A Quick Guide on WebRTC Video Calling API

The advancement of technology has made communication methods effortless and the major growth of that is real-time communication. It has brought the video and audio to the web by making it reach greater heights that we could never imagine. 

The global webRTC market size was valued at USD 2.3 billion in 2019 and is expected to grow at a compound annual growth rate (CAGR) of 43.4% from 2020 to 2027.

Real-time communication is the need of the hour ever since the pandemic devastated us across the globe. The demand for the Real-time communication platforms have increased tremendously as it offers high-speed networks and also mainly because of build a video chat app and voice chat application with WebRTC. 

WebRTC API: When it started and how does it work now?

Earlier in 2010, real-time communication systems were only available by additional software, plugins or by using Adobe flash. Later in 2013, a cross-browser video call was introduced where you can connect between Google Chrome and Firefox.

In 2014, a new trend developed as the first cross-border data transfer led to real-time communication through the client-side.

After all these developments, it is now known as the WebRTC peer-to-peer that we use in our everyday routine in all the available browsers such as Chrome, Firefox, Safari, iOS, Android, etc.

WebRTC video API is an HTML5 JavaScript specification that originates with a bunch of web APIs and communication protocols to launch an uninterrupted video/audio enabled communication between browsers. Video call API WebRTC is more popular as the APIs are plugin free, involves no tiresome downloads and installations.

Businesses use the WebRTC-enabled chat application for instant messaging, video conferencing, click-to-calls and peer-to-peer streaming. WebRTC live video chat app development offers the developers with a set of exclusive features and innovations making it a broadly accepted technology among the developers’ community.

Architecture and Functionality of WebRTC Video Chat App

Building a WebRTC video and voice chat application enables users with mutual communication through browsers to capture, transmit and encode real-time streaming between two parties that consists of 3 HTML5 APIs. There the WebRTC video call app is stated as the peer-to-peer technology that helps the users to directly communicate with each other via browser.

WebRTC’s beauty is that it doesn’t need any additional software or hardware, the audio and video streaming can be done without the need of the intermediate web services. Real-time communication and the URL based meeting rooms are the prime examples for the level of comfort provided by video call WebRTC API integration.

The simplest video call API WebRTC can achieve everything with a connected webcam and browser whereas some require an IP camera, encoder and streaming solution. It can be played on any HTML5 player dissimilar to a Flash-based video player that involves additional plugins. 

Nevertheless, WebRTC cannot handle large audiences; it was primarily designed for native information exchange without an intermediate server. Anyone looking to stream with a WebRTC video chat app development will need the help of a streaming server or service.

Why WebRTC API for Video & Voice Call?

1.Removes Noise: 

The ambient noise from an audio file can be removed by a WebRTC voice call; it is the same case for video calls too. 

2.Compress files:

WebRTC can compress and decompress the audio or video by treating it with codec. 

3.Transmits via Firewalls:

WebRTC transmits to create interactive Connectivity Establishment (ICE) by routing from one peer to another via firewalls. 

4.Data Encryption:

WebRTC aids the user to manage the bandwidth while securing the user data with end-to-end encryption before transmitting the connections. 

5.Peer-to-peer Communication:

WebRTC is an open source project that constantly evolves and improves the peer-to-peer communication by simplifying it.

6.Low bandwidth and Low latency:

Embedded with audio-video communication, WebRTC helps the user to consume very less bandwidth and offers zero latency that is supported with all the major browsers and mobile devices.

Limitations in WebRTC and how to overcome it?

Scalability

As the WebRTC API integration was not designed with the attention to scalability, it requires the user’s participating browser to connect with one another through a peer connection.

To solve this, the scalability can be increased with a real-time streaming server if the WebRTC is transcoded into HLS for unlimited distribution. As a result, it will lead the video conferencing API integration to large scale broadcasting.

Broadcast Quality

To enable the real-time delivery, WebRTC sacrifices the bidirectional frames from the GOP (Group of Picture) structure which results in a negative impact on the quality.

It can be resolved by reducing the number of connections between each client after all the participants connect to a common streaming server. It will thus lead to a streaming of larger scale and thereby optimizing the quality as well.

What CONTUS MirrorFly offers you with WebRTC Enabled Video Call Application?

Be it any industry like Healthcare, e-learning, integrating a real-time communication solution is the only way to benefit the business. Our solutions can be customized as per your needs as we offer added extensions with Video recording, security tools, video call api to enhance your business.

CONTUS Mirrorfly’s WebRTC streams can be transcoded into an HTTP-based protocol like MPEG-Dash or HLS that can broadcast up to 1000+ users for large scale distributions. CONTUS Mirrorfly offers you browser-based broadcasting and end-to-end broadcasting to any destination without the use of an encoder. 

Kick-start your business with a self-hosted streaming platform and get a distinguished publishing space to encash your exceeding revenue. Create high-quality video calling app using WebRTC video APIs to kindle media viewership at peak now!

Frequently Asked Questions (FAQ)