Complete guide to WebRTC and how it works

Web Real-Time Communication (WebRTC) has changed how we interact online, enabling real-time communication like video calls, voice chats, and data sharing through web browsers.

Unlike older solutions, WebRTC requires no plugins or additional software, making it a streamlined option for developers who need to embed communication features into websites and apps.

This blog post explains the fundamentals of WebRTC, how it operates, and the challenges and alternatives to this technology.

What is WebRTC?

WebRTC is an open-source technology developed by Ericsson and Google that enables real-time communication directly through web browsers and mobile applications. Built on JavaScript and C++ programming languages, WebRTC uses APIs that require no plugins, making it accessible across various platforms.

What does WebRTC do?

WebRTC lets users connect directly for real-time communication, including voice, video, and data transfers. Developers can embed these features into web pages and applications without additional software installations.

WebRTC has many use cases, such as:

• Video conferencing
• Voice over IP (VoIP)
• Online gaming
• Customer support platforms
• Telemedicine services

How does WebRTC work?

WebRTC establishes and maintains real-time connections through three main steps:

1. Initiating a call

When you start a WebRTC call, your app initiates a connection with other devices involved. It navigates through firewalls and Network Address Translation (NAT) devices to set up the connection.

2. Gathering public IP addresses

Your app contacts a STUN (Session Traversal Utilities for NAT) server to retrieve your public-facing IP address. It then gathers the public IP addresses of the other devices in the call.

The app generates a list of potential connection options, known as ICE (Interactive Connectivity Establishment) candidates, and chooses the most efficient one.

3. Establishing a data channel

Once the connection is established, WebRTC opens a private data channel to exchange audio, video, and other data in real time. During the call, WebRTC captures and transmits media streams using secure encryption protocols like DTLS and SRTP.

It also continuously manages the connection, adjusting for network changes to maintain quality, and uses a TURN (Traversal Using Relays Around NAT) server if direct peer-to-peer communication isn’t possible.

These processes ensure stable, secure, and uninterrupted communication.

WebRTC security

WebRTC encrypts all media and data transfers using SRTP. It also requires explicit user consent before accessing a device's camera or microphone for privacy control. Additionally, DTLS secures data channels, which results in encrypted and authenticated communication.

Challenges and limitations of WebRTC

Although WebRTC offers significant benefits, it does present a few challenges. One issue is browser compatibility. Not all browsers consistently support WebRTC, which can result in functionality differences.

Network quality plays a large role in the performance of WebRTC. Poor network conditions can result in latency or packet loss, which degrades the user experience.

Lastly, although WebRTC’s peer-to-peer model works well for one-on-one communication, scaling it for more extensive group sessions often requires additional infrastructure, such as media servers, to handle the increased demand.

Troubleshooting WebRTC challenges

The challenges outlined above can be addressed through ongoing improvements by browser developers and the WebRTC community.

For browser compatibility, updates and standardization across browsers can offer more consistent support.

Using adaptive bitrate streaming—which adjusts media quality based on network conditions—or faster networks like 5G can solve network quality issues.

To handle larger group communications, consider implementing Selective Forwarding Units (SFUs) or other media servers that can effectively manage increased traffic without sacrificing performance.

Get started with WebRTC

WebRTC is transforming real-time communication by enabling seamless, high-quality voice, video, and data sharing directly through browsers without the need for plugins or extra software. It empowers businesses and developers to build dynamic, interactive applications, from video conferencing platforms to collaborative work tools. As more industries embrace remote work and digital-first interactions, the demand for frictionless, real-time communication will only continue to grow.

For businesses looking to enhance customer interactions or optimize internal communication, integrating WebRTC into your stack is a strategic move. It reduces operational complexity and improves user experience by making communication more intuitive and accessible across devices and platforms.

Telnyx’s WebRTC solutions and our Quick Start Guide make it simple to integrate real-time communication features into your web and mobile applications. With our Voice SDKs for JavaScript, React, iOS, and Android, you can easily enable click-to-call, conferencing, and number masking. Our Voice API also supports features like push notifications, text-to-speech with Amazon Polly, call queues, and call recording, all backed by a reliable carrier-grade network.