The Ultimate Guide to WebRTC (Web Real-Time Communication) in 2024

Real-time video streaming has become essential in today’s digital-first world, where the demand for interactive and seamless online experiences is higher than ever. Businesses, educational institutions, entertainment providers, and individuals increasingly rely on live video to connect and engage with audiences in real-time.

With more events, training sessions, and customer interactions taking place virtually, peer-to-peer streaming with ultra-low latency has become crucial for fostering engagement and providing immediacy in communication. Large-scale live streaming suits events with larger audiences, but smaller, interactive experiences benefit from real-time technology like Web Real-Time Communication (WebRTC), which powers direct, high-quality connections with minimal delay.

In this post, we’ll explore everything you need to know about WebRTC. We’ll discuss its history, technical foundation, and how it has become a cornerstone for real-time peer-to-peer streaming. We’ll also look at specific use cases and the benefits of using WebRTC for streaming in today’s fast-paced, interconnected landscape.

Table of Contents

• The Rise of Peer-to-Peer Video Conferencing
• What is WebRTC?
  • The Technical Background of WebRTC
  • Support of WebRTC
• How Does WebRTC Work?
• What is WebRTC Used for?
• Benefits of Streaming with WebRTC
• WebRTC Streaming on Dacast
• Final Thoughts

The Rise of Peer-to-Peer Video Conferencing

Peer-to-peer communication refers to any instantaneous digital communication. Text messages, phone calls, and social media chats all fall into this category. Peer-to-peer video conferencing is when two people chat on camera from remote locations.

A decade ago, Skype and Facetime were some of the first video chatting options available to consumers. Between then and now, more of our favorite streaming apps have helped us connect with friends, family, and associates around the world. Facebook, Snapchat, Whatsapp, and other platforms have given users the ability to make video calls right in the app.

When the world shut down due to the spread of COVID-19 and in-person interactions were no longer possible, peer-to-peer conferencing kept the world afloat. Important meetings and events were forced to move online. People needed face-to-face contact for different reasons, and video conferencing made that happen. Meetings, classes, and even doctor appointments were done on video.

Peer-to-peer video conferencing is a little bit different from live streaming in the sense that live streams are typically single-sided and the viewer on the other side of the screen can’t talk back. 

Since live streams are typically broadcast to hundreds, thousands, or even millions of viewers, the technology that they rely on to deliver their content is a little bit different and has some latency. Large live streams are typically transported with a combination of RTMP and HTTP live streaming (HLS). However, peer-to-peer video streaming uses WebRTC.

What is WebRTC?

Web Real-Time Communication (WebRTC) is a streaming project that was created by Google. This open-source project was designed to support Google’s acquisition of Global IP Solutions, a video conferencing and VoIP technology company, in 2010. The WebRTC project was set into motion the next year.

Over the next few years, the project was tested with several other web conferencing projects. In 2014, WebRTC was implemented in Google Hangouts in a limited capacity. The developers had many triumphs and failures. They received plenty of feedback that helped them perfect the technology.

The first stable release of the WebRTC project was in May 2018, and in January 2021, WebRTC received a W3C recommendation.

The Technical Background of WebRTC

WebRTC is an open-source project that supports real-time video conferencing over both applications and browsers. This project is brought to life by several different standards and protocols.

The technology behind WebRTC is built upon the foundation that was laid with early VoIP technology. If you are not familiar, VoIP stands for “Voice Over Internet Protocol.” Essentially, this refers to phone calls powered by the internet.

Since this project was not built entirely from scratch, it has led to rapid development.

Support of WebRTC

WebRTC is a widely supported project. It is supported by most major browsers, including Safari, Google Chrome, Microsoft Edge, Mozilla Firefox, and more. 

The ultra-compatibility of this protocol makes it easy to integrate with just about any site or program for playback on any device or browser.

How Does WebRTC Work?

WebRTC is responsible for two major aspects of peer-to-peer conferencing. First, it is responsible for media capture on your device. That means that WebRTC is the technology that tells your device to start recording. Second, it is responsible for transmitting the data between the two devices.

The basis of WebRTC is a series of JavaScript APIs. The three main APIs include “getUserMedia,” “RTCPeerConnection,” and “RTCDataChannel.”

“getUserMedia” helps users capture audio and video content by making the connection with the camera and microphone on the user’s device. “RTCPeerConnection” facilitates the transmission of audio and video between peers’ devices. This API also handles the security of the call and manages the amount of bandwidth that is being used. “RTCDataChannel” allows devices to send arbitrary data between one another.

WebRTC can be incorporated into different sites and programs API. This structure eliminates the need for additional programs or plug-ins to tap into the real-time conferencing technology. This alone makes it very valuable to developers.

It is important to point out that WebRTC does not detect signals from other devices that want to initiate a web conference. It simply facilitates the conferences once the connection is made.

What is WebRTC Used for?

WebRTC is primarily used for peer-to-peer communication, specifically with web conferencing. WebRTC powers programs that facilitate both video and audio calls across the internet. This could be used for anything as simple as a video chat with a friend or as important as a conference call with your enterprise’s executive team.

WebRTC is slowly making its way into online video streaming. It is possible that streams that are currently transported by the RTMP and HLS protocols could be delivered by WebRTC in the future. This would allow online video platforms to offer streams with no latency.

Streaming with real-time latency would give a competitive edge to broadcasters who are covering events that are also being covered by other networks. This would allow them to deliver the event to their audience as fast as technologically possible. 

WebRTC is also very valuable for virtual events that involve real-time participation from the audience. Streaming with ultra-low or real-time latency allows them to be more engaged and partake to create a more lifelike experience.

Programs Using WebRTC

There are several major programs that you’ve likely used in the past that are powered by WebRTC. Some of these include:

• Google Meet 
• Google Hangout
• Slack
• Whatsapp
• Discord
• Facebook Messenger
• Gotomeeting
• Snapchat
• Houseparty

This goes to show how important this technology is in different areas of life. A lot of professional and personal communication is powered by this innovative project.

Benefits of Streaming with WebRTC

The WebRTC project packs a lot of value for developers that are looking to incorporate peer-to-peer conferencing into their sites or programs.

Let’s take a look at what this project has to offer.

Ultra-Low/Real-Time Latency

The primary benefit of WebRTC is its ability to support low-latency streaming. In fact, WebRTC is capable of real-time streaming which means there is virtually no latency at all.

Open-Source

The open-source nature of WebRTC makes it very easy for developers to incorporate real-time web conferencing into their site or program. It is as simple as integrating a few lines of code.

It’s Free

WebRTC is absolutely free to use, which makes it very accessible. On the same token, developers can experiment with this project without making any financial commitment, which is definitely a win-win.

Ultra-Compatibility

This project is compatible with virtually every device or browser. This compatibility is more desirable than ever since people use peer-to-peer conferencing on a wide variety of devices.

It is very important to specify that this technology is 100% compatible with mobile devices. This is major since many people use their smartphones and tablets for video conferencing.

It’s Secure

In the beginning, there were some concerns with the security of WebRTC. However, now the project enables encryption on every audio and video exchange. This protects your web conferences from hackers tapping in and eavesdropping or capturing your conversation. 

Since WebRTC encrypts the data that is being exchanged, it is safe to use public wifi networks for calls.

High-Quality Voice and Video

WebRTC is capable of carrying out very high-quality web conferences. This means that as long as a user’s internet is fast, calls can be carried out with excellent audio and video quality.

It’s Adaptive

WebRTC is capable of something that is equivalent to adaptive bitrate streaming. The technology adapts based on the speed of the internet to successfully deliver the audio and video of a conference call.

Interoperability with Other Technology

Another benefit of WebRTC is the interoperability with other communication technology, including VoIP and video. This means that WebRTC can successfully communicate with programs that use other internet-based communication technology.

It’s Still Developing

Although WebRTC is a truly reliable peer-to-peer conferencing technology, it has not yet reached its final form. WebRTC will likely continue to develop to improve its current functionality and potentially become valuable for different types of streaming. 

WebRTC Streaming on Dacast

Dacast now offers WebRTC streaming directly through our platform, making it easier than ever before to start live streaming. All you have to do is log into your Dacast account, type in a name for the stream, and turn your webcam on. Using WebRTC, you can be streaming in a matter of seconds. This is a free feature available to all Dacast subscribers, and Dacast offers a free 14-day trial of the platform. Therefore, you could still be live-streaming a few minutes from now, for free, even if you don’t have a Dacast account set up yet. 

WebRTC offers real-time latency and virtually no set-up beforehand. Dacast’s WebRTC feature is perfect for any live stream where the viewers want to feel like they are present in the moment, such as corporate meetings, virtual education, gaming, religious services, or more casual live streams where the audience can interact with each other.

FAQ

1. What is WebRTC, and how does it differ from other streaming technologies?

WebRTC, or Web Real-Time Communication, is an open-source protocol designed by Google for real-time peer-to-peer audio, video, and data sharing over the web without plugins. Unlike RTMP or HLS, which are often used for large-scale streaming with higher latency, WebRTC is optimized for real-time communication, making it ideal for interactive video calls and conferences. It achieves ultra-low latency, which allows for seamless, live interactions between users.

2. How secure is WebRTC for video conferencing?

WebRTC uses several built-in security protocols, including DTLS (Datagram Transport Layer Security) and SRTP (Secure Real-Time Protocol), to ensure the privacy of audio, video, and data streams. It automatically encrypts all media exchanges, reducing the risk of unauthorized access during a call. Additionally, WebRTC’s peer-to-peer nature minimizes server reliance, further limiting exposure to potential breaches or hacking attempts.

3. What are some popular applications and platforms that use WebRTC?

WebRTC powers many widely-used applications, such as Google Meet, WhatsApp, Slack, Discord, and Facebook Messenger, enabling seamless video and audio communication. These platforms rely on WebRTC for its real-time, low-latency video conferencing capabilities and support across major browsers and mobile devices. WebRTC is also used in telehealth, online gaming, and customer support apps, reflecting its versatility across various industries.

4. How does WebRTC handle network issues, like low bandwidth or packet loss?

WebRTC uses adaptive bitrate technology, adjusting the video and audio quality in real-time based on network conditions to provide a smooth experience. If bandwidth decreases or packet loss occurs, WebRTC reduces quality temporarily to maintain the connection, resuming full quality as soon as conditions improve. This adaptability allows users to continue their calls without disruptions, even on unstable networks.

5. What are some real-life use cases for WebRTC in different industries?

WebRTC is used across diverse fields, such as telemedicine for secure virtual consultations, remote education for interactive online classrooms, and customer support for real-time video help desks. In gaming, WebRTC enhances multiplayer and collaborative game experiences, and in business, it supports virtual meetings and collaboration tools like Slack and Microsoft Teams. This versatility makes WebRTC a key technology in any industry needing real-time, interactive communication.

6. How can video streaming broadcasters leverage WebRTC to improve their video streams?

Video streaming broadcasters can leverage WebRTC to deliver real-time, low-latency streams, allowing audiences to experience events as they happen with minimal delay. By utilizing WebRTC’s adaptive bitrate streaming, broadcasters can dynamically adjust video quality to maintain smooth playback, even on networks with fluctuating speeds. WebRTC’s peer-to-peer connectivity reduces the need for centralized servers, lowering bandwidth costs and enhancing the scalability of interactive streams. Additionally, WebRTC’s built-in security protocols protect content from unauthorized access, ensuring a secure streaming experience. This technology is ideal for applications that require live interaction, such as virtual events, Q&A sessions, and real-time gaming, creating a more engaging viewer experience.  

Final Thoughts

Looking for a highly capable live streaming video platform with video conferencing integrations? Dacast is the solution for you. Try our live streaming platform risk-free for 14 days with no binding contracts or credit cards required. Get started by creating an account today.

If you have additional questions about WebRTC and other protocols for low-latency streaming, please feel free to contact us and our highly educated support team.

In the meantime, feel free to check out our Knowledgebase. A quick search for “latency” or “protocol” will generate dozens of results with tons of related information. For regular live streaming tips and exclusive offers, you can join the Dacast  LinkedIn group.