Have you ever thought about how easily you can make voice and video calls to your friends these days right from your browser? Well, welcome to the world of WebRTC, which acts like a secret sauce for powering seamless and powerful real-time communication from the web. 

WebRTC, being simple, enables face-to-face connections without the need for any middleman, eliminating the requirement to download apps, plugins, or software. This straightforward approach makes WebRTC a developer’s dream, as it’s easy to implement and use. 

Plus, it is easy, fast, and reliable. Additionally, we will see some of the other perks in the post below, so keep tuned. 

What Is WebRTC?

WebRTC, short for Web Real-Time Communication, is an open-source protocol that makes voice and video communication or any collaboration feature between browsers possible by using simple Javascript APIs. 

Remember that it does not make use of any plugins or software and works just fine. 

And hence, WebRTC is well-liked by developers as it lets them add attributes like file sharing, screen sharing, voice chats, and group video calls as easily as ABC. But how does the protocol do it? 

Well, by using media codecs, encryption, network protocols, and with the help of two channels:

1. Media Channels – which is responsible for handling the smooth transfer of audio and video calls between users. by making use of codecs like VP8, VP9, and H.264
2. Data Channels – which handles all non-media data like file sharing, screen sharing, real-time chats, and more by using the SCTP ( Stream Control Transmission Protocol)

The best part of these channels is that they can operate in the same peer-to-peer connection without establishing a new communication medium. So, mates, with WebRTC, you get endless possibilities. 

Be it building video chat API solutions, audio calling, video conferencing platforms, or customer support solutions, you can make any app with the lowest latency. 

How Does WebRTC Work?

Now that we have seen the definition of WebRTC, we will look at its functional part. The protocol establishes a single P2P connection, which means that two users (also called peers) are connected using a single WebRTC connection. 

Through this, they can communicate, share, talk, etc. Plus, I’ll give you a detailed breakdown of its working culture to help you understand the topic a bit faster:

1. Signaling

Before users or peers can start communicating with each other, they first need to send out some basic info, like their IP addresses and session details, via the signaling server. Now, as the signaling server is not part of the WebRTC, you can use any tech you want, like WebSocket or HTTP.

2. Peer Connection 

Once the details are exchanged, peers can establish a peer connection to send video, audio, chat messages, or any data directly without the need for any middleman servers.

3. Media Capture & Processing

This step uses codecs like VP8 or H.264 to share audio or video data from a microphone or webcam by converting it to digital signals and then transferring it over the internet.

4. Media Rendering

The WebRTC protocol decodes the media files received by the other peer and displays them on their screen.

5. NAT Traversal

In case of any network failure, peers can’t establish a connection, so WebRTC uses NAT traversal techniques like STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relay NAT) to bypass the network issues and foster smooth connection.

WebRTC Video Call Flow

Below is a step-by-step process on how the WebRTC protocol establishes video call capability between peers: 

Heads Up: I’ll represent peers as user A and user B.

1. The user A first creates the RTCPeerConnection method to establish the connection
2. Then user A also creates an offer, also called an SDP message, using the createoffer() function and adds a description
3. Then sends it to the user B using the signaling server
4. Likewise, user B establishes a connection using RTCPeerConnection and sets A’s offer as a remote description
5. User B responds to the SDP message using the createAnswer() function and sends it back to A using the signaling server
6. Now, the user A sets B’s answer as the remote description

Once this info has been exchanged, the video call capability can be established.

That’s it! The flow is simple and efficient, thus making WebRTC a reliable protocol to foster smoother real-time communication experiences from the web.

Is WebRTC Secure?

Undoubtedly, WebRTC is secure and designed to keep safety parameters in check. However, developers are advised to take extra precautionary measures to ensure every message is safe, as it is an open-source technology.

1. Encryption

Though WebRTC uses DTLS (Datagram Transport Layer Security) to encrypt video and audio files, you can use some of the secure methods like HTTPS for signaling and for stronger authentication.

2. User Privacy

WebRTC protocol establishes video calls or audio data from a camera or microphone after the user gives consent to use. Therefore, you must make sure that this permission is correctly requested and recorded

3. Deployment and Management

When you deploy WebRTC into your apps, you must use some secure practices like WebSockets, HTTPS, and CORS (Cross-Origin Resource Sharing) and perform regular audit checks to safeguard data

We can wrap up by saying that WebRTC is secure, and developers like you must implement the above measures strictly to protect user privacy at any cost.

How is WebRTC Used in Different Use Cases?

WebRTC is transforming communication between peers positively across industries or sectors. 

And we will see how the protocol is benefitting such use cases in the section below:

1. WebRTC For Business Communication 

The enterprises use the protocol to build apps that make video and audio calling easy and accessible, along with:

• Adding video conferencing capability to connect with remote employees

• Record video calls and share it with other participants
• Improving call quality through AI-powered features like noise cancellation
• Sharing screens during presentations to improve communication
• Enhancing the privacy of the app, and more

2. In Contact Centers

• Allows easily to switch between communication channels like chat and email
• Agents can directly send call links to customers, inviting them to join an audio or video call
• Improves customer service efficiency and minimizes costs

3. WebRTC For Authentication

• The protocol is used to verify user identity before sharing sensitive information
• Agents can also check the user ID through live video call

4. Customer Feedback

• Through video calls, agents/businesses can collect honest feedback from customers.
• It is also helpful for carrying out online interviews by eliminating in-person necessity.

5. WebRTC For Recruitment

• Fosters secure recruitment process through live video calls
• Saves time and cost, especially with employees working remotely

A Sample WebRTC Video Call App

We now understand that WebRTC-based video calling apps allow users to engage in video and audio calls without lags or plugins. And now, we will see an example of how such an app works:

1. 1. Setting up peer connection: Establish a connection between users with the help of APIs like RTCPeerConnection.
2. 2. Capture media: Code the getUserMedia() to allow browsers to capture audio and video calls from the user’s device.
3. Signaling: Use a signaling server to exchange user information like ICE candidates, SDP messages, and so on between peers.
4. Establish connection: Once all the above is checked, peers can start connection and exchange calls.
5. Display video: Use the videoelement() feature to display the video captured by the user on the other peer’s device.

Wrapping Up!

Alas! Here we are, at the end of the post, and by now, you may have been clear that WebRTC is revolutionizing real-time communication by allowing businesses to easily add audio and video calling features into web and mobile apps.

Plus, the protocol can be used in any case, and it will prove to be a game-changer. So, if you want to create a seamless user messaging experience, make WebRTC your coding block. Bye!!