Have you ever wondered about having voice and video streaming without requiring any software or plug-ins?

There was a time when Flash and plug-ins were the only methods of real-time communication.

WebRTC (Web Real-Time Communication) technology delivers instant voice and video streaming without any plug-ins and Flash player.

Today, we at VideoEncrpyt will discuss the basics of Web real-time communication, APIs, Signaling, and Gateway.

What Is WebRTC?

WebRTC is a free, open framework for the web that enables the real-time communication of audio, video, and data in Web and native apps. Web Real-Time Communication leverages peer-to-peer connections between browsers for the exchange of data without using any third-party software or plug-ins. It is a combination of standards, protocols, and JavaScript APIs, to support interactive live streaming between individuals and browsers to communicate through a set of standard protocols.

What Is WebRTC API

There are three primary components of WebRTC. Here we will discuss the three components.

Media Stream

The Media Stream API controls the multimedia stream data and also controls the devices that are producing the media and identify the information of the devices that are able to store and stream the media. Thus, it allows access to devices like- cameras and microphones using JavaScript.

Peer Connection

Peer Connection is the core of the WebRTC as it controls the Codec implementations, SDP negotiation, Bandwidth management, NAT Traversal, Packet loss, and Media transfer. It allows users to create connections with their participants without using any intermediate server. The participants are required to plug the media fetched from the media stream into the peer connection to create an audio or video feed.

Data Channel

Data Channel uses UDP-based streamings with the configuration of the SCTP (Stream Control Transmission Protocol) protocol. It helps in transferring of data and media data directly between the participants. Thus, data channels allow better delivery with reduced congestion in the network.

What is signaling?

WebRTC uses Peer Connection to communicate by using streaming data between browsers at two endpoints i.e. sender and receiver to exchange data to coordinate communication over a call. This mechanism to coordinate communicating and sending messages is known as signaling.

Understanding STUN

STUN (Session Traversal Utilities for NAT) allows WebRTC users to find out their own public IP address by making a request to a STUN server so that the user can connect through the possible direct route.

Understanding TURN

TURN (Traversal Using Relays around NAT) acts as a relay server when the communication fails. It helps the users in determining the routers on their local network when a direct connection is not possible due to firewall restrictions.

WebRTC Gateways

WebRTC provides a simple way to implement peer-to-peer communications between browsers and focus primarily on the media plane. WebRTC gateway helps you in connecting the conventional IP communications networks with the open ecosystem of the Internet.

It connects the Web and telecom world through signaling, networking, media, transcoding, and various functions. It can be delivered in the form of a turn-key appliance or a software-based solution that runs on the standard server.

Conclusion

WebRTC combines real-time interactivity and streaming right in the browser and is going to be a new front in the long war for delivering real-time communication without using any Flash player and plug-ins. We at VideoEncrypt are planning to deliver you an interactive and streaming platform based on WebRTC.