The Most Comprehensive Guide on WebRTC

WebRTC or Web Real-Time Communications, though taken a relatively new web technology, has web-based communication at an entirely new level with the promise of heralding into a brave new world of communication on the horizon. The free, open-source WebRTC project makes use of a set of JavaScript APIs to facilitate peer-to-peer communication between web browsers and different devices. The question remains what makes it so popular.

A big draw with WebRTC is it eliminates the use of plugins or third-party software to facilitate real-time communication, helping achieve the ultimate goal of moving in a plugin-free world.

That’s not an ordinary feat anyway; it has revolutionized the face of web-based communication to an extent beyond imagination. Nowdaysfor making a video conferencing call or sharing a file—all that you need is a URL. No need to toggle between different interfaces. Thus, WebRTC has opened doors of infinite possibilities for developers and businesses in the form of a wide range of use cases across sectors and industries like HR & Recruitment, Banking, insurance, Healthcare, and Online Education.

How Does WebRTC Work?

Before digging deeper, having a brief understanding of the functioning of WebRTC will help you understand it better in the right context. It requires streaming audio, video, and messaging to interchange random data between browsers and applications. Under the hood, numerous linked APIs, protocols, and signaling-related technologies work in sync within WebRTC to achieve their purpose. These communications APIs protocolsallow the interchange of various media data streams.

To enable WebRTC communication, the following three steps are required:

  • Through GetUserMedia JavaScript API, a webcam or microphone can access various media streams.
  • The basic network-related information, Such as ports and IP addresses, needs to be shared with another browser through signaling (which is carried out by RTCPeerConnection JavaScript API).
  • Next, the basic information about media data is carried out by RTCPeerConnection JavaScript API.

What appears simple is, in reality, a complex process.

WebRTC: Some Interesting Use Cases

Within a short span, WebRTC stack has become the backbone of modern communication and several huge applications like Google Hangouts and Facebook Messenger, thanks to the ease of development, built-in security, flexibility, and scalability it offers. Here are some interesting use cases it offers:


Real-time videos can help retailers connect with prospective buyers and showcase products in a more organized way.

Virtual Consultations

Using in-app live video can get the medical answers quickly whenever you need them from the comfort and privacy of your home.

Virtual Recruitment

The remote recruiting environment should be jitter-free and more tech-driven than traditional recruitment.


Telehealth can be used to expand the doctors’ access where in-person appointments are not possible for specific reasons.


Motor insurance claim processes are often long and tedious but video-enabled motor insurance claim process can improve claimant’s satisfaction and adjusters’ productivity.

Video KYC

Video KYC for banks and financial institutions can be conducted online without having to visit the branch or have someone come for KYC.

WebRTC Adoption: 6 Key Considerations That You Need to Make

Without a doubt, WebRTC has opened a vast window of opportunities. Making a final choice, like any other technology, requires some careful considerations, such as whether you should opt for MCU or SFU topologyserver-side or client-side recording. As you continue reading this article, you will learn how to make an informed choice.

Here are the following six critical considerations before taking the very first step toward WebRTC adoption:

Should I Go For SFU or MCU?

WebRTC, by itself, can only provide peer-to-peer communication on the browser. To enable multiparty calls, an intermediate server is required to receive and send media data. This purpose can be achieved by two topologies: MCU (Multipoint Conferencing Unit) and SFU (Selective Forwarding Unit). Making the right choice between MCU and SFU topologies will be crucial to ensure the quality of communication delivery.

Server-side or Client-side Recording

The recording is not an inherent part of the WebRTC framework. Enabling this feature deserves careful thought like whether you should go for server-side or client-side recording capability.

It’s just VoIP, Can I Connect to PSTN?

Though VoIP can be very effective in managed networking setups, considering its better bandwidth management capability, elastic scalability and call quality, it can be problematic in unmanaged IT networks due to local issues, such as work from home and bad weather. Therefore, users must dial into an active VoIP-based WebRTC session from a PSTN when invited to join.

Know the Risks Before Diving Into WebRTC

WebRTC is powered by robust security architecture, it’s not a foolproof system like any other technology.

We need to understand that it’s not a standalone system; The WebRTC ecosystem comprises many client-side, hosts, servers, application & transportation layers and cyber security threats may emanate from anywhere. However, WebRTC is inherently safer in many ways.

Which of the Video Codecs To Use?

Selecting the right video codec is crucial in the WebRTC project as it enables real-time streaming without plugin installation. Therefore, a careful analysis of WebRTC video codecs deserves thorough consideration.

Understand the Limitations of WebRTC Stack

It is a fact that RTC has not been designed for group calling, and it always had a peer-to-peer orientation right from the beginning. That’s why scalability has always been a challenge with the ‘vanilla’ WebRTC protocol. So, before joining the WebRTC bandwagon, understanding its limitations is crucial to stave off disappointments later.

DIY or Outsource to a CPaaS Service Provider?

By now, if you have made up your mind about adopting WebRTC for communication, you deserve a pat on the back. You have to also think about whether DIY or Outsource to a Vendor. Both options deserve thorough consideration. You should know the pros and cons associated with each option.

If you have decided to go for a WebRTC vendor, it is highly recommended to learn how to choose a WebRTC vendor. To know more, go through our detailed whitepaper: 8 Factors to Consider While Hiring a CPaaS Vendor.

WebRTC Solutions: Are They All the Same?

Now, if you have decided to outsource, the question may arise in your mind that when all CPaaS providers use the same WebRTC protocol, are they all the same? It is to be noted that all CPaaS solutions are not similar despite being powered with similar technology at the core. A host of technologies are used on top of the WebRTC protocol, which acts as a big differentiator.

Different CPaaS solutions may significantly vary in various aspects like bandwidth adaptability which depends on Simulcast, single or mmultiple RTCPeerConnection, apart from the impact of legacy users who would still be using PSTN/VoIP.

To Conclude

By now, you must have developed a good understanding of WebRTC technology. This open-source technology has opened unlimited possibilities by changing the way real-time communication takes place and consumes. We’re confident that the above WebRTC guide will pave the path forward for WebRTC adoption.


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