Understanding WebRTC: Real-Time Communication Made Simple

What is WebRTC?

WebRTC (Web Real-Time Communication) is a Google open source software library and development framework. It was officially announced by Google in May 2011. WebRTC library exports methods to developers. WebRTC integration makes it easy to build a peer to peer audio, video and screen sharing functionality.

WebRTC library was first integrated by Google Chrome and Mozilla FireFox and made the access of WebRTC library method through JavaScript to the developers. While development started in 2011, WebRTC became a fully standardized and stable technology in 2021.

Why Was the WebRTC Needed?

In the modern era of VoIP communication (from 2005 onward), VoIP became a crucial component of the software and telecommunications industry. Various VoIP protocols were introduced to support internet-based calling and conferencing, enabling a shift away from traditional phone systems.

However, there was no standard library available for developers to easily implement VoIP functionality, especially on the web. Web developers lacked a unified solution to access media devices such as microphones and cameras and to handle real-time audio and video communication.

To address this gap, WebRTC (Web Real-Time Communication) was introduced. It provided a standardized open-source JavaScript API that allowed peer-to-peer communication directly in the browser without requiring any plugins or external software. This transformed web-based communication by making it easier, faster, and more secure to build real-time audio and video applications.

How Did Web Browsers Work Before WebRTC (for Real-Time Communication)?

Before the introduction of WebRTC, real-time communication in web browsers was achieved using downloadable client-side plug-ins. These plug-ins were typically delivered as ActiveX controls, .cab (cabinet) files, or .msi installers and were supported by browsers like Internet Explorer.

When a webpage included HTML tags referencing such plug-ins, the browser would prompt the user to download and install them. If the user accepted, the browser would install the plug-in, which then became accessible to client-side JavaScript. These plug-ins were often code-signed to assure users of their authenticity and to allow privileged access to system resources.

Developers used this client-side downloadable plug-in technology to implement complex features such as VoIP protocol handling, audio and video capture, microphone and camera access, and UDP communication. Since these capabilities were not natively supported by browsers, JavaScript running in the browser would interact with the installed plug-in to control real-time audio and video communication.

Downloadable client-side plug-in were eventually rejected by modern browsers due to serious security risks. They had deep access to the operating system and could be exploited by malicious actors to install malware or spy on users.

What is VoIP protocol handshaking?

VoIP protocols consist of two main parts: signaling and media.

The signaling part is responsible for the handshaking process. This involves sending and receiving protocol messages (such as SIP packets) to establish, manage, or terminate a call. During this exchange, both parties agree on how the call will be set up, including codecs, ports, and media types.

Once the handshaking is complete, the media part begins. This is where the actual voice or video data is transmitted using a media transport protocol like RTP (Real-time Transport Protocol).

Most VoIP signaling packets contain SDP (Session Description Protocol) data, which provides the details needed for media negotiation such as IP addresses, ports, codecs, and session parameters.

For example, SIP (Session Initiation Protocol) is a widely used signaling protocol for establishing, managing, and terminating real-time communication sessions such as voice, video, and messaging over IP networks.

Once the session is established through SIP, the media streaming begins using RTP.

In this process, SIP messages carry SDP data between peers. Each peer receives the other's SDP information, and once both sides have the necessary details, they begin exchanging RTP packets to transmit the actual media (audio or video).

How Does WebRTC Work?

WebRTC (Web Real-Time Communication) is a browser-based library that enables real-time audio, video, and data communication between peers without requiring plugins or external software.

When a developer uses WebRTC, it provides methods to access media devices such as the microphone and camera. Once the required audio or video streams are captured, WebRTC generates an SDP (Session Description Protocol) offer. This SDP packet contains all the necessary information such as supported audio and video codecs, IP addresses, ports, and other session details.

This SDP offer must be sent to the remote peer using a separate signaling mechanism, since WebRTC itself does not define a signaling protocol. Common signaling methods include WebSocket, HTTP, or other custom APIs.

The remote peer receives the SDP offer, processes it, and responds with its own SDP answer, completing the handshaking process. Once both peers have exchanged SDP information and negotiated the connection, WebRTC establishes a peer-to-peer connection and begins media transmission using RTP.

WebRTC also uses ICE (Interactive Connectivity Establishment), STUN, and TURN servers to handle NAT traversal and ensure connectivity between peers even if they are behind firewalls or on different networks.

What is the Difference Between SIP and WebRTC?

WebRTC is a library or framework used to develop peer-to-peer audio and video communication. It captures audio and video devices, generates SDP packets, and uses WebSocket or HTTP as signaling protocols to deliver the SDP packet to the remote peer.

Whereas SIP (Session Initiation Protocol) is a signaling protocol specifically designed to carry SDP packets and deliver them to the other peer for session negotiation.

What is the Difference Between RTP and WebRTC?

WebRTC is a development library and technology, not a protocol. It enables real-time peer-to-peer audio, video, and data communication. WebRTC generates SDP (Session Description Protocol) packets, which are exchanged between peers using signaling protocols such as WebSocket or HTTP.

After both peers exchange SDP packets and agree on media parameters, they start sending RTP or secure RTP (SRTP) packets to each other for actual media transmission.