What Is WebRTC? How Browser-Based Calling Works (2026 Guide)

WebRTC (Web Real-Time Communication) is an open-source technology built into all major web browsers that enables real-time audio, video, and data communication without requiring any plugins, downloads, or external software. Originally developed by Google and standardized by the W3C and IETF in 2021, WebRTC is the technology behind browser-based calling services, video conferencing platforms like Google Meet, and peer-to-peer communication tools. As of 2025, WebRTC reaches over 4.7 billion devices worldwide, according to WebRTCHacks, making it the most widely deployed real-time communications standard in history.

This guide explains how WebRTC works, why it has become the standard for browser-based calling, and how services like Kinvo use it to provide international phone calls from your browser at a fraction of traditional costs.

How WebRTC Works: A Technical Overview

WebRTC is a collection of APIs and protocols that handle three core functions: capturing media (audio/video from your device), establishing a connection (peer-to-peer or through a server), and transmitting data (with encryption and quality optimization).

Step 1: Media Capture

When you initiate a call, WebRTC uses the getUserMedia API to access your device's microphone. This works on any device — laptops, desktops, tablets, and smartphones — as long as it has a microphone and runs a modern browser. No driver installation or special hardware is needed.

Step 2: Audio Encoding

Your voice is captured and encoded using the Opus audio codec. Opus is an open-source codec specifically designed for real-time speech. It operates at variable bitrates from 6 kbps to 510 kbps and captures frequencies up to 20 kHz — compared to traditional phone networks that are limited to 3.4 kHz. This is why WebRTC calls sound noticeably clearer than regular phone calls.

Step 3: Encryption

WebRTC mandates encryption for all communication. Audio streams are encrypted using SRTP (Secure Real-time Transport Protocol), and signaling data uses DTLS (Datagram Transport Layer Security). This is not an optional feature — the WebRTC specification requires encryption by default. Every WebRTC call is encrypted end-to-end, which is a significant security advantage over traditional phone calls that travel unencrypted over the PSTN.

Step 4: Network Traversal

Establishing a connection between two endpoints across the internet is complicated by firewalls and NAT (Network Address Translation). WebRTC uses ICE (Interactive Connectivity Establishment), STUN (Session Traversal Utilities for NAT), and TURN (Traversal Using Relays around NAT) protocols to find the optimal path between caller and recipient — even when both are behind firewalls.

Step 5: Telephony Gateway (for Phone Calls)

For calling regular phone numbers (not just browser-to-browser), the WebRTC audio stream is routed to a telephony gateway — a server that bridges the internet and the traditional phone network (PSTN). The gateway converts the digital WebRTC audio into a standard phone signal and routes the call to the destination number. The recipient's phone rings normally, and they cannot tell the call originated from a browser.

WebRTC vs Traditional VoIP: What Is the Difference?

Traditional VoIP (like the technology Skype used) requires downloading and installing a separate application. The app handles audio capture, encoding, networking, and connecting to the phone network. WebRTC moves all of this functionality into the browser itself.

Feature	Traditional VoIP (App-Based)	WebRTC (Browser-Based)
App download required	Yes	No
Plugin or extension needed	Sometimes	Never
Works on all devices	Requires platform-specific apps	Yes — any device with a browser
Audio codec	Varies (often proprietary)	Opus (open source, HD)
Encryption	Varies by provider	Mandatory (SRTP + DTLS)
Latency	Moderate (100–300ms)	Low (50–150ms typical)
Maintenance	App updates required	Browser handles updates automatically

The practical impact: WebRTC eliminated the biggest friction point in VoIP adoption — requiring users to download and maintain a separate application. This is why Skype declined while browser-based services grew.

Browser Support for WebRTC (2026)

WebRTC is supported by every major browser as of 2026:

• Google Chrome — full support since 2012 (version 28+). Chrome has the most mature WebRTC implementation and is recommended for the best experience.
• Mozilla Firefox — full support since 2013 (version 22+).
• Apple Safari — full support since 2017 (version 11+, iOS 14.5+). Earlier versions had partial support.
• Microsoft Edge — full support (Chromium-based since 2020).
• Opera, Brave, Vivaldi — all Chromium-based, full support.

According to Can I Use, WebRTC APIs have 97.8% global browser support coverage as of 2025, meaning virtually every internet user can make WebRTC-based calls without installing anything.

Why WebRTC Makes International Calls Cheaper

The cost difference between WebRTC-based international calls and traditional carrier calls comes down to network routing:

• Traditional carrier call (US to UK): Your carrier routes the call through multiple international trunk lines and gateway exchanges. Each carrier in the chain — your local carrier, one or more international transit carriers, and the destination carrier — takes a margin. Result: $0.15–$1.50 per minute.
• WebRTC call (US to UK): Your voice travels over the internet (which you already pay for) to a telephony gateway located in or near the UK. The gateway makes a local call to the destination number. You are paying for one local call at the destination, not international transit. Result: $0.01–$0.03 per minute.

The key insight: WebRTC leverages the internet to bypass the expensive international transit layer. The call only touches the traditional phone network at the "last mile" — in the destination country — which is a local call, not an international one.

Quality and Reliability of WebRTC Calls

Modern WebRTC implementations include several technologies that maintain call quality even on imperfect internet connections:

• Adaptive bitrate: WebRTC automatically adjusts audio quality based on available bandwidth. If your connection drops, the codec reduces bitrate to maintain the call rather than dropping it.
• Jitter buffering: Internet packets sometimes arrive out of order or with variable delays. WebRTC includes a jitter buffer that smooths out these inconsistencies.
• Forward Error Correction (FEC): The Opus codec includes redundancy in the audio stream, allowing the receiver to reconstruct lost packets without retransmission.
• Echo cancellation: Built-in acoustic echo cancellation removes feedback loops when using speakers instead of headphones.
• Noise suppression: Background noise is filtered in real time using signal processing algorithms.

The minimum recommended internet speed for WebRTC calling is 100 kbps upload — achievable on virtually any modern connection including 4G mobile data. A typical WebRTC voice call uses approximately 1 MB per minute of audio data.

How Kinvo Uses WebRTC for International Calls

Kinvo is a browser-based international calling service built entirely on WebRTC. When you make a call through Kinvo, the process works as follows:

1. You open the Kinvo dialer in your browser and enter an international phone number
2. Your browser captures audio from your microphone using the WebRTC getUserMedia API
3. The audio is encoded with the Opus codec and encrypted with SRTP
4. The encrypted audio stream is sent to Kinvo's telephony infrastructure (powered by Telnyx with Twilio failover)
5. The telephony gateway converts the WebRTC audio to a standard phone signal
6. The call is routed to the destination phone number over the local phone network
7. The recipient's phone rings — they see your caller ID and answer normally

The entire process — from clicking "call" to the recipient's phone ringing — takes under 3 seconds. All audio is HD quality and encrypted end-to-end. Kinvo's dual-provider infrastructure (Telnyx primary, Twilio failover) provides 99.9% uptime by automatically switching providers if one experiences issues.

Frequently Asked Questions

Is WebRTC free?

WebRTC is a free, open-source technology. Using it in your browser costs nothing. However, connecting a WebRTC call to a regular phone number requires a telephony gateway, which incurs per-minute charges — this is what services like Kinvo charge for. Browser-to-browser WebRTC calls (like Google Meet) are free because they do not touch the phone network.

Is WebRTC secure?

Yes. WebRTC mandates encryption for all communication. Audio streams use SRTP encryption and signaling uses DTLS. This makes WebRTC calls more secure than traditional phone calls, which are not encrypted by default.

Does WebRTC work on mobile phones?

Yes. WebRTC works in mobile browsers (Chrome, Safari, Firefox on iOS and Android). You do not need to download an app — just open the calling service in your mobile browser.

What internet speed do I need for WebRTC calling?

A minimum of 100 kbps upload speed is recommended. This is available on virtually any modern connection, including 4G mobile data, hotel Wi-Fi, and home broadband. For reference, a WebRTC voice call uses approximately 1 MB per minute.

Key takeaway: WebRTC is the open-source technology that makes browser-based phone calls possible. Built into every major browser with 97.8% global coverage, it provides HD audio quality, mandatory encryption, and low latency — without any downloads or plugins. For international calling, WebRTC-based services like Kinvo leverage this technology to deliver calls at $0.01/min by routing over the internet and only connecting to the phone network at the destination, bypassing expensive international transit carriers.