The browser is no longer just a window to the web, it’s now a portal for conversation.

Web-based calling is fast becoming a go-to option for business communications. And why not? It’s simple, economical, and saves time, energy, and effort.

But do you know that behind that simple “Click to Call” button lies a complex network of signaling protocols, media streams, and encryption layers that make seamless browser-to-browser or browser-to-phone communication possible.

In this article, we’ll break down how web-based calling works under the hood. Everything from session initiation and audio codecs to the role of servers and APIs. Additionally, this technology is redefining the way developers build communication into modern web applications.

What is Web-based Calling

Web-based calling refers to voice (and video) calls conducted entirely through a web browser, using internet protocols rather than traditional phone lines. In short, web-based calling transforms the browser into a full-fledged communication endpoint.

It can handle everything from call initiation to media streaming, while integrating easily with APIs, CRMs, and customer service tools.

This is a form of VoIP because the audio travels over the internet. Still, it leverages WebRTC to work plugin-free and without dedicated software installs. WebRTC (Web Real-Time Communication) is the key HTML5 technology that enables real-time audio and video in browsers, supported natively by Chrome, Firefox, Safari, Edge, and others.

How Web-based Calling Works

With a SIP (Session Initiation Protocol) client running in the web page (and RTP that carries the media), the browser can register to a typical softphone or PBX just like any other phone, allowing users to connect through a URL.

1. WebRTC in the Browser

WebRTC provides the browser APIs that handle real-time media. Using JavaScript, a web app invokes WebRTC’s getUserMedia API to access the user’s microphone (and camera for video).

The audio or video data is then fed into a peer connection (RTCPeerConnection), responsible for negotiating a direct media path to the remote party and streaming the media in real-time.

The WebRTC engine in the browser encodes the audio and sends it over the network. It also handles decoding incoming audio and playing it through the user’s speakers. Here, the browser is acting as the VoIP phone’s media base, capturing and playing voice in real time.

2. Signaling and Call Control

Here’s how a WebRTC softphone registers and handles calls (SIP over WSS):

• The browser opens a secure WebSocket (WSS) to the SIP server (cloud PBX/proxy) and sends a SIP REGISTER with the same credentials an IP phone would use. The PBX authenticates and keeps the WSS session open.

• The web app sends a SIP INVITE (carrying SDP/ICE details) over the same WSS. The PBX completes the setup by applying its dial plan (IVRs, ring groups, or PSTN breakout).

• For inbound, the PBX pushes a SIP INVITE down the WSS. The browser rings, and the user answers.

• Mute/hold/transfer use standard SIP (re-INVITE/UPDATE/INFO). BYE ends the session.

It’s a persistent, TLS-encrypted transport that browsers can use. SIP messages are unchanged (only the transport is WebSocket), so web softphones plug cleanly into existing VoIP/PBX systems.

3. Session Negotiation (SDP Offer/Answer)

Before a web-based VoIP call begins, both the browser and the remote endpoint must agree on how they will exchange media. This process happens through the Session Description Protocol (SDP), which defines details like codecs, ports, and encryption keys.

In web-based calls, the browser sends an SDP offer and receives an answer, similar to how SIP phones exchange information via INVITE and 200 OK messages. WebRTC adds modern, security-first, and NAT-smart details into that contract. A browser’s SDP usually includes ICE candidates and DTLS/SRTP attributes (so media is encrypted by default).

Modern SIP platforms (Asterisk with PJSIP, FreeSWITCH) parse these just fine by negotiating ICE, accepting encrypted media, and bridging to the rest of your VoIP phone system or PSTN as needed. Older SIP gear, however, may choke on these attributes or lack SRTP/ICE support, so either enable WebRTC features on the PBX or place a WebRTC gateway in front to translate.

4. Media Transport (RTP and SRTP)

Once the call is established, the actual voice packets stream via SRTP. In WebRTC-powered softphones, SRTP is mandated, meaning the audio packets are encrypted during transmission using keys negotiated via DTLS (Datagram TLS).

• Encryption: Every WebRTC call uses SRTP, encrypting media packets in transit.

• Compatibility: WebRTC-ready PBXs terminate DTLS/SRTP and bridge audio internally.

• Transcoding: If a call connects to the PSTN or a legacy SIP endpoint, the PBX converts codecs and decrypts or re-encrypts as required.

• Quality handling: WebRTC continuously manages jitter, packet loss, and echo to maintain stable, high-definition voice quality.

Web-based calls use the same RTP foundation as VoIP, but with built-in encryption, modern codecs, and adaptive audio optimization for a secure, high-quality user experience.

5. NAT Traversal (ICE, STUN, TURN)

One major hurdle is NAT and firewall traversal, especially when browsers sit behind private networks and can’t be reached directly by the cloud PBX system. Web call solves this through the ICE framework, automatically finding a viable media path between the caller and callee.

• STUN (Session Traversal Utilities for NAT): Determines the browser’s public IP and port by querying an external STUN server.

• TURN (Traversal Using Relays around NAT): Acts as a relay when direct routes fail, forwarding encrypted media through a dedicated relay server.

• Automatic selection: ICE intelligently decides whether to send media directly to the PBX or via TURN, all within milliseconds during setup.

During call setup, the browser gathers multiple ICE candidates (potential network routes) and tests them to select the best working path for media flow. Deploy geographically close TURN servers for remote teams to reduce latency and ensure reliable connectivity.

The Future of Business Communication is Browser-Based

Web-based calling represents a major shift in how we think about voice communication. By leveraging WebRTC, SIP, and VoIP technologies, developers can now embed real-time calling capabilities directly into browsers, without relying on traditional telecom infrastructure.

It’s a lightweight, secure, and scalable communication layer that’s redefining what’s possible in modern web experiences. Whether you’re building customer support tools, collaboration apps, or unified communication platforms, understanding how web-based calling works opens the door to endless innovation in business communication.