SIP over WebSocket

Browser SIP Calling over WebSocket, Priced by the Minute

Dial straight from a browser tab and pay only for the minutes you use. Nothing to install, and a number is needed only when you want to receive inbound calls.

Product facts

  • Outbound: No dedicated number needed.
  • Inbound: Requires US/Canada digital number ($2.14 to $5/month).
  • No apps: Works in Chrome, Safari, Edge, Firefox.
How it works
Browser SoftphoneReady

What is SIP over WebSocket?

With SIP over WebSocket, ordinary SIP signaling travels inside WebSocket frames, letting up to date browsers register as SIP clients. Pair that with WebRTC media and you get genuine telephony running straight in the browser.

Native to the Browser

Skip plugins and installers. Chrome, Firefox, Edge, and Safari can place and receive SIP calls straight from the web, while Intercalling manages the SIP over WebSocket client stack on your behalf.

Global PSTN Reach

Intercalling links WSS to Twilio and SIP trunks, so calls from your browser connect to any phone number on earth. Handle virtual numbers and inbound routing from the dashboard.

AI and Analytics

Switch on live copilot assistance, transcriptions, and real-time analytics over SIP over WebSocket to drive smarter voice workflows.

How Intercalling delivers SIP over WebSocket

A standard call path links the browser to the PSTN with secure WebSocket signaling and WebRTC media. Intercalling coordinates every piece, so calling stays your only concern.

Step by step

  1. The browser opens a secure WebSocket (WSS) to the Intercalling signaling gateway.
  2. Intercalling converts that WSS SIP signaling into standard SIP messages for Twilio or your own SIP trunk.
  3. WebRTC negotiates the media, which then routes through Intercalling or Twilio as needed.
  4. Twilio bridging carries the call to PSTN numbers, and call events flow back to power analytics and AI features.

Built for Developers

Intercalling hands you tokens and SDKs to register browser clients over SIP over WebSocket. Wire it up fast and subscribe to webhooks for call events, recordings, and transcriptions.

Immediate browser registration and dialing.
Worldwide PSTN connectivity through Twilio and SIP trunks.
Encrypted WSS connections and SRTP media keep things private.

Secure and Reliable Calling

SIP over WebSocket travels over TLS, and Intercalling protects media with SRTP. We connect through Twilio and run hardened gateways to keep calls intact and available.

Encrypted Signaling

WSS over TLS protects every SIP message moving between browser and gateway.

Secure Media

SRTP scrambles audio streams so your media stays private.

Monitoring

Live analytics track call quality and usage, with alerts when something needs attention.

Ways Teams Put It to Work

With nothing more than a browser, SIP over WebSocket opens up fresh workflows across sales, support, and voice automation.

Contact Centers

Agent consoles live in the browser, complete with live AI assistance and call recording.

Remote Teams

Sales and support staff place secure PSTN calls from any device, no softphone required.

Voice Automation

Put the Intercalling voice AI stack to work handling inbound or outbound SIP calls automatically.

Frequently Asked Questions

Which browsers can run SIP over WebSocket?

Current browsers such as Chrome, Firefox, Edge, and Safari all support WebSocket and WebRTC. Intercalling checks the common setups so compatibility is covered.

Does my firewall need special ports opened?

Configured with WSS, SIP over WebSocket rides standard HTTPS ports, which keeps firewall headaches to a minimum. Media uses RTP across standard ranges, all documented by Intercalling for your network setup.

Will my current SIP provider still work?

Yes. Intercalling can bridge WSS SIP signaling to Twilio and to many SIP trunks, letting you keep the SIP infrastructure you already have while adding browser endpoints.

How does voice AI work with SIP calls?

Intercalling can run autonomous voice agents for outbound campaigns, or act as a live copilot mid call by transcribing audio and proposing responses in real time.

Ready to take SIP into the browser?

Create an account to switch on SIP over WebSocket and get instant browser based calling, AI features, and worldwide PSTN reach at pay as you go rates.

Making an international call from a laptop

Call from any browser

No app and no SIM. Place the call from your computer or phone in seconds.

World map of Intercalling calling coverage

Low rates to 190+ destinations

Clear per minute pricing, shown before you dial. Pay as you go, no monthly fee.

Calling from a phone and a laptop

One balance, every device

Start on your laptop, continue on your phone, all at the same low rate.

SIP Over WebSocket: How Browser Calling Works Without an App

In short

SIP over WebSocket means carrying SIP call signaling inside a WebSocket connection (the SIP WSS transport) so a web browser can place and receive calls without any plugin or softphone. Intercalling uses this same browser native stack, so you make outbound calls in Chrome, Safari, Edge, or Firefox with no app download, no SIM, and no phone number, billed per minute.

What SIP over WebSocket actually does

SIP (Session Initiation Protocol) sets up, manages, and tears down calls. Traditionally it travels over UDP or TCP, which browsers cannot open directly. SIP over WebSocket wraps those same SIP messages inside a WebSocket, normally the secure WSS variant on port 443, so the signaling rides the same encrypted channel a website already uses. The browser then negotiates the actual audio with WebRTC.

In short, WebSocket handles the call setup messages and WebRTC handles the live voice. Together they let a plain tab register, dial, and ring with no installed software. This is exactly why a service can offer calling from a URL instead of a download.

How Intercalling puts it to work

Intercalling is built on this browser native model, so you never touch SIP configuration yourself. You open the site, and the page handles the signaling and media for you. Outbound calls need no phone number and are pay as you go, billed per minute, which suits international and ad hoc calling.

If you also want to receive calls, you add a US or Canada digital number for roughly $2.14 to $5 per month, and inbound calls ring straight in your browser. All voice is end to end encrypted, which matters for business and privacy focused calling.

Frequently asked questions

Is SIP over WebSocket the same as WebRTC?+

No. They work together. SIP over WebSocket carries the call control and signaling messages, while WebRTC carries the encrypted audio. A browser dialer typically uses both at once.

Do I need to set up a SIP client or softphone to use Intercalling?+

No. Intercalling runs entirely in the browser. There is no softphone to install, no SIP credentials to enter, and no SIM card. You just open the site in Chrome, Safari, Edge, or Firefox and call.

Why use WebSocket instead of regular SIP ports?+

Browsers cannot open raw UDP or TCP SIP sockets for security reasons. WebSocket, especially the WSS form on port 443, passes cleanly through firewalls and proxies, so calling works on networks that block traditional SIP.

Are calls encrypted?+

Yes. Intercalling calls are end to end encrypted, and the WSS signaling layer is secured as well, which makes the setup well suited for business and privacy sensitive calls.

Last reviewed June 2026Reviewed by the Intercalling calling teamDialing rules cross checked against ITU international dialing procedures
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