Glossary
WebRTC (Web Real-Time Communication)
The open standard and browser API for real-time audio, video and data shared directly between browsers and apps, with no plugins. WebRTC powers most in-browser voice and video calling.
General definition
WebRTC, or Web Real-Time Communication, is an open standard and set of browser APIs for sending audio, video and arbitrary data in real time between two endpoints. It is maintained jointly by the W3C and the IETF and is supported natively in every major browser, so no plugin or download is required.
A WebRTC session is built from a few core pieces:
- getUserMedia captures the microphone and camera
- RTCPeerConnection negotiates and carries the encrypted audio and video streams
- RTCDataChannel sends low-latency arbitrary data such as chat or game state
- STUN and TURN servers help endpoints find a path to each other through firewalls and NAT
- A signaling channel, which WebRTC deliberately leaves undefined, exchanges the initial connection details
For calls with more than a couple of participants, a media server such as an SFU (Selective Forwarding Unit) routes streams efficiently instead of every participant sending to every other.
WebRTC vs other real-time approaches
WebRTC is peer-oriented and built for live media, which sets it apart from request-and-response HTTP or WebSocket messaging. Chat and presence often ride on WebSocket or XMPP, while the actual voice and video flow over WebRTC. The two are complementary: one carries the conversation, the other carries the call.
WebRTC in the Ethora ecosystem
Ethora’s voice and video calling is built on WebRTC. The platform handles the media layer while XMPP and the Ethora Chat Protocol provide signaling and presence, so developers get calls, chat and AI agents from one SDK rather than stitching several services together.
See the voice and video call API and Chat SDK for how calling fits alongside real-time messaging.
