WebRTC (Web Real-Time Communications)

What is WebRTC (Web Real-Time Communications)?

WebRTC (Web Real-Time Communications) is an open source project that enables real-time voice, text and video communications capabilities between web browsers and devices. WebRTC provides software developers with application programming interfaces (APIs) written in JavaScript.

Developers use these APIs to create peer-to-peer (P2P) communications between internet web browsers and mobile applications without worrying about compatibility and support for audio-, video- or text-based content.

With WebRTC, data transfer occurs in real time without the need for custom interfaces, extra plugins or special software for browser integration. WebRTC enables real-time audio and video communication simply by opening a webpage.

How does WebRTC work?

WebRTC uses JavaScript, APIs and Hypertext Markup Language to embed communications technologies within web browsers. It is designed to make audio, video and data communication between browsers user-friendly and easy to implement. WebRTC works with most major web browsers.

WebRTC APIs perform several key functions, including accessing and recording video-, audio- and text-based data from devices to initiating, monitoring and ending P2P connections between devices via browsers and facilitating bidirectional data transfer over multiple data channels.

In most cases, WebRTC connects users by transferring real-time audio, video and data from device to device using P2P communications. In situations where users are on different Internet Protocol (IP) networks that have Network Address Translation (NAT) firewalls that prevent RTC, WebRTC can be used in conjunction with Session Traversal Utilities for NAT (STUN) servers. This enables a given IP address to be translated into a public internet address so peer connections can be established.

But there are also networks that are so restrictive that even a STUN server cannot be used to translate IP addresses. In these cases, WebRTC is used with a Traversal Using Relays around NAT (TURN) server, which relays traffic between users, enabling them to connect. The Interactive Connectivity Establishment protocol is used to find the best connection.

WebRTC example with a signaling server

Before audio and video files are sent, they must be compressed due to their large size. Also, media that is received over a peer connection must be decompressed. WebRTC uses a codec process to do this.

What is WebRTC used for?

The goal of WebRTC is to facilitate real-time P2P communications over the internet. There are several use cases for WebRTC, including the following:

• WebRTC is used for video chats and meetings on video calling platforms, such as Zoom, Microsoft Teams, Slack or Google Meet.
• Industries, including healthcare, surveillance and monitoring, and internet of things, use WebRTC. For example, WebRTC use in Telehealth enables doctors to conduct virtual office visits with a patient over a web browser.
• In the field of home and business security and surveillance, WebRTC is used as a connecting agent between browsers and security cameras.
• WebRTC is heavily used for real-time media.
• WebRTC provides the underlying connection between instructors and students for online education.

What are the pros and cons of WebRTC?

WebRTC presents opportunities and challenges to organizations.

The advantages of WebRTC include the following:

• eliminates much of the in-house manual integration work required of IT;
• can adjust communication quality, bandwidth and traffic flow whenever network conditions change;
• is supported by most major web browsers, including Google Chrome for desktop and Android, Mozilla Firefox for desktop and Android, and Safari;
• works on any operating system as long as the browser supports WebRTC;
• does not require third-party components or plugins; and
• is free as open source software.

Disadvantages of WebRTC include the following:

• Each user must establish a P2P browser connection, making bandwidth an issue.
• Maintenance costs can be high because WebRTC requires powerful servers.
• Security and privacy standards are still unclear, leaving it up to IT departments to ensure that corporate security and privacy standards can be met.
• There are no definitive quality of service standards, which means that quality of video or audio over the internet may be inconsistent.

Is WebRTC secure?

Every WebRTC software component is encrypted, and every WebRTC API requires secure origins via Hypertext Transfer Protocol Secure (HTTPS) or localhost. Nevertheless, there are still open security questions that developers using WebRTC must consider. Signaling processing methods, or the methods used to exchange metadata, are not specified for WebRTC signaling. This means that developers must decide which security protocols to use and ensure that the protocols they select can be maintained with WebRTC.

WebRTC Protocol: What is it and how does it work?

WebRTC is a highly flexible streaming protocol suitable for nearly every audio/video use case. While this characteristic and its media encryption capabilities make it a popular choice among broadcasters, this protocol does have three major drawbacks potential streamers should be aware of. Read on to learn what they are, how much they might impact your stream, and learn if WebRTC is right for your project.

What Is WebRTC?

Web Real-Time Communications (WebRTC) is an open-source communications protocol that enables real-time voice, text, and video streaming between web browsers and devices. With the help of signaling servers, WebRTC is able to manage multiple device connections and ensure their integrity.

WebRTC provides software developers with application programming interfaces (APIs) written in JavaScript. Developers can leverage APIs to create a peer-to-peer conferencing (P2P) connection between internet web browsers and mobile applications without worrying about compatibility and support for multimedia content.

Data transfers occur in real-time without needing custom interfaces, extra plugins, or additional software — WebRTC enables real-time audio and video streaming simply by opening a webpage. However, to access more advanced features like screen sharing, you can either install a plug-in or build your own.  

How Does WebRTC Work?

WebRTC embeds real-time communications technologies within web browsers using JavaScript, APIs, and Hypertext Markup Language. It makes audio, data, and video streaming between most major web browsers user-friendly and easy to implement. Before audio and video files are sent, they must be compressed due to their large size. Also, media received over a peer-to-peer (P2P) connection must be decompressed. WebRTC uses a codec process to do this.

WebRTC typically connects users by transferring real-time audio, video streams, and data from device to device using P2P. But, if users are on different Internet Protocol (IP) networks with Network Address Translation (NAT) firewalls that prevent real-time communication (RTC), Session Traversal Utilities for NAT (STUN) servers can be used to translate an IP address into a public internet address, so P2P connections can be established. WebRTC APIs initiate and monitor P2P connections between devices via browsers and facilitate bidirectional data transfer over multiple channels.

Why Choose WebRTC? 

WebRTC has a wide range of benefits for both the broadcaster and the receiving audience. Here are the top three: 

1. WebRTC eliminates much of the in-house manual integration work required of IT teams by adjusting communication quality, bandwidth, and traffic flow whenever network conditions change.
2. WebRTC is supported by most major web browsers, including Google Chrome for desktop and Android, Mozilla Firefox for desktop and Android, and Safari, and works on any operating system.
3. WebRTC does not require third-party components or plugins and is free open-source software.

When To Use WebRTC

Use WebRTC to facilitate real-time communications over the internet for a number of use cases: 

• Businesses can host video chats and meetings on video calling platforms, such as Zoom, Microsoft Teams, Slack, or Google Meet.
• Telehealth organizations can connect patients and providers securely over a web browser for virtual appointments.
• Home and business surveillance companies can use WebRTC as a connecting agent between browsers and security cameras.
• Sporting, news, and other live events can use WebRTC to deliver a high-quality stream in real time.
• Students and teachers can facilitate remote learning over WebRTC.

When Not To Use WebRTC

While WebRTC is encrypted and its APIs require secure origins via Hypertext Transfer Protocol Secure (HTTPS), there are still some concerns around security questions developers must consider. There are no signaling processing methods specified for WebRTC signaling. If you do not wish to decide which security protocols to use and are unsure if the protocols you select can be maintained with WebRTC, you might consider looking into an alternative video streaming protocol.

While security is a top concern among many potential WebRTC users, here are three other aspects of the protocol worth considering where they fall on your list of development priorities:

1. Bandwidth: Each user must establish a P2P browser connection, making bandwidth an issue.
2. Cost: Maintenance costs can be high because WebRTC requires powerful servers.
3. Service Standards: There is no definitive quality of service standards, which means that the quality of video or audio over the internet may be inconsistent.