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What Is WebRTC? How It Works with VoIP

Quick Answer

WebRTC (Web Real-Time Communication) is an open standard that lets web browsers and mobile apps exchange voice, video, and data in real time — without plugins. It complements VoIP by enabling browser-based calling, while the signaling and media still rely on the same underlying IP principles.

What Is WebRTC?

WebRTC is a set of browser APIs and protocols (defined by the W3C and IETF) that establish peer-to-peer media streams directly between endpoints. It powers click-to-call buttons, browser video meetings, and live support widgets.

Definition: WebRTC is a standard that provides real-time audio, video, and data communication directly between browsers or apps using built-in APIs.

How WebRTC Works

  1. The app obtains media from the camera/microphone via browser APIs.
  2. An offer/answer exchange (often over SIP or a custom signaling server) negotiates the session.
  3. Connectivity is established using ICE, STUN, and TURN to traverse NAT.
  4. Media flows via Secure RTP (SRTP) peer-to-peer or through a relay.

WebRTC vs. VoIP

VoIP is the broad concept of voice over IP; WebRTC is a browser-native way to implement real-time media. WebRTC handles capture, encryption, and transport in the browser, while a SIP back end can manage sessions and connect to the telephone network.

Benefits

  • No plugins or downloads for users.
  • Built-in encryption (DTLS-SRTP).
  • Easy integration into web and mobile apps.
  • Works with existing VoIP/SIP infrastructure via gateways.

Frequently Asked Questions

Does WebRTC replace SIP?

Not necessarily. WebRTC often handles the browser side, while SIP connects to the PBX and PSTN.

Is WebRTC secure?

Yes, media is encrypted with DTLS-SRTP by default in compliant implementations.

Can WebRTC call a regular phone?

Through a gateway or SBC that bridges WebRTC to SIP and the telephone network.

Conclusion

WebRTC brings real-time voice and video into the browser, lowering friction for customer communication. It pairs well with VoIP: WebRTC for the browser edge, SIP and an IP PBX for session control and PSTN connectivity. Newrock gateways can bridge WebRTC and SIP environments for unified communications.

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