NAT Traversal Protocols

This section covers the core components that enable WebRTC (Web Real-Time Communication) to establish peer-to-peer connectivity, particularly across NAT/firewall environments.

Component

Description

Use Case

ICE (Interactive Connectivity Establishment)

Framework that tries multiple methods to connect peers. Combines STUN and TURN for NAT traversal.

Establishing peer-to-peer connections.

STUN (Session Traversal Utilities for NAT)

Helps clients discover their public IP and NAT type. Enables direct peer-to-peer communication.

Enables connections behind NAT/firewalls.

TURN (Traversal Using Relays around NAT)

Relays media when direct connection fails. Acts as a media proxy server.

Ensures reliable communication fallback.

WebRTC (Web Real-Time Communication)

A set of APIs and underlying protocols enabling peer-to-peer audio, video, and data in browsers. Enables real-time communication with NAT traversal support via ICE, STUN, and TURN.

Browser-based peer-to-peer media/data connections

RFC: RFC 8445

Main Features:

  • Connectivity framework for NAT/firewall traversal

  • Uses STUN and TURN as fallback mechanisms

  • Collects multiple candidates (IP addresses and ports)

Use Cases:

  • Peer-to-peer connection setup in WebRTC

  • Negotiating the best path between peers

Alternative Protocols:

  • None – ICE is the standard coordination mechanism for WebRTC

Standard: W3C Recommendation (latest March 13, 2025) Components: ICE (RFC 8445), STUN (RFC 5389), TURN (RFC 8656), and Media/Data APIs Status: Defines APIs (e.g., RTCPeerConnection) and underlying NAT traversal mechanisms :contentReference[oaicite:0]{index=0}

Main Features:

  • Browser-based framework for real-time peer-to-peer audio, video, and data

  • Integrates ICE, STUN, and TURN for NAT/firewall traversal :contentReference[oaicite:1]{index=1}

  • Exposes APIs (such as RTCPeerConnection) to manage connection setup and media handling

  • Supports media handling, secure transport (DTLS/SRTP), and data channels

Use Cases:

  • Video conferencing, voice calls, and real-time messaging in browsers

  • Peer-to-peer file sharing and data exchange via WebRTC Data Channels

  • Any app needing real-time, low-latency, NAT-traversing communication directly between clients

Alternative Protocols/Frameworks:

  • SIP/WebRTC Gateway – for interoperability with legacy VoIP

  • Traditional real-time transport using centralized servers (e.g. WebSockets, media servers)

  • Custom media/data transport over VPN or proprietary mechanisms