The Ultimate Guide to WebRTC (Web Real-Time Communication) in 2021

WebRTC

Real-time video streaming has become more important than ever. The rise in demand for this technology has coincided with the massive shift towards video conferencing among businesses, organizations, and individuals.

Since the initial COVID-related lockdowns, many day-to-day activities and special events have become virtual. While live streaming with relatively low latency has worked for the larger scale events, the smaller events that involve interaction with or participation from the audience have relied on peer-to-peer streaming that has real-time or ultra-low latency.

Web Real-Time Communication (WebRTC) has made peer-to-peer streaming possible.

In this post, we’re going to cover everything you need to know about WebRTC. We will discuss the history and technical background of WebRTC before we take a specific look at how this project has supported the rise of peer-to-peer streaming. Additionally, we’ll review some use-cases and benefits for streaming with WebRTC.

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 Video Conferencing
Peer-to-peer streaming proved very valuable during COVID-related lockdowns.

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 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 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 uses 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?

peer-to-peer streaming
WebRTC is used for peer-to-peer streaming.

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 with 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

WebRTC Streaming
WebRTC is slowly making its way into professional video hosting.

Currently, Dacast is using WebRTC streaming to a limited degree through an integration with a very popular video conferencing application, Zoom. It has been reported that Zoom uses WebRTC to transmit video but uses another technology to transmit audio.

The Dacast Zoom integration allows Dacat broadcasters to stream video-conferenced meetings to their audience. This taps into the real-time latency of Zoom while also taking advantage of the professional streaming features of our platform.

The Dacast developers hope to make WebRTC streaming available on the platform down the road so that our broadcasters can experience ultra-low or real-time latency and the other benefits for WebRTC. 

Right now, Dacast uses primarily RMTP ingest and HLS delivery to an HTML5 video player. In order to make WebRTC streaming possible with this setup,

Final Thoughts

Looking for a highly capable online video platform with video conferencing integrations? Dacast is the solution for you. Try our platform risk-free for 30-days with no binding contracts or credit card 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.

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