Reference / glossary
Real-time debugging is often a naming problem first. Use the same word for the same layer, separate the signal path from the media path, and the failure starts to look structural instead of mysterious.
Transport is not topology
WebSocket is a signaling path. WebRTC is a media and data transport. SFU is a topology that uses it.
Signal is not media
Closing the signaling socket does not itself tear down an established peer connection. The media path has its own state.
Encryption is a hop
DTLS-SRTP protects each transport hop. A standard SFU terminates it, so end-to-end confidentiality is a separate design.
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A persistent, full-duplex TCP connection opened by an HTTP Upgrade handshake. Both sides can send messages at any time without a new request.
WebSocket carries signaling messages and application data, but it does not define media transport. WebRTC uses it only for signaling.
The client repeatedly asks the server for new data on a fixed interval. Simple, but wastes requests and adds latency.
The server holds an HTTP request open until data exists or a timeout fires. It emulates push, but each message still needs a new request.
Server-Sent Events · EventSource
A one-way stream from server to browser over plain HTTP. Ideal for live feeds, notifications, and dashboards where the client rarely sends.
A browser API and set of protocols for encrypted, low-latency audio, video, and data directly between peers or through media servers.
peer-to-peer
A direct connection between two endpoints. Media may still relay through TURN, but no application media server processes it.
The out-of-band exchange of session descriptions and network candidates needed before and during a WebRTC connection. WebRTC does not prescribe a signaling protocol.
Session Description Protocol
A text format describing media capabilities: codecs, directions, encryption fingerprints, and ICE credentials. It is negotiated through offer and answer.
An SDP description created by the initiating peer that proposes media sections, codecs, and transport parameters.
The responding peer's SDP that accepts, rejects, or narrows the offer. Both descriptions must be set before ICE can complete.
Interactive Connectivity Establishment
The framework that gathers candidate addresses and tests pairs until it finds a working path between peers.
One possible transport address and protocol a peer can use. Types include host, server-reflexive (srflx), and relay.
Session Traversal Utilities for NAT
A lightweight server that tells a peer its public IP and port as seen from the internet. STUN discovers addresses; it does not relay media.
Traversal Using Relays around NAT
A relay server that forwards encrypted media when direct paths fail. Reliable but bandwidth-intensive and expensive at scale.
Network Address Translation
A router feature that rewrites private addresses to a public address. Different NAT mapping and filtering behaviors break naive direct connections.
Sending ICE candidates as they are discovered instead of waiting for gathering to finish. It speeds up connection setup considerably.
A renegotiation that gathers a fresh set of candidates without tearing down the whole peer connection. Useful after network changes.
The browser API that manages ICE, DTLS, SRTP, media tracks, data channels, and the lifecycle of a WebRTC connection.
The browser API that requests camera and microphone access and returns a MediaStream of live tracks. Requires a secure context and user permission.
The browser API for screen, window, or tab capture. Tracks are usually short-lived and must be stopped manually when sharing ends.
MediaStreamTrack
A single audio or video source, such as one microphone or one camera. Tracks can be added to peer connections and stopped independently.
A sender/receiver pair attached to a peer connection. Transceivers make media direction and m-line reuse explicit.
A media section in SDP, such as audio, video, or application data. The order of m-lines must stay consistent across renegotiation.
The algorithm that compresses and decompresses audio or video, for example Opus, VP8, VP9, H.264, or AV1. Both peers must agree on one.
Sending multiple resolutions of the same video as separate encodings so a server can forward the appropriate quality to each viewer.
Scalable Video Coding
Encoding a video into spatial and temporal layers so receivers or servers can select layers without re-encoding.
Selective Forwarding Unit
A media server that receives each participant's upload once and selectively forwards streams to other participants without decoding or re-encoding.
Multipoint Control Unit
A server that decodes every participant's media, mixes it into one stream, and re-encodes it. Low client bandwidth, high server CPU.
A peer-to-peer topology where every participant connects directly to every other participant. Connection count grows as n(n−1)/2.
Multiple SFUs connected across regions. Each participant uploads to a nearby SFU, and SFUs forward streams between regions.
WebRTC-HTTP Ingestion Protocol
A simple HTTP-based standard for sending a WebRTC stream into a media server or broadcast pipeline.
WebRTC-HTTP Egress Protocol
A simple HTTP-based standard for pulling a WebRTC stream for playback from a media server.
HTTP Live Streaming
Segment-based streaming over HTTP with high CDN scalability but typically several seconds of latency. Good for large audiences that tolerate delay.
RTCDataChannel
A bidirectional channel for arbitrary application data over SCTP, with configurable ordering and reliability. It shares the encrypted peer connection.
Stream Control Transmission Protocol
The transport underneath WebRTC data channels. It provides multiple streams and configurable reliability over DTLS.
Secure Real-time Transport Protocol
The encrypted transport for WebRTC audio and video. Encryption keys are derived during the DTLS handshake.
Datagram Transport Layer Security
A TLS variant for datagrams. It authenticates peers and derives the keys used by SRTP and data channels.
Variation in packet arrival timing. A jitter buffer smooths playout, at the cost of added latency.
Round-trip time
The time for a packet to travel to a destination and back. High RTT makes interactive conversation feel laggy.
Packets that never arrive. Small loss can be hidden by FEC and retransmission; sustained loss causes freezing and robotic audio.
The amount of data used per second by a media stream, usually measured in kbps or Mbps.
BWE
The sender's estimate of available network capacity, used to adapt video quality, resolution, and frame rate in real time.
RTCPeerConnection.getStats
A browser API returning detailed counters and metrics for senders, receivers, candidate pairs, transports, and data channels.
A pattern for handling simultaneous offers with polite and impolite peers, avoiding glare during renegotiation.
A testing and policy mode that uses only relay candidates. It trades direct-path efficiency for predictable connectivity.
Short-lived username and password used to authenticate against a TURN server. Never ship permanent TURN secrets in a browser bundle.
Paired HTTP-based WebRTC standards: WHIP ingests a stream, WHEP plays it back. Both keep WebRTC latency while using ordinary HTTP signalling.
A server that forwards, mixes, records, or processes WebRTC media. SFUs, MCUs, recording services, and gateways are all media servers.
Active speaker detection
Detecting which participant is speaking, using audio levels or server-side analysis, so the UI or SFU can promote that participant.
A gateway that connects WebRTC rooms to traditional phone networks, making a media server a participant in the call.
An HTTPS or localhost origin. Camera, microphone, and many modern browser APIs require it.
Exposure of a participant's network address through ICE candidates or WebRTC metadata. Relay-only policy and TURN can reduce exposure.
Maximum Transmission Unit
The largest packet size a network path can carry without fragmentation. WebRTC data channels must account for DTLS/SCTP overhead.
Looking for the compact version? The cheat sheet pairs each term with the decision it changes.
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