- Last Updated: 09 Jun 22
- 12 min read
What is WebRTC and how does it work?
- Last Updated: 09 Jun 22
- 12 min read
For real-time communication (RTC) to work between browsers, a number of technologies are required that are well integrated: audio and video processing, application, and networking APIs, and additional networks protocols that enable real-time streaming.
This results in WebRTC – over a dozen different application protocol standards and browser APIs that provide real-time communication over the internet.
What is WebRTC?
Real-time communication over the Web is called Web-RTC. Real-time communication direct from the browser is enabled by this transformative piece of technology.
In the absence of Web RTC, devices need to be connected through an intermediary server. It means that both devices need to have the same plugin-in or app installed for communication to work. The one sends the communication to the server, and the server sends it to the other device.
An example of this can be found in Skype – it is a great video communication tool but only works when you have Skype on every device involved.
In an environment where customer service is involved, this can be annoying. The WebRTC is here to save the day! With Web-RTC, devices communicate directly with each other, eliminating the server.
Devices can communicate directly without downloading anything or installing any plugins via the web browser, which automatically embeds the capability.
The Web RTC protocol supports audio, video, and data transmission over the internet and can be used to make web calls, video chats, online messaging, and screen sharing within web pages and mobile applications.
History of WebRTC
RTC used to be complex, requiring expensive licenses or custom development of audio and video technologies. Web-based RTC integration has been slow and difficult owing to poor integration of existing content, data, and services.
In 2008, Google introduced Gmail video chat to the market. A Google+ feature that began in 2011, Hangouts became a standalone product in 2013. RTC technologies were highly favored by Google. The company acquired GIPS, a developer of many RTC components, including codecs and echo cancellation techniques.
In order to ensure industry consensus, Google made the tools developed by GIPS open-source and engaged with relevant organizations such as the Internet Engineering Task Force (IETF) and World Wide Web Consortium (W3C).
Web-RTC was first implemented by Ericsson in May 2011. It was a turning point in the growth of the communication industry.
It was so evident that the goal of Web-RTC was building a way to communicate in real-time, without the need for plugins.
For many web services, like Skype, Facebook, and Hangouts, RTC was required. However, you had to download apps, install plugins, or install downloading services in order to access the RTC. Updates, downloads, and installations of plugins require considerable effort.
In most cases, errors occur, and the plugins do not work as they should. The use of plugins may require you to pay for certain licenses or technologies. Additionally, it is really hard to get people to download a plugin.
These principles led to Web-RTC being developed as an open-source, free, standardized, and browser-integrated standard.
Basic Concept of Web-RTC
Web-RTC is a technology that enables you to communicate directly from the web browser in real-time (via voice, video, and other data). This can be done without the use of any plugins or downloads.
Somehow, that’s not saying much.
Now let’s talk about what makes WebRTC unique in browsers.
If it was up to now you envisioned a web application as client and server, you would think of your browser as the client. Whenever it wants something from the server, it connects to it. Let’s call these requests. In turn, the server replies with its response.
While we’ve grown beyond that with Web Sockets, it is still somewhat similar. The message that I want to send to someone who is currently using his own browser needs to go to the server and then to them. In much the same way as the post office.
Web-RTC is the point of divergence between browsers and HTML.
In the meantime, both browsers need to communicate in some way to locate each other, but once they have done that signaling, they can communicate directly with each other – without ever touching the webserver. It’s like magic.
That’s why Web-RTC is sometimes called peer-to-peer technology. In short, P2P. This is due to the fact that browsers can communicate directly.
How does WebRTC work?
In general, WebRTC connections go as follows:
Your device > STUN server > Peer-to-peer communication channel > Recipient’s device
However, each step has its own process. The process is as follows:
You need to establish a connection between all the devices being used on the call by your Web RTC app before you can make an audio or video call.
NAT and a firewall must be traversed before Web-RTC can establish a connection. By creating a public access IP address for your computer, a firewall or NAT device hides your private IP address and broadcasts it to the world.
Only your computer knows your personal IP address. To retrieve your public-facing IP address, the Web-RTC app contacts the STUN server (Session Traversal Utilities for NAT). The Web-RTC application can then route incoming connections to the correct IP address.
The Web-RTC app retrieves your public-facing IP address from the STUN server, after which it retrieves the public-facing IP addresses for all the other devices that will be participating in the call.
A list of possible connection configurations is designed once all IP addresses have been known to the app-also known as ICE candidates (Interactive Connectivity Establishment)-and the most efficient configuration is chosen.
All the devices participating in the WebRTC call can then exchange audio and video data in real-time using the private data channel generated by the Web RTC app. In addition, only devices on the call know the configuration, so it’s a private, protected connection that anyone else can’t access.
All of the devices on Web-RTC communicate directly with one another using this connection method, through which the audio and video streams are transmitted directly to one another. Unfortunately, establishing a direct connection is not always possible.
Web-RTC uses TURN (Traversal Using Relays Around NAT) in these circumstances. Turn servers are merely repeaters.
The app moves audio and video data from the computer to a TURN server, which transmits the data to the receiving device and vice versa if there isn’t a direct connection between the device and the TURN server during a Web-RTC call.
Using a TURN server for communication with Web-RTC, however, is the last resort. A TURN server must have tremendous processing power and bandwidth in addition to being super reliable. In addition to being costly, this is also complicated.
Direct connections for WebRTC are therefore the most efficient option whenever possible.
This is a very generalized description of Web RTC. Several technologies and APIs are used in this process.
However, the supporting technologies provide a means for devices on a Web-RTC call to communicate between themselves and navigate the complex system of ports, protocols, and networks.
A set of APIs also allows cameras and microphones to be accessed and audio and visual data to be collected. Using peer-to-peer connections to achieve real-time communication is a great way to communicate, but it’s not perfect.
Your data will remain secure with WebRTC as it employs a number of security measures. Among them are:
1. Browser Protection
With WebRTC, we don’t need to install plugins between browsers since the transmission is done directly between them. Therefore, Web-RTC is inherently safer because it offers additional protection against malware and unwanted software installation concealed as plugins.
In addition, Web-RTC is an integral part of browsers, so it is typically updated automatically by vendors as any potential vulnerabilities are addressed.
2. Media Access
The Web-RTC specification addresses possible concerns regarding the condition of accessing media resources by requiring explicit permission to use the camera or microphone.
The Web RTC protocol prohibits applications from accessing devices without consent. The users’ UIs and their hardware will also display information when the device is active.
The SSL encryption standard is a mandatory component of Web RTC and is enforced at all connection establishment and maintenance stages. Securely exchanging key data is best achieved using DTLS (Datagram Transport Layer Security) handshakes with perfect forward secrecy (PFS) ciphers.
AES (Advanced Encryption Standard) keys can then be generated from key data, and SRTP (Secure Real-time Transport Protocol) will encrypt and decrypt the media as necessary. As a result, these acronym-laden technologies translate into extremely secure connections that are difficult to circumvent with current technology.
Reasons to Choose WebRTC
- WebRTC is open source.
- It is completely free to use for both commercial and private purposes. Those are sufficient in themselves. The fact that it is constantly evolving means you can rely on it for many years to come.
- It is widely supported.
- Almost every browser supports Web-RTC now. Therefore, it is used in almost every possible application scenario. Web-RTC is also available for mobile applications.
- Several mobile applications use it currently. It is possible to use SDKs on both mobile and embedded platforms, so there are no limitations!
- Don’t think it is only for voice or video chat.
- In addition to using Web RTC for peer-to-peer communication, you can create group conferencing solutions, create video conferences, and record meetings using it.
Benefits of WebRTC
Web-RTC provides developers with a simple channel for achieving high-quality real-time communication. In addition, Web-RTC is easier for end-users, which makes for a more pleasant experience for them.
1. Better Sound Quality
With Web-RTC, you can cancel echo and reduce noise, and your microphone sensitivity can be adjusted automatically. The clarity of Web-RTC calls alone makes it stand out from other technologies. Even on slow connections, WebRTC delivers great call quality through the use of modern audio and video codecs.
2. Open source technology
The WebRTC community quickly evaluates and checks open-source code. Additionally, the open-source code leads to widespread adoption.
3. Outstanding compatibility
Since WebRTC is an open-source protocol, all of the major operating systems support it. Additionally, WebRTC is compatible with HTML5 and Web Sockets. Therefore, Web-RTC is compatible with most browsers.
4. Requires no plugins
When using real-time communication technologies, a browser plugin is needed to make calls. The end-user must install these plugins, resulting in a worse end-user experience. Most browsers support Web-RTC without requiring any plugins.
Drawbacks of WebRTC
1. Quality of Service
It is impossible to guarantee the quality of service on the internet because it is in the public domain (no one owns it). It can be problematic for a business, school, or other entity relying on a communication system on a continuous basis.
2. Integration Inhibitors
Organizations must realize that it is not a magic bullet that will solve all their problems. Many companies will face integration challenges and adoption barriers.
3. Conference calling is resource-intensive
Multiple video and audio streams cannot be synchronized by every web browser. The audio and video streams need to be mixed by a conferencing server before they are distributed as a unified data stream during a conference call.
For Web-RTC group calls, most companies use IaaS cloud solutions in place of building conferencing servers on-premise.
What does Web-RTC mean for contact centers and customer service?
Communication barriers are reduced with Web-RTC. When it comes to communicating with customers, Web-RTC provides a vast array of possibilities.
Web-RTC is supported by the majority of major web browsers, including Chrome, Edge, Firefox, and Safari, which means that contact centers can offer customers more and more contact channels without requiring them to download anything. These include:
The Web-RTC protocol makes it possible for customers to call companies from their websites. The benefit for the customer is that they don’t need to leave their current web page to reach out, and the benefit for the contact center is that it provides more information on the customer journey. Furthermore, it is the ideal way to offer callbacks.
Prior to Web-RTC, an agent and a customer would have to have an application like Skype installed in order to video chat. Web-RTC enables contact centers to have live video chats with their customers directly through their website without downloading or installing any software.
WebRTC allows for live chat, which is a form of online messaging. It facilitates real-time communication through a website. An agent can immediately respond to a message sent by a customer. Due to Web-RTC’s ability to embed in both mobile apps and browsers, contact centers can use mobile devices to provide live chat for customers.
Get Started with Web-RTC
WebRTC is an optimal choice if you need to build real-time communication functionality. Neither the developers nor the end-users find it complicated, and none of the other protocols do what Web-RTC does so well.
Because Web-RTC just works, build your solution on a network that just works, as well as on a VoIP solution that’s easy and intuitive enough for Web-RTC users.
KrispCall provides an excellent package of VoIP features and modern communication technologies built in a way to maximize the web RTC’s abilities to provide a truly transformative experience for small and large businesses.
At an affordable price, KrispCall offers a ton of amazing features that work great for your business communications. And there is a Schedule Free KrispCall demo now, to cherish about. You can get started with its services without spending a penny.
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