WebRTC 101: Everything You Need To Know!

September 27, 2018

Saurabh Tripathi
Everything you need to know about WebRTC.

The fast-paced evolution of technology has made communication simpler! One important development is that of real-time communication.


What Is WebRTC?

WebRTC stands for Web Real-Time Communication. WebRTC is a profound technology that takes unified communications to a whole other level.

It is essentially an HTML5 JAVASCRIPT SPECIFICATION. It comes with a set of web APIs and communication protocols to establish a bidirectional, uninterrupted audio/video-enabled communication between web browsers.

These include data streams, STUN/TURN servers, signaling, JSEP, ICE, SIP, SDP, NAT, UDP/TCP, network sockets, and more. These APIs are plugin free, require no additional tedious installations and downloads.

Businesses, especially, can benefit a lot from WebRTC by using it for video-conferencing, click-to-call, peer-to-peer streaming and instant messaging.


What Makes WebRTC So Important?

WebRTC provides developers with a set of unique features and advancements making it a widely accepted technology among the developers’ community. Let us have a glance at WebRTC’s offerings.

  • It is an open source project: WebRTC is an open source platform. Thus, making it adaptable to build applications for anyone who wishes to use this technology.

  • It is constantly evolving and improving: With every releasing version, WebRTC thrives to be more user-centric, feature-packed and reliable. Each newly released version is implemented towards crisp user experience and also resolves queries registered in the prior versions.

  • Simplified peer to peer communication: This enables developers to use WebRTC to make web applications deliver a richer user experience by facilitating video chats, video calling, and peer-to-peer file sharing between the browsers.

  • Low bandwidth consumption and latency: The peer-to-peer communication mechanism results in low bandwidth consumption and reduced latency that users may face while sharing files.

  • Embedded audio-video communication: The wide set of functions delivered by WebRTC allow developers to embed audio/video communication into their web applications.

  • Secured data transmission: It secures the data through the source to destination data transfer. Encryption is used to secure the components while JavaScript apps are accessed through https, local hosts, etc.

  • Supported by widely used web browsers and mobile devices: This helps developers to embed audio or video tag through html5 embed tags. The reason being, WebRTC is supported by all the major browsers on the internet.


How To Create Voice And Video Calling Apps Using WebRTC?

When it comes to building a video chat application, WebRTC is the most sought-after solution. WebRTC voice/video chat app not only results in cost saving but also provides technical and architectural benefits with respect to a real-time scenario.

The major industrial use-case segments of WebRTC video/voice chat happen to be consumer web, mobile, social, broadcast, education, sales, and telecoms.

Let’s get down to developing your first WebRTC audio-video enabled communication application.


Technical specifications to consider for WebRTC applications:

  • Streaming of audio, video or other data.

  • Getting network information such as IP addresses and ports. Exchange this information with other WebRTC clients (peers) to establish a connection even through NATs and firewalls.


To achieve this functionality, WebRTC employs the following APIs:

  • MediaStream (commonly known as getUserMedia)
  • RTCPeerConnection
  • RTCDataChannel

1. MediaStream:

getUserMedia or MediaStream gets the access to the data streams through a web camera or a microphone. MediaStream is available in Chrome, Firefox, and Opera. getUserMedia() method takes three parameters:

  • A constraint’s object.
  • Success callback which, if called, passes a MediaStream.
  • Failure callback which, if called, passes an error object.

Each MediaStream has a label. An array of MediaStreamTracks is returned by the getAudioTracks() and getVideoTracks() methods.

getUserMedia is compatible with Chromium-based apps and extensions. audioCapture and/or videoCapture permissions can only be requested and granted once, during the installation.

Same goes with the pages using HTTPS: the permissions have to granted during the first request, for getUserMedia(). Very first time an Always Allow button is displayed in the info bar of the browser.

MediaStream has to be enabled for any streaming data source. This enables streaming from disc or from arbitrary data sources such as other inputs and sensors.

Note that getUserMedia() must be used on a server and not on the local file system, otherwise a PERMISSION_DENIED: 1 error will be thrown.


2. RTCPeerConnection:

Audio or video calling demands high-quality encryption and bandwidth management. It gets supported in Chrome (in desktop and Android both), Opera (on desktop and Android) and in Firefox too.

RTCPeerConnection is implemented by Chrome and Opera as WebkitRTCPeerConnection and by Firefox as mozRTCPeerConnection.

RTCPeerConnection is the WebRTC component responsible for handling stable and efficient communication of streaming data between peers.

RTCPeerConnection minimizes the myriad complexities. The codecs and protocols used by WebRTC are self-functional and performs many tasks by themselves so as to make real-time communication even over unreliable networks. These tasks include:

  • Packet loss concealment
  • Echo cancellation
  • Bandwidth adaptivity
  • Dynamic jitter buffering
  • Automatic gain control
  • Noise reduction and suppression
  • Image ‘cleaning’

What is signaling?

WebRTC uses RTCPeerConnection to communicate streaming data between browsers.

Signaling in real time communication is nothing but a mechanism to coordinate the communication and exchange of control messages between the sender and the receiver.

An important fact to note here is signaling is not part of the RTCPeerConnection API.

WebRTC app developers can choose whatever messaging protocol they prefer, such as SIP or XMPP, and also an appropriate duplex (two-way) communication channel.


WebRTC needs 4 server-side functionalities…

  • User discovery and communication.
  • Signaling.
  • NAT/firewall traversal.
  • Dynamic jitter buffering.
  • Relay servers (in case of a peer-peer communication failure)

3. RTCDataChannel:

Browsers with versions Chrome 25, Opera 18 and Firefox 22 and above support this peer-to-peer communication API.


WebRTC: What are your other options?

Creating a WebRTC powered Voice And Video Calling App is not an easy task and has its own challenges. There are multiple phases to this build and making sure that every element of your app works is rather difficult.

Coding every aspect of the app and ensuring that all API’s, modules, and servers are working effectively requires you to spend precious developer hours, maintaining this chat app is also an exhaustive and expensive affair.

Scalability is another issue with such an app where you have to ensure that your source app is able to update, upgrade and keep up with current market trends and new technologies.

In such a situation what are your other options? There’s a simpler solution that is cheaper, customizable, sustainable and scalable.


CometChat And WebRTC

CometChat is one such application that brings the world together effectively. It is powered by seamless video streaming, voice, text chat with a variety of innovative features.

It has contributed towards gathering the users of the apps/sites to communicate and collaborate.

WebRTC is a basic building block for CometChat’s audio video chat, conference calling features using its peer to peer technology. CometChat is based on WebRTC audio chat, WebRTC audio conference, WebRTC video streaming technology to bring its real-time communication to life.

This WebRTC enabled real-time video streaming, audio embedding has made CometChat a full fledge on the go user collaborator chat application with soaring advantages and benefits.

Some of the robust features provided by CometChat are:

  • Real-time text messaging: This feature allows users to send messages on the go!

  • One-on-one chat: If a user wants to communicate privately with another user, this feature comes in handy.

  • Group chat: For people who want to form special interest groups, or want to have real-time discussions, group chats are of great help! Users can decide if they want to keep these open to all, or make them ‘invitation-only’ groups.

  • Voice and video calling & conference: Users can use these features to chat face-to-face with those online.

  • File sharing: It allows users to send files on the go.

  • Collaborative whiteboard & Collaborative document: Collaborative whiteboard allows users to share ideas on a whiteboard on the fly! Collaborative document feature helps multiple users to work on a single document in real time.

  • Multi-language support & emojis: Users can change the UI for CometChat into a language of their preference, write in their local language, and use a vibrant mix of emojis to enhance their communication.


Why Choose CometChat Over Building Your Own WebRTC Voice & Video Calling App?

There are many libraries and SDKs available for iOS and Android to build real-time communication chat applications.

Many of them are based on WebRTC technology, CometChat being the most popular among the various chat applications available in the market.

The CometChat plugin is compatible with 92+ integrations including WordPress, which is popular with most businesses.

Here are some of the benefits of choosing CometChat over building your own WebRTC app!


1. No need to work from level 0!

As developers need to work from scratch to build a WebRTC voice/video calling chat app, it would require a significant amount of time. It has to go through all the stages of a software development lifecycle model.

CometChat is an extensively used collaborative chat application and has WebRTC as its powering tool for real-time audio and video calling.

When developers have a ready application it’s quite unnecessary to develop one from the level 0!


2. Facilitated integration

Developers need to integrate the CometChat SDK with their apps/websites. It does not take more than 20 minutes to get CometChat up and running.


3. Reduced development cost and time

Considering CometChat instead of developing a new application will not only save developers their time but also the cost budgeted for the project until it’s up and running.

Integration of CometChat would also lead the development of the project with the latest technologies being used in the application. If you choose to integrate CometChat, over developing your own chat application, it will surely reduce the project overheads and time.


4. Next generation features

CometChat provides with a wholesome of great features that ensures prolonged user presence on the apps.


Takeaway..

WebRTC is the reality of on-the-fly communication. With its promising features and ease of accessibility, there’s a good reason to get started with WebRTC as your voice/video streaming solution for users who will be communicating in real-time.

WebRTC with CometChat will allow you to save a lot of time, resources, and also give you the unique features you are looking for!

CometChat