WebRTC Glossary by WebFL.US

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WebRTC Glossary

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  • AJAX : Asynchronous JavaScript and XML is a JavaScript web programming technique used to send/retrieve XML, JSON and other data to/from a server asynchronously.
  • API : An application programming interface specifies how one software program asks another to perform functions, provide information or grant access to resources.
  • app : "App" is short for computer software programming application. If preceded by "web" or "mobile" it is often a marketing more so than technical term.
  • BICC (ITU-T Q.1901) : The ITU-T Bearer Independent Call Control protocol describes the adaptation of narrow-band ISUP to enable ISDN services over a broadband backbone network.
  • DTLS (RFC 6347) : Datagram Transport Layer Security over SCTP provides communications privacy designed to prevent eavesdropping and detect tampering or message forgery.
  • DTMF (ITU-T Q.23) : Dual-Tone Multi-Frequency signalling is used for telecommunications between telephones and switches over analog telephone lines in the voice-frequency band.
  • ECMAScript : The scripting language standardized by the ECMA-262 specification and ISO/IEC 16262 widely used for programming on the web in the form known as JavaScript.
  • EIM : Endpoint-Independent Mapping allows P2P apps to learn and advertise the external IP address and port allocated to an internal endpoint to external peers.
  • G.711 : The high bit rate (64 Kbps) voice codec standard aka pulse code modulation (PCM) of voice frequencies used by PSTN and ISDN and required by H.320 and H.323
  • getUserMedia API | Media Capture API : A WebRTC API used to get access to local devices that can generate multimedia stream data and by which JavaScript can process or manipulate the stream data. See also:
  • H.264 : A.k.a. International Standard ISO/IEC 14496-10, an evolution of existing video codec standards developed to deliver even higher compression for apps like WebRTC.
  • H.265 (HEVC) : High Efficiency Video Coding is the successor to H.264/MPEG-4 AVC (Advanced Video Coding) that doubles the data compression ratio of AVC at the same level of video quality.
  • H.323 : This ITU-T recommendation addresses entities providing multimedia services over packet-based networks which may not provide guaranteed Quality of Service.
  • hole-punching, TCP : TCP Hole Punching (Simultaneous TCP Open) is a NAT traversal technique for establishing a P2P connection between endpoints that are both behind an EIM-NAT.
  • hole-punching, UDP : UDP Hole Punching relies on properties of EIM-NATs to allow P2P applications to "punch holes" through a NAT or firewall and establish direct connectivity.
  • HTTP (RFC 2616) : Hypertext Transfer Protocol is the application-layer protocol for distributed, collaborative, hypermedia information systems that drives the World Wide Web.
  • HTTPS (RFC 2818) : Hypertext Transfer Protocol Secure is not an actual protocol but a layering of HTTP over TLS that adds security capabilities to standard web communications.
  • Hybrid HTML5 Mobile Apps : A CIO-preferred WebRTC webdev strategy whereby the bulk of mobile app development is done in HTML5 employing native OS app wrappers if and where necessary.
  • ICE (RFC 5245) : Interactive Connectivity Establishment is an offer/answer model extension for NAT traversal that enables P2P media streams connectivity using STUN and TURN.
  • IETF : The mission of the Internet Engineering Task Force is to make the Internet work better by producing technical documents that influence Internet operations.
  • iLBC (RFC 3952) : Internet Low Bit Rate Codec is a free narrowband voice codec used in VoIP and streaming audio applications and included in the WebRTC browser voice engine.
  • IP (RFC 791) : Internet Protocol is used for transmitting blocks of data called datagrams from sources to destinations both of which are hosts with fixed length addresses.
  • IPv4 (RFC 791) : Internet Protocol Version 4 was the first public version of IP and was developed by DARPA before becoming the foundation for the Internet and World Wide Web. See also:
  • IPv6 (RFC 2460) : Internet Protocol Version 6 is being implemented to replace IPv4 with addresses represented as eight groups of four hexadecimal digits separated by colons. See also:
  • iSAC : Internet Speech Audio Codec is a robust wideband voice codec used in VoIP and streaming audio applications and included in the WebRTC browser voice engine.
  • ISDN : Integrated Services for Digital Network is a suite of communications protocols for simultaneous digital transmission of voice, video and data over the PSTN.
  • ISUP (ITU-T Q.76x) : ISDN user part is the SS7 protocol for the signaling functions that support basic bearer services and supplementary services for voice/non-voice ISDN apps.
  • ITU-T : The ITU Telecommunication Standardization Sector is part of the International Telecommunication Union, the United Nations agency focused on telecom and IT.
  • Jingle : An XMPP protocol extension for initiating and managing P2P media sessions between two XMPP entities in a manner interoperable with existing IP standards.
  • JSEP : Javascript Session Establishment Protocol describes how the RTCPeerConnection is used to control the setup, management and teardown of a multimedia session. See also:
  • JsSIP : A pure JavaScript SIP stack using SIP WebSocket Transport for signaling and WebRTC for audio/video that works with Asterisk, Kamailio and OverSIP servers.
  • Media Capture Task Force : A joint task force of the W3C Web Real-Time Communications and Device APIs Working Groups developing APIs to capture media from cameras and microphones.
  • MediaStream : A container or array of MediaStreamTrack objects created from accessible media sources returned by getUserMedia that can be sent over an RTCPeerConnection. See also:
  • MediaStreamConstraints : Audio/video boolean or MediaTrackConstraints to tell user agents what type of MediaStreamTrack objects to include in a MediaStream returned by getUserMedia.
  • MediaStreamTrack : A MediaStream object comprised of one+ channels representing audio, video or other media in the user agent that may also be represented by similar objects. See also:
  • NAT (RFC 1631) : The IP Network Address Translator changes a packet's source IP address (local address) to another (global address) before forwarding the packet on to a WAN. See also:
  • NetEQ : A VoIP receiving-end speech-processing solution to improve sound quality and minimize jitter buffer delay that is part of the WebRTC browser voice engine.
  • Opus (RFC 6716) : An open source audio codec (RFC 6716) for interactive speech and music transmission plus storage and streaming based in part on Skype SILK and Xiph CELT.
  • OSI (ITU-T X.200 | ISO/IEC ISO/IEC 7498-1) : The Open Systems Interconnection model abstracts communications systems into seven layers: physical, data link, network, transport, session, presentation and application.
  • P2P : Peer-to-peer is a decentralized, distributed and democratized network architecture in which individual nodes can act as both servers and clients.
  • privacy : The right and mandate that the personal, confidential or sensitive information of a WebRTC user will not be conveyed or exposed to unauthorized parties.
  • PSTN : The public switched telephone network is the global aggregate of circuit-switched telephone networks operated by all national, regional or other operators.
  • RTCConfiguration : Defines an array of RTCIceServers (STUN and TURN server credentials) to be used by ICE across a signaling channel via XHR to establish an RTCPeerConnection. See also:
  • RTCDataChannel : A WebRTC API that enables real-time P2P exchange of arbitrary data with low latency and high throughput, simultaneous channels and built-in DTLS security. See also:
  • RTCDTMFSender | DTMF API : Created by calling createDTMFSender() on an RTCPeerConnection to construct an object that exposes the functions required to send DTMF on a MediaStreamTrack.
  • RTCIdentityAssertion | Identity API : Extends RTCPeerConnection with getIdentityAssertion() to obtain an identity assertion and setIdentityProvider() if the browser has no IdP already configured.
  • RTCP (RFC 3550) : RTP Control Protocol is based on the periodic transmission of control packets to all participants using the same distribution mechanism as the data packets.
  • RTCPeerConnection : A WebRTC API that encapsulates all connection setup, management and state including local streams, remote streams and ICE agent STUN/TURN NAT traversal. See also:
  • RTCStats | Statistics API : Extends RTCPeerConnection with getStats() which gathers statistics for the given MediaStreamTrack (or other) selector and reports the result asynchronously.
  • RTCWEB : The IETF working group charged with defining wire protocols and methods to support WebRTC such as communications, security, and firewall and NAT traversal.
  • RTP (RFC 3550) : Real-Time Transport Protocol provides end-to-end network transport functions for transmitting audio, video and data over multicast/unicast network services.
  • SAVPF (RFC 5124) : Secure Audio Video Profile with Feedback combines the profiles of RFC 3711 for SRTP and RFC 4585 for RTCP to enable secure RTP communications with feedback.
  • SBC : A Session Border Controller is a VoIP device that exerts control over signaling and media streams involved in setting up, conducting and tearing down calls.
  • SCTP (RFC 4960) : Stream Control Transmission Protocol provides PSTN over IP (VoIP) communications and applications with strong reliability and variable quality transmission.
  • SDP (RFC 4566) : Session Description Protocol provides a standard representation for conveying media details, transport addresses and other metadata to session participants. See also:
  • security : Precautions taken and resources applied to guard against crime, attack, sabotage, espionage and unwarranted surveillance of WebRTC browser communications. See also:
  • SIP (RFC 3261) : Session Initiation Protocol is an application-layer control protocol used to establish, modify and terminate multimedia exchanges such as VoIP phone calls.
  • SIP-I (ITU-T Q.1912.5) : An ITU-T recommendation that defines the signaling interworking between the Bearer Independent Call Control (BICC) or ISDN User Part (ISUP) protocols and SIP.
  • SIP-T (RFC 3372) : Session Initiation Protocol for Telephones is set of mechanisms used to provide protocol translation and feature transparency for PSTN-SIP interconnections.
  • sipML5 : A pure JavaScript SIP stack using SIP WebSocket Transport for signaling and WebRTC for audio/video that works with no extension, plugin or gateway needed. See also:
  • Skype : A freemium VoIP service released in 2003 and maintained by Estonian developers with ongoing privacy and security issues WebRTC developers must try to avoid.
  • SRTP (RFC 3711) : Secure Real-time Transport Protocol can provide confidentiality, message authentication and replay protection for RTP traffic and related control traffic.
  • SS7 : Signalling System No. 7 is the core set of PSTN communications protocols that enable telephone calls, SMS text messaging, billing and other services. See also:
  • SSE | Eventsource API : Server-Sent Events describes how servers can initiate data transmissions to clients once connection has been established. See also HTML5 push notifications.
  • STUN (RFC 5389) : Session Traversal Utilities for NAT is a protocol and set of methods to allow an end host to discover its public IP address if it is located behind a NAT. See also:
  • TCP (RFC 793) : Transmission Control Protocol enables highly reliable host-to-host exchanges over packet-switched computer networks plus interconnections of such networks.
  • TLS (RFC 5246) : Transport Layer Security allows client/server applications to communicate in a way that is designed to prevent eavesdropping, tampering, or message forgery.
  • TURN (RFC 5766) | TURN (RFC 6156) : Traversal Using Relays around NAT is a protocol that allows a host behind a NAT or firewall (client) to request that another host (server) act as a relay. See also:
  • UC : Unified Communications is the integration of real-time services such as telephony and video with non real-time services such as voicemail and email.
  • UDP (RFC 768) : User Datagram Protocol defines an IP-based, transaction-oriented procedure for applications to exchange messages with other applications across a network.
  • VA4Most: A jQuery/JavaScript FlowPlayer/SoundManager mashup developed in 2010 by Bruce Arnold of WebFL.US to deliver W3C-valid HTML5 video/audio with Flash fallback.
  • VoIP : Voice over Internet Protocol is a suite of technologies for delivery of voice communications and multimedia sessions over IP networks such as the Internet.
  • VP8 (RFC 6386) : The video compression component combined with Vorbis in the WebM audio-video container format for HTML5 video and a part of the WebRTC browser video engine.
  • W3C : The World Wide Web Consortium (est. 1994) helps the Web achieve its potential by developing standards to promote its evolution and assure interoperability.
  • WebRTC : Web Real-Time Communication is a set of APIs that enable browser-to-browser applications for P2P video, voice and file sharing without downloads or plugins. traversal. See also:
  • WebRTC API : A set of ECMAScript APIs in WebIDL to allow P2P media exchanges between browsers being developed by the W3C, IETF RTCWEB group and Media Capture Task Force.
  • WebSocket (RFC 6455) | WebSockets Wire Protocol : WebSocket Protocol enables two-way communication between a client running untrusted code and a remote host that opts in to communications from that code.
  • WebSocket API | HTML5 WebSockets : A W3C specification that defines an API to enable web pages to use the WebSocket protocol defined by the IETF for two-way communication with a remote host.
  • XHR : XMLHttpRequest is a W3C ECMAScript API that provides an easy way to retrieve data from a URL without having to do a full page refresh used heavily in AJAX.
  • XMPP : Extensible Messaging and Presence Protocol is an open technology for real-time communications such as IM, chat, voice, video calls and routing of XML data. See also JavaScript XMPP stacks:
 

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