WebRTC (Web Real-Time Communication)

What is WebRTC (Web Real-Time Communication)

WebRTC (Web Real-Time Communication) is the open-source standard that lets a browser or app handle audio, video and data directly, peer-to-peer, without installing a plugin or a separate app. It’s the tech running quietly under Google Meet, in-browser click-to-call widgets on e-commerce sites, and telehealth apps where a patient just clicks a link and starts a video call. Where VoIP is the broad umbrella for calling over the internet, WebRTC is one specific, browser-native way of doing it – no SIP client, no download, just a URL.

How does WebRTC work

WebRTC works through a set of JavaScript APIs built straight into modern browsers – Chrome, Firefox, Safari, Edge – so there’s nothing for the user to install. When two browsers want to talk, they first exchange “signaling” messages (usually over a regular WebSocket or HTTPS connection you build yourself) to agree on things like codecs and session details. Then comes the tricky part: getting two devices, often sitting behind different home routers and NAT firewalls, to find a direct path to each other. That’s what STUN and TURN servers are for – STUN helps a device discover its public IP so peers can connect directly, and TURN steps in as a relay when a direct connection isn’t possible. Once that peer connection is set up, audio, video and data flow encrypted and directly between browsers, with no server in the middle of the actual media – which is why WebRTC calls tend to feel faster and cheaper to run at scale than server-relayed alternatives.

How OneXtel helps with WebRTC

Here’s the honest bit: in-browser, peer-to-peer calling over WebRTC isn’t the layer OneXtel’s Voice stack runs on, and we’d rather tell you that upfront than oversell an SDK we don’t have. OneXtel’s Smart IVR is SIP-based telecom voice infrastructure – the same kind of setup you’d use for SIP Calling or SIP Trunking – reached over standard telephony networks rather than a browser API. So if you’re building a Google-Meet-style click-to-call widget that runs entirely inside a webpage with no phone number involved, that’s a WebRTC SDK’s job, not ours. What OneXtel does give you is the voice layer businesses actually build call flows on: Smart IVR for routing calls, collecting input and running voicebot-style menus, plus voice OTP as a fallback channel alongside SMS, WhatsApp and email through the Aura API. If your product needs telecom-grade voice – inbound support lines, OTP-by-call, automated outbound reminders – that’s squarely what OneXtel’s Voice/IVR product handles, even though it isn’t the same thing as a browser WebRTC widget.

Why it matters

WebRTC matters because it removed the biggest reason people used to bounce off click-to-call and video features: having to download something first. A support widget, a telehealth consult, a live-shopping video call – all of it now just opens in a tab, which is a big reason conversion rates on browser-based calling features tend to beat app-download flows. For businesses picking a communication strategy, it also matters to know where WebRTC fits versus telecom-grade voice: WebRTC is great for browser-to-browser sessions where both people are already on a webpage, but it doesn’t reach a plain phone number the way a SIP call or an IVR system does. So a delivery company running OTP verification or a bank running an automated collections call still needs telephony-grade voice, not a browser API – and that’s exactly the gap SIP-based infrastructure like OneXtel’s Smart IVR is built to cover, working alongside WebRTC rather than replacing it.

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