What is WebRTC?

WebRTC (Web Real-Time Communication) is an open-source standard that lets web browsers and apps exchange audio, video, and data directly between devices — with no plugins, downloads, or extensions. Built into every major browser, it powers the “join from your browser” experience in video meetings, click-to-call buttons, telehealth visits, and in-app voice and video. WebRTC is a core technology behind modern VoIP and browser-based softphones.

How WebRTC works

WebRTC establishes a peer-to-peer media path between two browsers or apps, then streams encrypted audio and video over it. Getting there involves three jobs:

• Signaling: the two sides exchange session details (codecs, network candidates) through a server the application provides. WebRTC deliberately leaves the signaling channel up to the developer.
• NAT traversal: because most devices sit behind routers and firewalls, WebRTC uses ICE, STUN, and TURN servers to discover a reachable path between the peers.
• Media transport: once connected, audio and video flow over SRTP, which is encrypted by default.

Every WebRTC session is encrypted end-to-end on the media path, which is why it is considered secure by design.

What makes WebRTC different

• No installation: it runs inside the browser, so users join calls from a link instead of installing software.
• Open and royalty-free: any developer can build with it without licensing fees.
• Low latency: the peer-to-peer model keeps voice and video responsive, important for natural conversation.
• Cross-platform: the same code works across desktop and mobile browsers and can be wrapped into native apps.

WebRTC vs. VoIP and SIP

WebRTC and traditional VoIP both carry voice over IP, but they come from different worlds:

• Traditional VoIP often uses the SIP protocol and dedicated softphone apps or desk phones. It connects readily to the phone network through SIP trunking.
• WebRTC is browser-native and excels at in-app and web calling, but needs a gateway to reach regular phone numbers.

In practice, modern platforms combine both: WebRTC for the browser experience, SIP and trunking to bridge calls onto the public phone network.

Common WebRTC use cases

• Browser-based video meetings with no app download
• Click-to-call and live support widgets on websites
• Telehealth and online consultation platforms
• In-app voice and video for marketplaces and social apps
• Contact center agent consoles that run entirely in a browser

WebRTC frequently asked questions

Is WebRTC the same as VoIP?

WebRTC is a technology for real-time audio, video, and data between browsers and apps; VoIP is the broader practice of carrying voice over IP. WebRTC can be used to build VoIP experiences, especially browser-based calling, but VoIP also includes SIP-based desk phones and softphones that do not use WebRTC.

Does WebRTC require a plugin or download?

No. WebRTC is built directly into modern browsers, so users join calls from a web link without installing plugins or apps. That zero-install experience is one of its biggest advantages for guest meeting join and web support widgets.

Is WebRTC secure?

Yes. WebRTC encrypts media by default using SRTP, and the data channel uses DTLS. Encryption is mandatory in the standard, not optional, so audio and video between peers are protected in transit.

Can WebRTC call regular phone numbers?

Not on its own. WebRTC connects browsers and apps peer-to-peer. To reach a landline or mobile number, the call passes through a media gateway and SIP trunking that bridge WebRTC onto the public telephone network.

See how DialPhone uses WebRTC

DialPhone’s browser app and video meetings use WebRTC so anyone can make calls or join meetings straight from a browser — no downloads — while SIP trunking connects those sessions to any phone number worldwide.