LayerLink
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LayerLink Viewer

Welcome to the LayerLink Client Terminal. This secure web utility allows users to view remote broadcasts, presentations, and screen shares in real-time using peer-to-peer WebRTC technology.

How to Connect to a Stream

  • Obtain a secure session URL from the host broadcaster.
  • Paste the complete URL into your browser's address bar.
  • The system will automatically authenticate and establish a secure connection.

* Note: This page serves as the viewer client. Without a valid session ID, no video data will be transmitted or displayed.

WebRTC Architecture (Viewer)

LayerLink utilizes WebRTC (Web Real-Time Communication) to establish a secure, peer-to-peer (P2P) connection directly between the host and your browser. As a viewer, you receive the video feed directly from the host's device, ensuring minimum latency and maximum privacy.

Client Presentations

Join a high-quality screen share instantly without needing to download desktop applications, install plugins, or create user accounts.

Remote Code Reviews

Watch a developer navigate their IDE in real-time. P2P WebRTC technology ensures the text remains crisp and the frame rate stays high.

Secure IT Support

View sensitive server configurations or internal dashboards knowing that the video feed is end-to-end encrypted and never passes through a recording server.

UI State Simulation

Test the dynamic UI state handling of the LayerLink Viewer terminal without needing an active Firebase session URL.

Viewer FAQ

No. LayerLink relies entirely on native browser APIs. As long as you are using a modern browser (Chrome, Firefox, Safari, Edge), the WebRTC engine will process the incoming video stream directly in the DOM.

Absolutely not. This terminal acts strictly as a "Receiver." It only interprets the incoming video tracks provided by the host's WebRTC offer. It does not request microphone, camera, or screen recording permissions from your device.

If the host stops sharing their screen or closes their browser tab, the peer-to-peer connection state will immediately detect the failure, terminate the video feed, and display a "Signal Lost" error state.

What the viewer does, and why it works almost everywhere

The viewer's job is the easy half of a WebRTC session: receive a stream and render it. No screen capture permission is required, no local media is accessed, and nothing is transmitted from your device except the connection negotiation needed to establish the peer link. That is why viewing works on any reasonably modern browser — including phones and tablets, where hosting a share is not possible at all.

The asymmetry surprises people, so it is worth stating: mobile operating systems deliberately do not expose screen capture to web pages, which is a platform restriction rather than a limitation of this tool. Receiving video needs none of that machinery. Our sharer page covers the full connection sequence.

A session ID is a capability

Anyone holding the session identifier can attempt to join. Treat the link like a temporary password: send it over a channel you trust, and do not paste it into a public ticket, a shared document or a group chat that outlives the session.

Quality adapts to the link

WebRTC continuously adjusts resolution and frame rate to the bandwidth available. Text becoming briefly unreadable during a screen change is the codec reallocating bits after a keyframe, not a fault — it resolves within a second or two on a stable connection.

Why a session may not connect

If both peers are behind restrictive NAT or a corporate firewall, no direct path may exist and the connection cannot establish without a relay. A session that works between two home networks and fails from an office is almost always this, not a browser problem.

Troubleshooting a session that will not start

In rough order of likelihood:

  • The host has not started sharing yet. The viewer cannot render a stream that does not exist; the session must be live at the sharing end first.
  • The session has ended. Identifiers are not durable — once the host stops sharing or closes the tab, the peer connection is gone and the link is spent.
  • The identifier was altered in transit. Chat clients and email readers sometimes mangle or truncate long links. Copy it as plain text.
  • A network is blocking peer traffic. Corporate networks and some VPNs block the UDP traffic WebRTC prefers. Trying a different network is the fastest way to confirm.
  • A browser extension is interfering. Aggressive privacy extensions sometimes block WebRTC entirely, specifically to prevent local IP disclosure during candidate gathering.

That last point is a genuine trade-off worth understanding rather than working around blindly: ICE candidate gathering can reveal local network addresses to the other peer, which is inherent to establishing a direct connection. Extensions that block it are protecting that information at the cost of peer-to-peer connectivity.

What is and is not private here

The media stream is encrypted in transit with DTLS-SRTP, which WebRTC mandates rather than offers, and it travels directly between the two browsers rather than through a server. The signalling service sees only the metadata needed to introduce the peers — never video. This is the one part of Layerbit that involves a network service at all, which is why it carries a Cloud API badge while the rest of the toolkit does not.

Deployment Changelog

  • v1.3.0 Deployed enterprise documentation architecture and UI state simulation tools for the viewer terminal.
  • v1.2.0 Added automatic fallback handling for browser autoplay policies to ensure the stream plays even if muted by the client OS.
  • v1.1.0 Resolved a critical race condition by queueing remote ICE candidates until the local WebRTC description is fully established.
  • v1.0.0 Initial deployment of the LayerLink WebRTC Viewer Terminal utilizing Firebase signaling.