WebRTC

What is WebRTC: How It Works and Its Key Applications

6 min read

February 26, 2024

In recent years, real-time communication has become a key component in enabling collaboration over the Internet. Whether remote learning, virtual team meetings, virtual events or customer support, most business operations today rely on real-time communications and data transfer.

WebRTC, or Web Real-Time Communication, plays a crucial role in enabling real-time communication in direct and P2P video, audio, and chat communication, as well as enhanced privacy, live streaming, and redefining conventional broadcasting.

In this article, we will explore WebRTC, how it works, and where it’s implemented.

Table of Contents

What is WebRTC?
How does WebRTC work?
What is WebRTC used for?
What are the pros and cons of WebRTC?
Embracing the future of real-time communication with WebRTC

What is WebRTC?

Web Real-Time Communication (WebRTC) is a revolutionary technology that empowers real-time communication directly in web browsers. It’s an open-source project initiated by Google and has become a fundamental part of modern web applications. WebRTC enables users to engage in direct video, audio and data exchange without needing additional plugins or complex installations.

At the heart of WebRTC's functionality is its peer-to-peer communication capability. Unlike traditional methods that rely on server-based data routing, WebRTC allows for direct transfer of data, audio and video streams between users’ browsers. This peer-to-peer approach significantly reduces latency, enhances the transmission quality, and ensures a more secure exchange of information.

WebRTC integrates various JavaScript APIs to facilitate this seamless interaction. Notably, it utilises components like getUserMedia, which grants access to a user’s camera and microphone; RTCPeerConnection, for handling the complexities of the direct connections; and RTCDataChannel, for enabling arbitrary data transfer.

The importance of WebRTC in today's digital landscape cannot be understated. It has transformed how we communicate online, making real-time, interactive web applications more accessible and efficient.

How does WebRTC work?

WebRTC employs three significant components to enable real-time P2P links between two or more browsers for video and audio calls as well as sharing data.

RTCPeerConnection

RTCPeerConnection is a core component of the WebRTC API, utilised for establishing peer-to-peer communication between browsers or devices. It allows direct audio, video and data exchange, bypassing the need for intermediary servers.

This interface manages the connection, including creating and maintaining it. The process starts with creating a new RTCPeerConnection object, which is configured with ICE servers for IP address discovery and NAT traversal.

Using WebRTC through RTCPeerConnection

To establish a connection using RTCPeerConnection, peers follow a specific protocol.

Initially, one peer generates an offer using the createOffer() method and sets it as the local description. This offer includes supported media formats, codecs and potential connection pathways (ICE candidates).

The offer is then sent to the other peer using a signalling server, which merely forwards it without processing. Upon receiving the offer, the receiving peer sets it as its remote description, creates an answer with createAnswer(), and sends it back. Both parties exchange ICE candidates through the signalling server, facilitating connection establishment.

Signalling

Signalling in WebRTC is the process used to coordinate communication and manage peer sessions. It involves the exchange of information, like session descriptions and ICE candidates.

As WebRTC doesn't define a specific signalling protocol, it's up to the developers to implement one, which can be based on WebSocket, HTTP or other protocols. Signalling is crucial for initiating, controlling and terminating the peer-to-peer communication process. It's also essential to exchange media metadata, such as resolution and codecs, and other control messages between peers to establish a direct connection.

What is WebRTC Signalling?

Read the blog

What is WebRTC used for?

Peer-to-peer video and audio communication

WebRTC facilitates direct P2P video and audio communication, which is crucial for WebRTC-based applications like video conferencing. By eliminating the need for third-party plugins or software, it offers a seamless and efficient user experience.

Its ability to compress and decompress media files for transfer also enhances the quality and speed of these communications, making it ideal for real-time interactions across various platforms.

Data transfer

WebRTC is not just limited to audio and video streaming; it also supports the transfer of arbitrary data directly between peers. This capability is essential for applications that need real-time, bidirectional communication - like file sharing, chat applications, and collaborative tools.

Peer-to-peer screen sharing

WebRTC enables peer-to-peer screen sharing, allowing users to share their screens or specific application windows during video calls or conferences.

This feature is handy in educational, business and technical support scenarios, where remote collaboration, visual aids and real-time guidance are crucial.

Privacy

Privacy is the foundation of WebRTC's design. It incorporates end-to-end encryption for all data streams, safeguarding sensitive information during transmission. This encryption is crucial in scenarios like telehealth consultations, private meetings and secure data exchanges where sensitive information is transferred.

Live streaming and broadcasting

WebRTC is also valuable in live streaming and broadcasting. Its low-latency streaming capabilities are suitable for real-time video content delivery, such as live events, gaming and interactive media.

WebRTC adapts to network conditions, ensuring high-quality streaming even in fluctuating bandwidth scenarios. Its widespread browser compatibility and ease of integration make it a popular choice for developers creating live-streaming platforms.

Exploring WebRTC Market Trends & Predictions for 2024

Read the blog

What are the advantages and disadvantages of WebRTC?

Pros of WebRTC

Open source

WebRTC, being open-source, provides a platform for community-driven development and innovation, allowing for continuous improvement and customisation.

Support for all modern web browsers

WebRTC is widely supported across major browsers like Google Chrome, Mozilla Firefox and Safari, ensuring accessibility and compatibility.

Versatile

WebRTC is also adaptable for various applications, from simple video chats to complex enterprise solutions, making it a versatile choice for developers. Some prominent applications using WebRTC include Google Meet, Facebook Messenger, live and VR chat, Instagram Live, and Digital Samba Embedded, to mention a few.

Secure and stable

With built-in end-to-end encryption and stable connectivity, WebRTC ensures a secure communication channel crucial for sensitive data exchange.

Better video and audio quality

It offers superior video and audio streaming quality thanks to efficient codecs and real-time adjustments to network conditions.

Cons of WebRTC

Complex signalling process

The absence of a standardised signalling protocol in WebRTC can complicate the process of establishing connections.

Inadequate retransmission strategy

In unstable network conditions, WebRTC might face challenges in effectively retransmitting lost packets, which can affect the quality of communication.

Inflexible bitrate adjustment

Its approach to adjusting bitrates might need to be more flexible and efficient in varying network scenarios, potentially affecting stream quality.

Embracing the future of real-time communication with WebRTC

WebRTC has revolutionised modern web communications, offering a flexible and secure platform for real-time audio, video, and data sharing in web browsers.

Digital Samba leverages WebRTC's capabilities to provide GDPR-compliant, end-to-end encrypted group video conferencing and seamless integration into various software, mobile apps or websites. Our API and SDK support clear, lag-free conferencing that is adaptable to diverse user needs, ensuring high-quality and secure virtual interactions.

If you are looking to incorporate WebRTC into your projects, Digital Samba offers an efficient and reliable solution. Explore the potential of WebRTC in your own projects with Digital Samba and experience seamless real-time communication.

FAQ

WebRTC stands for Web Real-Time Communication. It is an open-source technology that enables direct communication between browsers and mobile applications, allowing audio, video, and data sharing in real-time. WebRTC is often used for applications such as video calling, online conferencing, and peer-to-peer file sharing.

WebRTC works by establishing peer-to-peer connections between users' devices, enabling them to share audio, video, and data streams. Through a series of APIs, WebRTC handles the complex tasks of network negotiation and media handling, making it easy to set up real-time communication directly within a browser.

WebRTC is used for a variety of real-time communication applications, including video conferencing, voice calling, live streaming, and file sharing, which are done directly in the browser. Its capabilities make it ideal for social media platforms, e-learning tools, telehealth, and any platform that requires fast, reliable communication without needing plugins.

Unlike traditional communication methods, WebRTC does not require external plugins or installations. It provides an API-based framework that enables real-time video, audio, and data sharing directly within compatible WebRTC browsers, enhancing user experience and reducing latency in communication.

WebRTC technology is widely used in video applications like video conferencing and telemedicine, as well as in IoT (Internet of Things) devices, online gaming, customer support chat, and collaborative tools. Its versatility makes it a popular choice across various industries for applications that require real-time interaction.

In terms of browser compatibility, WebRTC is supported by most major browsers, such as Chrome, Firefox, Safari, and Edge, making it accessible to a wide range of users and platforms.

The main benefits of WebRTC include low latency, high-quality media streaming, enhanced privacy, and the elimination of third-party plugins. WebRTC's seamless integration into browsers improves user experience and reduces the complexity of setting up real-time communication.

Janus is a WebRTC server that helps manage and distribute WebRTC media streams, often used as a gateway for applications requiring real-time communication. It enhances WebRTC capabilities by handling tasks like scaling, recording, and managing media flows across devices.

Request a free consultation

Experience the power of Digital Samba's WebRTC video conferencing solution

Get a consultation

← Why Janus is Digital Samba's Favorite SFU for WebRTC Applications

The Role of Scalable Video Coding (SVC) in Video Calls →