Glossary · VoIP and call quality
WebRTC
In one sentence
WebRTC is an open standard built into web browsers that lets people make voice and video calls from a web page with no plug-in or app to install, with encryption always switched on.
Also known as : Web Real-Time Communication
What is WebRTC?
WebRTC (Web Real-Time Communication) is a set of standards from the W3C and the IETF that gives web browsers direct access to the microphone, camera and network so they can exchange voice, video and data in real time. Before WebRTC, calling from a web page meant installing a plug-in such as Flash or Java, or a separate app.
In business telephony, WebRTC is what makes a webphone work: staff open a browser tab, sign in and take calls without installing anything. It also powers the “call us” buttons in many websites and customer apps.
How it works
A WebRTC call goes through three stages:
- Signalling. WebRTC deliberately doesn’t define how the two parties find each other. Each service picks its own method; when the browser must connect to a phone system, that is often SIP carried over WebSockets.
- Finding a path. The ICE process tests possible routes between the parties. A STUN server tells each browser its public address, and a TURN server relays the media when a firewall blocks a direct connection.
- Encrypted media. Audio travels as SRTP with keys agreed over DTLS. The mandatory codecs are Opus, which adapts to changing network conditions, and G.711, which makes it easy to connect to the traditional phone network.
To reach a regular phone number, a gateway, often a WebRTC-capable session border controller, converts the browser call into a standard SIP call that the carrier then routes.
Example
A London charity runs a helpline staffed partly by volunteers using their own laptops. Asking every volunteer to install and configure software was a support headache. The charity switched to a browser-based phone: volunteers log in through Chrome or Edge, appear as ordinary extensions in the call queue, and answer with a headset. Because media is always encrypted and nothing is stored on the laptop, the trustees were also more comfortable with the data protection side.
WebRTC at Universal Telecom
Our virtual PBX includes a WebRTC webphone, so your team can make and take calls from the browser using the business number. Software companies that want calling inside their own product can build on our developer APIs. If you prefer an installed app with deeper desktop integration, a softphone is also available.
Frequently asked questions
Can WebRTC call ordinary phone numbers?
Not on its own. WebRTC connects browsers; to reach a landline or mobile, the call goes through a gateway that turns it into a SIP call and hands it to a carrier.
Is WebRTC secure?
Encryption is mandatory. The standard requires DTLS for key exchange and SRTP for audio and video, so there is no unencrypted WebRTC mode. Overall security still depends on how the service handles logins and signalling.
Does WebRTC need a fast connection?
A voice call needs very little bandwidth, typically well under 100 kbps each way with the Opus codec. Stable latency and low packet loss matter more than raw speed.
Related terms
- Webphone
A webphone is a business phone that runs in a browser tab with nothing to install. It uses WebRTC to make and receive calls through the company phone system on the business number.
- Softphone
A softphone is a phone application that runs on a computer or smartphone. It replaces the desk phone and connects to the company phone system over VoIP, using the business number.
- SIP
SIP (Session Initiation Protocol) is the standard signalling protocol that sets up, changes and ends VoIP calls. It handles ringing, answering and hanging up; the voice itself travels separately.
- Codec
A codec is the algorithm that digitises and compresses your voice during a VoIP call and turns it back into sound at the other end. It sets both the audio quality and the bandwidth used.
- SBC (session border controller)
A session border controller (SBC) sits at the edge of a network and controls SIP calls going in and out: it filters traffic, hides the internal network, handles NAT and can transcode or encrypt calls.
- Telecom API
A telecom API is a programming interface that lets software control communication services: place or receive calls, manage phone numbers, send SMS, pull call history or get notified in real time when something happens on a call.
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