***

id: c6da0b8a-b482-47f1-bc5d-e3b0270ee292
title: WebRTC
description: What WebRTC is and how SignalWire uses it for real-time communication.
slug: /voice/what-is-webrtc
x-custom:
ported\_from\_readme: true
tags:

* 'product:voice'
* 'product:video'

***

WebRTC (Web Real-Time Communication) is a free, open-source project that adds real-time voice and video capabilities to web browsers and mobile applications. It lets web apps stream audio and video media directly between peers without plugins or third-party software.

WebRTC combines a set of W3C APIs and IETF protocols that enable peer-to-peer data sharing, audio/video capture, and network traversal. SignalWire uses WebRTC as the transport layer for browser-based voice and video calling.

### Key benefits

* **No plugins required** — works natively in all major browsers.
* **Cross-platform** — connects devices regardless of operating system.
* **Encrypted by default** — all media is secured with SRTP (Secure Real-Time Transport Protocol).
* **Adaptive quality** — adjusts bitrate and resolution based on network conditions.
* **Interoperable** — works alongside [SIP](/docs/platform/voice/sip), PSTN, and other voice/video protocols.

### Browser support

All major browsers support WebRTC natively:

* Google Chrome
* Mozilla Firefox
* Safari (macOS and iOS)
* Microsoft Edge
* Opera
* Brave
* Vivaldi

No plugins or extensions are needed — users grant microphone/camera permissions and connect immediately.