How WebRTC File Transfer Works

WebRTC (Web Real-Time Communication) is an open web standard that enables browsers to exchange audio, video, and arbitrary binary data directly peer-to-peer. Understanding the underlying protocol handshake reveals how browser file transfer achieves high speeds without cloud servers.

The three phases of a WebRTC transfer session

1. Signaling (The Introduction)

Before two browsers can connect directly, they must know each other's network coordinates. This is handled by a lightweight signaling server:

• Browser A creates a session room and generates an SDP (Session Description Protocol) offer containing supported codecs and cryptographic keys.
• Browser B joins the room and responds with an SDP answer.
• Both browsers exchange ICE (Interactive Connectivity Establishment) candidates, which list their local IP addresses and port bindings.

2. Peer connection & DTLS handshake

Once ICE candidates are exchanged, the signaling server's job is finished. The two browsers initiate a direct UDP connection. They negotiate a DTLS (Datagram Transport Layer Security) handshake, deriving temporary cryptographic session keys. This ensures all subsequent packets are encrypted end-to-end.

3. SCTP DataChannel streaming

On top of the secure DTLS tunnel, browsers open an SCTP (Stream Control Transmission Protocol) DataChannel. SCTP combines the speed of UDP with the guaranteed packet delivery and ordered sequencing of TCP. Files are read from storage, split into chunks, transmitted across the DataChannel, and reassembled on the receiver's disk.

Frequently asked questions