WebRTC File Transfer: A Complete Guide

Published February 21, 2026 · 10 min read

What Is WebRTC?

WebRTC (Web Real-Time Communication) is an open-source framework that enables real-time communication directly between web browsers and mobile applications. Originally developed by Google and now standardized by the W3C and IETF, WebRTC was initially built for video and audio calling — it's the technology powering Google Meet, Discord voice chat, and Facebook Messenger video calls.

But WebRTC isn't limited to media. Its data channel API allows browsers to send arbitrary binary data directly to each other, making it equally capable as a file transfer mechanism. This is the capability that StreamSnatcher and similar peer-to-peer file sharing tools leverage.

What makes WebRTC significant for file transfer is what it eliminates: no browser plugins (like the now-defunct Flash or Java applets), no native app installations, no server-side storage, and no account registration. If your browser supports WebRTC — and every major browser does since 2017 — you can exchange files directly with another browser.

How WebRTC Data Channels Work

A WebRTC data channel is a bidirectional communication pipe between two browsers. Unlike the audio and video tracks in WebRTC (which use RTP), data channels use SCTP (Stream Control Transmission Protocol) running over DTLS (Datagram Transport Layer Security). This combination was specifically chosen to provide:

• Reliable, ordered delivery — unlike raw UDP, SCTP retransmits lost packets and delivers data in order, similar to TCP. This is essential for file transfers where every byte matters.
• Encryption by default — DTLS encrypts all data channel traffic. This isn't optional — the WebRTC specification mandates it. Every file transferred over a data channel is encrypted in transit.
• Congestion control — SCTP includes built-in congestion control that prevents the data channel from overwhelming the network, dynamically adapting throughput based on available bandwidth and packet loss rates.
• Multiplexing — multiple data channels can run simultaneously over a single peer connection, each with independent reliability and ordering settings.

From a developer's perspective, creating a data channel is straightforward. After establishing a RTCPeerConnection, you call createDataChannel() to create a named channel. The remote peer receives a corresponding channel through the ondatachannel event. Once open, both sides can send and receive ArrayBuffer, Blob, or string data.

Message size limits

WebRTC data channels have a maximum message size that varies by browser implementation — typically between 64 KB and 256 KB. For file transfers, this means files must be split into chunks smaller than this limit. The sender reads the file using the browser's File API, slices it into ArrayBuffer chunks, and sends each chunk through the data channel. The receiver buffers these chunks and reassembles them into the original file.

This chunking approach has a practical advantage: it allows progress tracking. Each received chunk represents a known fraction of the total file size, enabling accurate progress bars and transfer speed calculations.

The Connection Process: Signaling, ICE, STUN, and TURN

Before two browsers can exchange data, they need to find each other and negotiate a connection. This involves several protocols working together:

Signaling

WebRTC intentionally does not define a signaling protocol — it leaves this up to the application. Signaling is the process of exchanging the initial connection metadata between peers before the peer-to-peer connection is established. The metadata exchanged includes:

• SDP (Session Description Protocol) offers and answers — these describe each peer's capabilities: supported codecs, encryption parameters, and network information.
• ICE candidates — potential network paths (IP address + port combinations) that each peer can use to receive incoming connections.

Most WebRTC applications, including StreamSnatcher, use WebSockets for signaling. The signaling server acts as a message relay — it forwards SDP offers, answers, and ICE candidates between peers but never sees the actual file data. Once the peer connection is established, the signaling server's job is done.

ICE: Finding a Path

ICE (Interactive Connectivity Establishment) is the process by which WebRTC systematically discovers and tests potential network paths between peers. ICE generates three types of candidates, tested in priority order:

1. Host candidates — the device's local network addresses (e.g., 192.168.1.5:54321). These work when both peers are on the same local network — the ideal scenario for speed, since data never leaves the LAN.
2. Server reflexive candidates — the public-facing IP address and port as seen by a STUN server. This is the most common successful candidate type for home and mobile networks, where devices sit behind NAT routers.
3. Relay candidates — traffic routed through a TURN server. This is the fallback when direct connectivity fails due to symmetric NATs or corporate firewalls. The TURN server relays encrypted packets between peers but cannot decrypt them.

STUN: Discovering Your Public Address

STUN (Session Traversal Utilities for NAT) is a lightweight protocol that helps a browser discover its public IP address and port mapping. Because most devices sit behind NAT routers (which assign private IPs like 192.168.x.x), the browser doesn't inherently know its public-facing address. STUN servers are simple: the browser sends a request, and the STUN server responds with the source IP and port it observed — the browser's "server reflexive" address.

STUN requests are lightweight (a few hundred bytes) and fast (typically under 50ms). Public STUN servers are operated by Google, Mozilla, and others, and they don't see any file data — only the connectivity probe.

TURN: The Relay Fallback

When direct connectivity fails — commonly in corporate networks with symmetric NATs or aggressive firewalls — WebRTC falls back to TURN (Traversal Using Relays around NAT). A TURN server acts as a relay: both peers send their data to the TURN server, which forwards it to the other peer.

TURN connections are slower than direct connections because data must travel an extra hop through the relay server. However, the critical privacy property remains: the TURN server relays encrypted DTLS packets. It can see that two peers are exchanging data, but it cannot read the contents. The encryption keys are negotiated directly between the peers during the DTLS handshake.

TURN is the mechanism that makes WebRTC work even on the most restrictive networks — a key advantage over simpler peer-to-peer approaches that fail when NATs are uncooperative.

Security: DTLS Encryption in Depth

Security is not an optional feature of WebRTC — it is a mandatory requirement of the specification. Every WebRTC data channel is encrypted using DTLS (Datagram Transport Layer Security), which is the UDP equivalent of TLS (the encryption behind HTTPS).

During the peer connection setup, both browsers perform a DTLS handshake to negotiate encryption parameters. Each browser generates a self-signed certificate, and the certificate fingerprints are included in the SDP exchange. This means both peers can verify that they're talking to the expected party — not a man-in-the-middle — by comparing fingerprints.

The practical implications for file transfer are significant:

• File contents are encrypted before they leave the sender's browser
• No intermediary (including the signaling server, ISP, or TURN relay) can read the data
• Encryption keys exist only in the browsers' memory and are discarded when the connection closes
• There are no encryption settings to configure — it's mandatory and automatic

This is a meaningfully stronger privacy model than most cloud file-sharing services, which encrypt data in transit (TLS to/from their servers) but have access to the decrypted files on their infrastructure.

Browser Support and Compatibility

WebRTC has achieved universal support among major browsers. The data channel API (the part relevant to file transfer) is supported in:

• Chrome — since version 56 (January 2017), including Chrome on Android and iOS
• Firefox — since version 52 (March 2017), including Firefox for Android
• Safari — since version 11 (September 2017), including Safari on iOS
• Edge — Chromium-based Edge (since January 2020) has full support
• Opera — since version 43 (based on Chromium), including Opera for Android
• Brave — all versions, based on Chromium

The only notable exception is Internet Explorer, which Microsoft discontinued in 2022 and which never implemented WebRTC. For all practical purposes, WebRTC data channels are available to approximately 97% of internet users globally, according to Can I Use statistics.

Practical Considerations for WebRTC File Transfer

While WebRTC is a powerful foundation for file transfer, there are practical factors to consider:

Both peers must be online simultaneously

Unlike cloud services where you upload a file and the recipient downloads later, WebRTC requires both peers to have the page open at the same time. This is inherent to the peer-to-peer model — there's no server storing the file for later retrieval.

Network quality affects speed

Transfer speed depends on the network conditions between the two peers, particularly the sender's upload bandwidth (which is typically lower than download bandwidth on consumer connections). On a local network, speeds can be very high because data never leaves the router. Over the public internet, expect speeds comparable to the slower peer's connection.

Background tab throttling

Modern browsers aggressively throttle background tabs to conserve CPU and battery. If a StreamSnatcher tab moves to the background during a transfer, the browser may reduce its processing priority, slowing or pausing the transfer. For best results, keep the tab in the foreground during active transfers.

Memory considerations for large files

The receiving browser buffers incoming file chunks in memory before assembling them into a downloadable file. For very large files (10+ GB), this can consume significant memory. Implementations that stream chunks to disk using the File System Access API can mitigate this, but browser support for streaming writes is still evolving.

Conclusion

WebRTC data channels represent a fundamental shift in how files can be shared between devices. By enabling direct, encrypted, browser-to-browser connections, WebRTC eliminates the need for intermediary servers, cloud storage accounts, and file size limits that have defined online file sharing for decades.

The technology isn't perfect — it requires simultaneous online presence, is affected by network conditions, and faces browser-imposed memory limits for very large files. But for the vast majority of real-world file transfer scenarios, WebRTC provides a faster, more private, and simpler alternative to the upload-wait-download cycle of cloud services.

StreamSnatcher is built entirely on these WebRTC primitives. If you'd like to experience the technology firsthand, start a transfer — it takes under 10 seconds.

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