How Browser-Based Calling Works: WebRTC Explained Simply

You open a website, click a button, and suddenly you’re talking to someone on the other side of the world through your browser. No app download, no installation, no plugins.

How does this actually work? The technology behind it is called WebRTC (Web Real-Time Communication), and it’s pretty fascinating once you understand it.

This guide explains the tech in a way that makes sense, without drowning you in acronyms and engineering speak.

What Is WebRTC?

WebRTC is a technology built into modern web browsers that lets them handle real-time audio and video communication directly.

Think of it like this: Your browser already knows how to display images, play videos, and show text. WebRTC gives it the ability to also handle live conversations.

When Did This Become Possible?

2011: Google releases WebRTC as an open-source project

2013: Chrome and Firefox add WebRTC support

2017: Safari adds WebRTC support (finally)

2020: All major browsers support it

2025: WebRTC is mature, stable, and powers most browser-based calling

Before WebRTC, you needed to install plugins (remember Flash?) or dedicated apps. Now it’s just built into the browser.

How a Browser Call Actually Works

When you make a call from your browser, here’s what happens in the 2-3 seconds between clicking “call” and hearing the other person:

Step 1: Permission Request

The browser asks for microphone access. You click “Allow.”

This only happens once per website. The browser remembers your choice.

Step 2: Connection Setup

Your browser connects to the calling service’s servers. This happens instantly and establishes a secure connection.

Step 3: Audio Capture

Your browser starts capturing audio from your microphone. It samples your voice thousands of times per second.

Step 4: Audio Encoding

The raw audio is compressed using codecs (audio compression algorithms). This makes the data small enough to send over the internet efficiently.

Common codecs: Opus (best for voice), G.711, G.722

Step 5: Packet Creation

The compressed audio is split into small data packets. Each packet is about 20-30 milliseconds of audio.

Step 6: Encryption

Each packet is encrypted using DTLS and SRTP protocols. This prevents anyone from intercepting and listening to your call.

Step 7: Transmission

Packets are sent over the internet to the calling service’s servers.

Step 8: PSTN Connection

The calling service connects to traditional phone networks (PSTN - Public Switched Telephone Network) to reach the person you’re calling.

Step 9: Two-Way Communication

The same process happens in reverse for the audio coming back to you. Both directions happen simultaneously.

All of this happens in real-time with minimal delay.

The Technology Stack Behind It

Browser Layer (What You See)

Clean interface
Call button
Volume controls
Mute button

WebRTC Layer (Built Into Browser)

Handles audio capture
Manages compression
Deals with network issues
Adjusts for packet loss

Signaling Layer (Behind the Scenes)

Sets up connections
Exchanges information between browser and servers
Manages call state (ringing, connected, ended)

Media Server Layer (Service Provider)

Routes audio packets
Connects to phone networks
Handles call recording (if offered)
Manages call quality

PSTN Layer (Traditional Phone Network)

Actually connects to landlines and mobile phones
The “old” phone system that still runs the world
Where your call reaches its destination

Why Browser Calling Didn’t Exist Before

Before WebRTC, browsers couldn’t handle real-time audio for a few reasons:

Technical Limitations

Early browsers weren’t fast enough to process audio in real-time. They would lag, stutter, or crash.

No Standard Protocol

Different browsers did things differently. There was no common way to handle audio.

Security Concerns

Letting websites access your microphone was considered risky. Browsers didn’t have good permission systems.

Plugin Dependency

You needed Flash or Java plugins. These were clunky, had security issues, and didn’t work on mobile.

Network Requirements

Internet connections were too slow and unreliable for real-time audio in many places.

WebRTC solved all of these problems.

How WebRTC Handles Call Quality

Adaptive Bitrate

If your internet connection gets slower, WebRTC automatically adjusts. It reduces quality slightly to keep the call connected.

You might notice this as slight quality changes during a call - that’s WebRTC adapting to your network.

Packet Loss Concealment

If some audio packets don’t arrive (network issues), WebRTC intelligently fills in the gaps. It’s similar to how your brain fills in words when someone talks in a noisy room.

Small gaps are barely noticeable thanks to this.

Jitter Buffering

Network delays aren’t consistent. Packets arrive at slightly different times. WebRTC buffers them briefly to smooth out the irregularities.

This is why there’s a tiny delay (usually 100-300ms) in internet calls.

Echo Cancellation

Your microphone picks up sound from your speakers. WebRTC detects this and cancels it out, preventing the other person from hearing their own voice echoed back.

This works surprisingly well in most cases.

Noise Suppression

Background noise (fans, traffic, keyboard typing) is reduced automatically. The other person hears mostly your voice.

Modern WebRTC implementations use machine learning for even better noise filtering.

Security: How Safe Is Browser Calling?

What’s Encrypted

Audio data: Fully encrypted using SRTP (Secure Real-Time Transport Protocol)

Signaling: Encrypted using TLS/WSS

Metadata: Partially encrypted (who you called, when, duration might be logged)

What This Means

Someone intercepting your internet traffic (like on public WiFi) cannot:

Listen to your conversation
Understand what you’re saying
Record your audio

They might see:

That you’re making a call
To which service
Approximate duration

For most purposes, this is secure enough.

When It’s Not Enough

For extremely sensitive conversations:

Government whistleblowing
Discussing trade secrets
Legal matters under privilege

Use end-to-end encrypted apps like Signal instead. Browser calling goes through servers that can technically access the audio (though reputable services don’t record it).

Browser Compatibility in 2025

Fully Supported (Works Great)

Chrome/Chromium: Full WebRTC support, best performance

Firefox: Full support, excellent privacy features

Safari: Full support (finally), works well on Mac and iOS

Edge: Full support (it’s Chromium-based)

Opera: Full support

Brave: Full support with enhanced privacy

Mobile Browsers

Chrome Mobile: Works perfectly

Safari iOS: Works great (as of iOS 14+)

Firefox Mobile: Full support

Samsung Internet: Works well

What You Need

Browser version:

Chrome 90+
Firefox 88+
Safari 14+
Edge 90+

If your browser is remotely recent (past few years), you’re fine.

Operating system: Any modern OS works - Windows, Mac, Linux, Android, iOS, ChromeOS

Bandwidth Requirements

How much internet speed do you actually need?

Minimum for Acceptable Quality

Download: 100 kbps (0.1 Mbps)
Upload: 100 kbps
Latency: Under 300ms

This is achievable even on 3G mobile data.

Recommended for Good Quality

Download: 500 kbps (0.5 Mbps)
Upload: 500 kbps
Latency: Under 150ms

4G/5G and most WiFi easily exceeds this.

For HD Voice Quality

Download: 1 Mbps
Upload: 1 Mbps
Latency: Under 100ms

This is what most modern connections provide.

What This Means

If you can stream Spotify or watch YouTube without buffering, you can make browser calls with good quality.

Video calls require much more (1.5-3 Mbps), but voice-only is lightweight.

Common Technical Issues Explained

Issue: Echo During Calls

What’s happening: Your microphone is picking up sound from your speakers

Why: Echo cancellation isn’t working properly

Solutions:

Use headphones (best solution)
Lower speaker volume
Move away from speakers
Try different browser

Issue: Robotic or Choppy Audio

What’s happening: Packet loss or insufficient bandwidth

Why: Network issues or browser struggling

Solutions:

Close other bandwidth-heavy apps
Switch to better WiFi or mobile data
Close unnecessary browser tabs
Restart browser

Issue: Delay in Conversation

What’s happening: High latency between you and server

Why: Geographic distance, poor routing, or network congestion

Solutions:

Use VPN to different location (sometimes helps with routing)
Try at different time of day
Switch network (WiFi vs mobile data)

Issue: Cannot Hear Other Person

What’s happening: Audio output issue

Why: Volume muted, wrong output device, or browser issue

Solutions:

Check volume settings
Verify correct speaker selected
Try headphones
Refresh page

Issue: They Cannot Hear You

What’s happening: Microphone access issue

Why: Permissions denied, wrong input device, or hardware issue

Solutions:

Check browser permissions
Select correct microphone in settings
Test microphone in other apps
Try different browser

WebRTC vs Traditional Phone Calls

Traditional Phone Call

Your voice is:

Converted to electrical signal
Sent over phone lines or cellular network
Uses circuit-switching (dedicated connection)
Goes through phone company infrastructure
Arrives at destination

This technology is 100+ years old (with upgrades).

WebRTC Browser Call

Your voice is:

Captured digitally by browser
Compressed and encrypted
Sent over internet in packets
Routes through multiple servers
Connects to phone network
Arrives at destination

This technology is 10-15 years old.

Quality Comparison

Traditional phone: Consistent quality (usually), works without internet, some frequency limitations

WebRTC: Can be better quality (HD codecs), requires internet, adapts to conditions

In good conditions, WebRTC can sound better than traditional phones. In poor conditions, traditional phones are more reliable.

The Future of WebRTC

What’s Coming

Better codecs: Improved compression means better quality at lower bandwidth

AI noise cancellation: Machine learning filtering out even more background noise

Real-time translation: Speak English, they hear Spanish, all in real-time

Better mobile integration: Progressive Web Apps making browser calling feel more like native apps

5G optimization: Specifically tuned for ultra-low latency on 5G networks

What’s Already Here

HD voice: Wideband audio for clearer calls

Opus codec: Best-in-class voice compression

Simulcast: Different quality streams for different network conditions

Screen sharing: Built into WebRTC

Video calling: Though most calling services focus on voice

Why This Matters to You

Understanding WebRTC helps you:

Troubleshoot Issues

When quality is poor, you know whether it’s your internet, your browser, or the service.

Choose Better Services

Services using modern WebRTC implementations with good codecs will sound better.

Set Proper Expectations

You know what quality to expect based on your internet connection.

Make Smart Decisions

You understand why browser calling can be better than apps (no installation) and when traditional phones might be more reliable (no internet required).

Browser Calling vs Apps: Technical Differences

Browser Calling

Uses WebRTC directly in browser
No installation overhead
Updates automatically (service updates their servers)
Cross-platform by default
Same interface everywhere

VoIP Apps

Custom implementation of VoIP
Need installation and updates
Can work offline (some)
Native OS integration
Different interface per platform

Technical advantage of browser calling: Universal compatibility without installation

Technical advantage of apps: Can optimize more deeply, work offline in some cases

For most users, browser calling’s advantages outweigh any technical limitations.

Testing Your Setup

Want to see if your browser and internet are ready for calling?

Test Your Browser

Visit webrtc.github.io/samples/
Try the microphone test
Check if audio works
Verify permissions work

Test Your Internet

Visit fast.com or speedtest.net
Check you have 1+ Mbps up and down
Look for latency under 150ms
Make sure connection is stable

Test A Service

Find a browser calling service with test call
Make a short test call
Verify quality is acceptable
Check if UI works for you

Do this before you actually need to make an important call.

The Bottom Line on the Technology

WebRTC enables browser calling through:

Built-in browser audio handling
Smart compression and encryption
Adaptive quality management
Connection to traditional phone networks

You don’t need to understand all the technical details. What matters:

It works: No installation needed It’s secure: Encrypted by default It’s reliable: Mature technology in 2025 It’s accessible: Any modern browser can do it

The technology keeps getting better, but it’s already good enough for daily use right now.

Want to experience WebRTC-powered calling yourself? NomaPhone uses modern WebRTC technology to deliver crystal-clear calls from any browser. No downloads, no plugins, just open and call. Starting at $0.03/minute. Join the waitlist to try it when we launch.