Understanding WebRTC Architecture: The Core Components

Introduction

Now that we have a basic understanding of what WebRTC is and why it was developed, let's dive deeper into its architecture. To enable seamless real-time peer-to-peer communication, WebRTC leverages a structured architecture composed of multiple interdependent components. In this blog, we'll explore the key building blocks of WebRTC.

WebRTC Architecture

WebRTC follows a layered architecture that enables real-time peer-to-peer communication. The architecture consists of multiple components working together to handle media capture, processing, transmission, and connectivity. Below is a detailed breakdown of the core architectural components. WebRTC can be broadly divided into three main layers:

• Application Layer - Interfaces like getUserMedia, RTCPeerConnection, and RTCDataChannel that developers use to integrate WebRTC into applications.

• Transport Layer - Protocols such as SRTP (Secure Real-time Transport Protocol) for media encryption and SCTP (Stream Control Transmission Protocol) for data transfer.

• Network Layer - The ICE framework along with STUN/TURN servers that assist in NAT traversal and connectivity establishment.

The WebRTC architecture consists of multiple components that work together to establish and maintain real-time communication. These include:

1. Web API Layer

The Web API layer provides interfaces that developers use to integrate WebRTC into applications. This includes:

• MediaStream (getUserMedia API): Responsible for capturing audio and video from the user's device. It allows access to the microphone and camera, providing the media streams that will be transmitted over the WebRTC connection.

• RTCPeerConnection API: The heart of WebRTC, responsible for:

  • Establishing a peer-to-peer connection between two devices.

  • Handling audio, video, and data streams.

  • Managing network traversal through ICE (Interactive Connectivity Establishment).

  • Encrypting and securing the media exchange.

• RTCDataChannel API: Enables peer-to-peer data exchange outside of media streams.

2. WebRTC Core (C++ API)

At the core of WebRTC lies the WebRTC C++ API, which provides the fundamental mechanisms for peer-to-peer communication. It includes:

Session Management and Abstract Signaling

• Handles connection establishment, media negotiation, and ICE candidate gathering.

• Uses SDP (Session Description Protocol) to negotiate media capabilities.

Voice Engine:

• iSAC / iLBC Codec: For high-quality audio transmission.

• NetEQ for voice: Manages jitter buffering to smooth out audio playback.

• Echo Canceler / Noise Reduction: Enhances audio clarity.

Video Engine:

• VP8 Codec: Used for video compression and transmission.

• Video Jitter Buffer: Ensures smooth video playback.

• Image Enhancements: Improves video quality dynamically.

Transport Mechanism:

• SRTP (Secure Real-time Transport Protocol): Encrypts media streams.

• Multiplexing: Combines multiple data streams efficiently.

• P2P (Peer-to-Peer Communication): Enables direct communication between devices.

• STUN + TURN + ICE: Assists with NAT traversal and connectivity establishment.

3. Network I/O and Connectivity

WebRTC relies on a robust Network I/O layer to ensure seamless peer-to-peer connectivity. This includes:

• ICE (Interactive Connectivity Establishment): A framework that helps determine the best network path for communication.

• STUN (Session Traversal Utilities for NAT) Servers: Help peers discover their public IP addresses to establish a direct connection.

• TURN (Traversal Using Relays around NAT) Servers: Used as a relay mechanism when direct peer-to-peer communication is not possible.

Basic Workflow of These Components

The WebRTC architecture follows a sequence of steps to establish communication:

1. Capture Media: The user’s device captures audio/video using getUserMedia.

2. Signaling Exchange: Peers generate and exchange SDP (Session Description Protocol) offers and answers through a signaling server to negotiate media capabilities and connection parameters.

3. ICE Candidate Discovery: STUN/TURN servers assist in finding the best path for connectivity.

4. Peer Connection Establishment: RTCPeerConnection sets up a secure and optimized connection.

5. Media/Data Transmission: Audio, video, and data streams are transmitted directly between peers.

Conclusion

WebRTC’s architecture is designed to enable efficient real-time communication through a layered approach. The Application Layer provides developer-friendly APIs, the Transport Layer ensures secure and efficient media transmission, and the Network Layer handles connectivity challenges. Each of its core components plays a crucial role in ensuring smooth media and data exchange. In the next blog, we’ll break down the workflow of these components in detail.