Telecoms glossary • W
WebRTC (Web Real-Time Communication)
The reason a call can happen in a browser tab with nothing installed. It carries media; it does not replace SIP.
What is WebRTC?in VoIP and business telephony
1. WebRTC is a set of open standards that let a web page use the microphone and camera and carry live audio and video directly, with no plugin or download. It is a W3C Recommendation with the underlying protocols specified by the IETF, and it is what makes calling from inside a browser work.
2. WebRTC deliberately leaves signalling out of scope, so it is not an alternative to SIP. Business deployments usually pair the two, signalling with SIP over WebSockets and carrying the media with WebRTC.
How a browser makes a call
The browser already contains everything needed to run a call. WebRTC is the set of interfaces that lets a page use it, plus the protocol suite the IETF specified underneath.
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The page asks for the microphone
The user grants permission once per site. Nothing is installed, and nothing runs when the tab is closed.
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Signalling happens outside WebRTC
The two ends must exchange details about the session before media flows. The W3C specification is explicit that this is provided by unspecified means, generally by a script in the page talking to a server. In business platforms that script usually speaks SIP over a WebSocket, standardised as RFC 7118.
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ICE finds a path
STUN and TURN are used to work around NAT, exactly as they are for any other VoIP client. Where a direct path cannot be found, media is relayed through a TURN server that somebody has to run and pay for.
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Media flows, encrypted, always
RFC 8827 requires that all media channels be secured with SRTP and that traffic must not be sent over unencrypted RTP. Encryption is not a setting in WebRTC; unencrypted media is simply not a valid configuration.
WebRTC does not compete with SIP
This is the misconception worth killing. The two standards solve different halves of the same call and are routinely deployed together.
| SIP | WebRTC | |
|---|---|---|
| What it defines | Signalling: setting up, changing and ending a session | Media capture and transport in a browser |
| Signalling protocol | Its own | None, deliberately left to the application |
| Media encryption | Optional, via SRTP | Mandatory, DTLS-SRTP |
| Where it runs | Handsets, apps, servers | Browsers and embedded engines |
| Typical business use | Registering extensions | Giving one of those extensions a browser client |
A browser agent console on a South African Cloud PBX is normally a SIP extension whose media happens to run through WebRTC. Nobody replaced anything. The provider added a way to reach the same extension without installing a client.
Where it earns its place
Three deployments account for most business use, and the value in each is the absence of an installer.
- Browser agent consoles
- Agents sign in on any machine and work their queue with no software deployment. Useful where laptops are locked down by IT policy, and useful for short-term or seasonal staff who will never have a device provisioned.
- Click-to-call on a website
- A visitor speaks to sales from the page they are reading, without dialling and without cost to them. It works, but treat it as an addition to a published number rather than a replacement, because many buyers still want a number they can save.
- Video meetings
- The mainstream browser meeting platforms are built on it. This is the deployment most people have used without knowing the name.
The limits are worth stating plainly. A browser tab shares a machine with everything else the user is running, so audio quality is less consistent than a dedicated client or a desk phone. There are no physical keys for a receptionist. And a browser update can change behaviour under you, which is a maintenance model rather than a fault. For an agent taking calls all day, a softphone or a desk phone remains the steadier choice.
Ask before you rely on it
Ask whether the browser client is feature-equivalent to the installed one. Transfer, queue login, presence, call recording and hold music are the functions most often missing from a first-generation web client, and the gap is rarely advertised.
Then ask who runs the TURN relay and what happens on a restrictive corporate network. When UDP is blocked, WebRTC needs a relay to fall back to, and the answer to who pays for that relay and where it sits geographically determines both call quality and whether the browser client works from a client site at all.
About this entry
Definitions are written for South African business buyers and checked against primary sources – provider documentation, standards bodies and ICASA – not vendor marketing. Reviews are independent and sponsors are always disclosed. Read our editorial policy and scoring methodology.
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