WebRTC

Beyond audio/video, WebRTC's RTCDataChannel API enables direct, bidirectional transfer of arbitrary data. This is used for:

• File sharing and P2P transfers within a browser.
• Multiplayer gaming with low-latency state synchronization.
• Collaborative editing tools where cursor positions or text changes are broadcast instantly.
• IoT device control for real-time command and sensor data streams.

WebRTC's getDisplayMedia API allows users to securely share their entire screen, a specific application window, or a browser tab. This is critical for:

• Remote support and troubleshooting (e.g., customer service portals).
• Online presentations and webinars.
• Collaborative code reviews or design sessions. The stream is captured locally and transmitted via a secure peer connection to the viewer.

In decentralized networks, WebRTC provides the signaling and transport layer for direct peer-to-peer communication between nodes or users. Use cases include:

• Decentralized video conferencing (e.g., Huddle01).
• P2P data marketplaces and mesh networks.
• Decentralized VPNs and privacy tools that route traffic through peer networks. It enables dApps to have real-time features without relying on centralized WebSocket servers.

WebRTC facilitates direct communication between web dashboards and IoT devices (like cameras, drones, or sensors). It allows for:

• Live video feeds from security cameras or drones viewed in a browser.
• Real-time telemetry data streaming with sub-second latency.
• Bidirectional control, sending commands to devices while receiving sensor data over the same persistent connection established by the ICE framework.

WebRTC establishes peer-to-peer (P2P) connections directly between users' browsers or applications without a central server for data relaying. It provides:

• Signaling: A process to exchange connection metadata (using STUN/TURN servers for NAT traversal).
• Secure Data Channels: Encrypted, bidirectional communication for arbitrary data.
• Media Streams: For real-time audio and video, though less common in pure Web3 contexts. This direct P2P architecture is inherently aligned with decentralization principles.

WebRTC acts as the transport layer for decentralized applications (dApps) requiring real-time interaction. It enables features like:

• P2P Messaging & Chat: Direct user-to-user communication in social dApps or marketplaces.
• Multiplayer Gaming: Low-latency state synchronization for on-chain games.
• Decentralized Video/Audio: For meetings or streams in DAO governance tools. By removing centralized relay servers, it enhances privacy, reduces costs, and eliminates single points of failure for communication.

WebRTC is often implemented as a transport protocol within the libp2p networking stack, which is used by protocols like IPFS and Ethereum. Key integrations include:

• libp2p WebRTC Transport: Allows browser-based nodes to directly connect to other libp2p peers.
• Browser Node Participation: Enables browsers to act as full participants in decentralized networks, not just clients.
• NAT Traversal: Leverages STUN/ICE to connect peers behind firewalls, a common challenge for P2P networking.

WebRTC can facilitate secure, off-chain signaling for blockchain transactions. Use cases include:

• Wallet-to-Wallet Communication: Directly proposing transactions or sharing session keys between wallets.
• State Channel & Layer 2 Setup: Exchanging initial parameters for opening payment channels (e.g., in Lightning Network implementations).
• Decentralized Exchanges (DEXs): Enabling direct order book communication or negotiation between trading peers before settling on-chain.

In Decentralized Physical Infrastructure Networks (DePIN), WebRTC enables direct device coordination. Examples:

• Sensor Networks: Direct data streaming from IoT devices to processing nodes.
• Edge Computing: Coordinating compute tasks between devices in a mesh network.
• Bandwidth Sharing: Projects that create decentralized CDNs or VPNs use WebRTC for efficient P2P data routing between user devices.

While powerful, WebRTC in Web3 faces specific hurdles:

• NAT/Firewall Traversal: Requires STUN/TURN servers, which can become centralization points if not decentralized.
• Peer Discovery: WebRTC handles connection, not discovery. This relies on separate decentralized systems (e.g., DHTs, blockchain).
• Scalability: Maintaining many simultaneous P2P connections in a browser is resource-intensive.
• Security: The signaling channel must be secured, often via the underlying dApp or blockchain identity system.

While WebRTC media streams are encrypted using DTLS-SRTP, the signaling process (establishing the connection) is not defined by the standard and is often handled over unsecured WebSockets or HTTP. An attacker intercepting signaling messages could perform a session hijack or redirect the media stream. Secure implementation requires using TLS (HTTPS/WSS) for all signaling traffic and validating certificates.

WebRTC can be abused as a DoS amplification vector. An attacker can trick a victim's browser into sending high-bandwidth media streams to a target server, multiplying the attack traffic. This exploits the browser's ability to generate data without user consent once a connection is established. Defenses include rate limiting, origin validation on TURN servers, and requiring user gesture approval for media capture before connecting.

The peer-to-peer data channels (RTCDataChannel) can be used to create covert communication channels for malware to exfiltrate data from a compromised network, bypassing traditional firewall rules that allow WebSocket traffic. The data is encrypted end-to-end, making inspection difficult. Network monitoring solutions must be aware of WebRTC traffic patterns and consider deep packet inspection or behavioral analysis for detection.

WebRTC's powerful APIs require a secure origin (HTTPS or localhost). Browsers enforce a permissions model for accessing the camera and microphone, but malicious sites can still attempt to trick users. Developers must implement clear user consent flows and use the Permissions API to check status. Best practices include:

• Using the getUserMedia() promise to handle denials.
• Stopping tracks when they are no longer needed.
• Providing clear UI indicators when media is active.

TURN servers are used for relaying traffic when direct peer-to-peer connections fail (due to NATs/firewalls). They see all media content, making them a high-value target. Critical security measures include:

• Using TLS or DTLS for TURN connections.
• Implementing short-lived, ephemeral TURN credentials.
• Regular auditing and isolation of TURN servers from other infrastructure.
• Monitoring for unusual bandwidth or connection patterns.