Browser-Based

A Browser-Based Softphone. No Install Required.

Send a user a link. They open it, log in, and the phone works. No download, no IT ticket, no SIP config to explain. Works in Chrome, Firefox, Edge, and Safari.

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The problem with desktop softphones

Installs are where softphone deployments go to die

Different experience on every machine

The Windows version and macOS version behave differently. One crashes after a system update. The other has a UI that was last updated in 2019. You're supporting two different products.

Updates break things

A new OS release changes something in the audio pipeline. Suddenly every user on macOS has no audio. You spend two days tracking down a workaround and pushing an update that half your users won't install.

IT involvement for every new user

Adding a new user means: send install link, wait for them to install it, walk them through SIP configuration, confirm registration. It takes 30 minutes per user. It shouldn't.

Tied to a specific machine

Remote employees get a call, switch to their laptop, and the softphone isn't there. Desktop apps don't follow people.

The browser softphone difference

Send a link. Done.

The only thing a user needs is a browser. Every modern browser supports WebRTC. Nothing to install, nothing to update, nothing to break.

Chrome Firefox Edge Safari

One link to send

Provision a user in the admin portal. Send them your branded login URL. They open it and log in. No SIP server, no registration URL, no codec settings.

Any device, any location

Works on any device with a modern browser. Remote employees, field staff, and office workers all use the same softphone from wherever they are.

Updates happen automatically

You deploy an update to your Netlify or Railway instance. Every user gets it the next time they refresh. No rollout, no version mismatch, no support calls.

Encrypted audio, every call

WebRTC uses DTLS-SRTP for media encryption on every call. Audio doesn't travel unencrypted.

Performance

It works. Consistently.

Browser-based doesn't mean lower quality. WebRTC is the same protocol used in Google Meet, Zoom, and Teams. The audio quality and reliability are the same.

99.98%
Uptime, last 90 days
<180ms
Avg. API response
500+
Seats deployed
How it works

Getting users on the phone is three steps

Step 01

Connect your PBX

Enter your SIP domain in the admin portal. Stylish Softphone connects via SIP over WSS. Works with Asterisk, FreePBX, FusionPBX, and others.

Step 02

Add users in the portal

Create user accounts, assign extensions, and generate credentials. No CLI. No config files. Done in under two minutes per user.

Step 03

Send them a link

Users open your branded login URL, enter their extension and password, and start taking calls. That's the entire user onboarding.

No per-seat pricing until you scale

Flat-rate plans from $49/mo. All plans include BLF, call park, admin portal, and white-label branding.

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$49
Starter
$99
Growth
$199
Pro

A browser softphone your users will actually open

Free 14-day trial. No credit card. Connect your existing PBX.

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