usage charging for H.264 and other codecs
OpenH264 Cisco's open-sourcing of the H.264 codec allowing free use of the codec with Cisco bearing the royalty costs
Opus Open and royalty free audio compression format designed for internet environments, supports narrow and wideband, and is mandated by WebRTC
RTCP RTP Control Protocol - provides statistics and control information for a RTP stream
RTCP multiplex Real Time Control Protocol (RTCP) Multiplexing - allows RTCP packets to share the same port as SRTP packets to simplify network management and increase connectivity
RTP Real-time Transport Protocol - standardized packet format for delivering real time media such as audio and video over an IP network
RTP Bundle Technique used inside SDP to allow multiple streams within a single IP address and port to facilitate Firewall and NAT traversal
SDP Session Description Protocol - format for streaming media initialization used in VoIP networks and WebRTC
Simulcast A technique in WebRTC where a single camera can be used to transmit multiple video streams from the same source with variable bitrates (often for use with a SFU)
SVC Scalable Video Coding - a video encoding technique that utilizes progressive of increasing bitrates to allow for dynamic adjustment of bitrate vs video size, frame rate, and/or quality
Transcoding Coversion of one audio and/or video compression technique to another
VP8 Royalty free video Compression format open sourced by Google used by Chrome, Opera, and Firefox, and Android for WebRTC
VP9 Royalty free video compression format and successor to VP8 with the goal of reducing bandwidth by 50% and providing better compression efficiency than HEVC (aka H.265); road mapped by Google for use in WebRTC

Network & Security

 
DTLS Datagram Transport Layer Security - security encryption mechanism mandated in WebRTC
ICE Interactive Connectivity Establishment - protocol for traversing NATs that can leverage STUN and ICE
ICE-lite Implementing Interactive Connectivity Establishment (ICE) in Lite Mode - lightweight, easier-to-implement version of ICE where one peer has a public address
NAT Network Address Translator - widely used technology for providing a private address spaces behind a public address used for segmentation and increasing the number of addresses
RTCweb Real Time Communications Web - the name of the Internet Engineering Task Force (IETF) working group to standardize WebRTC in that body
SRTP Secure Real-time Transport Protocol - encrypted RTP mandated in WebRTC
STUN Session Traversal Utilities for NAT - tool used in NAT traversal for identifying an address behind a NAT
Trickle-ICE Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) - version of the ICE protocol the allows faster connectivity
TURN Traversal Using Relays around NAT - a protocol used as part of NAT that provides media relaying when two peers cannot otherwise directly communicate
createPeerConnection JavaScript API that makes a secure connection to another "peer" - another browser or a server side element
CU-RTC-Web Customizable, Ubiquitous Real Time Communication over the Web - Microsoft's alternative recommendation for WebRTC API's. (See ORTC for the evolution of this)
DataChannel JavaScript API that allows arbitrary data to be sent between peers
getUserMedia JavaScript API that allows access to the Camera and Microphone
H2S HTTP-to-SIP conversion - element that converts proprietary web-based signaling to SIP
JSON JavaScript Object Notation - a lightweight data exchange format that is native to JavaScript and easy for humans to read and write
JSONoWS A WebRTC signaling mechanism that utilizes proprietary signaling in JSON format with WebSockets as a transport protocol
orca.js Open Realtime Communications API - Alliance for Telecommunications Industry Solutions effort to provide a standardized API into Teclo Service Provider IMS networks
ORTC Object Real Time Communication - World Wide Web Consortium's (W3C) Community Group working and recommendation for augmenting the WebRTC APIs with increased controls to provide broader use-case applicability
RESTful Representational state transfer - an API architectural style that embodies common web systems
SIPoWS A WebRTC signaling mechanism where a SIP stack in the browser with WebSockets as a transport protocol to signal a SIP-based network
WebRTC Web Real Time Communications - the umbrella term for this technology and name of the World Wide Web Consortium's (W3C) working group to standardize the technology in that body

Now that you are WebRTC 50 terms smarter, do you want to know more about server-side media processing in WebRTC?

Check out WebRTC blogger Tsahi Levant-Levi’s whitepaper:

Seven Reasons for WebRTC Server-Side Media Processing

Do you need an actual media server? Check out PowerMedia XMS or

download & try it out yourself here

Let me know in the comments if you think I missed anything.