WebRTC is defined as an industry-wide open-source project that provides real-time voice and video communications to web-browsers and mobile applications through application interfaces.
Endorsed by scores of applications, users, and programmers worldwide, WebRTC has become a powerful and reliable open-source tool capable of equipping web browsers with the ability to support real-time voice, video, and data sharing. Implementing the appropriate real-time protocols, WebRTC allows peers to send and receive media through web browsers, thereby assisting them to initiate click-to-start meetings without downloading dedicated apps or plugins.
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What is Web Real-Time Communication (WebRTC)?
WebRTC is an industry-wide open-source project that provides real-time voice and video communications to web-browsers and mobile applications through application interfaces.
Since it was first introduced in 2012 as an in-browser solution to real-time communications (RTC), WebTRC has been the subject of much speculation. Currently, and all the more due to the impact of the Covid-19 pandemic, it is considered to be a game-changer in the broader communications fabric. WebRTC application holds the potential to reduce communication and collaboration costs, drive improved revenue, create opportunities for innovation and competitive differentiation, and improve customer services – all areas critical to business success.
In simple terms, WEBRTC is a plugin-free communication technology that allows access to devices for real-time communication and media sharing. It makes use of Javascript, application programming interfaces (APIs), and HTML5 to embed audio/video streaming, and data sharing in the browser or application and needs no additional plugins for doing so.
As a result of the interoperability of the booming smartphone market today, employees now demand simple, reliable, and seamless experience while using their company’s conferencing systems. WebRTC addresses this turns this demand into a reality by standardizing media streaming across web browsers and devices, thereby enabling click-to-start meetings, and removing the hassle of extra software.
This means that users can directly start a voice/video call, or initiate screen sharing quickly and painlessly without actually leaving the browser or application. Today, WebRTC plays a fundamental role in empowering businesses to boost their sales by allowing real-time interactions among employees as well as customers and partners.
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WebRTC Design Structure
WEBRTC web applications are programmed as a mixture of HTML and JavaScript to allow communication between web browsers through standardized WebRTC APIs. It provides a range of utilities such as connection management (in a peer-to-peer manner), encoding/decoding capabilities negotiation, selection, media control, firewall, and more.
As a free and open-source project, companies can integrate various WebRTC APIs and customize them to meet their specific requirements. Over the years, WebRTC was able to spur a vibrant and dynamic ecosystem supported by a range of open-source projects and frameworks, along with numerous commercial offerings to improve real-time communications between browsers. WebRTC is a popular choice for real-time communications today, with integrations into numerous commercial products such as Google Hangouts, Whatsapp, Facebook Messenger, Zoom Team Communication, Skype et al, and more.
Developers can leverage WebRTC to facilitate peer-to-peer communication between two browsers without putting extra time and effort. Additionally, it lets web applications deliver richer user experience by facilitating video chats, video calling, and peer-to-peer file sharing. Beyond providing a superior user experience, WebRTC also plays a crucial role in creating new layers for embedding seamless visual and audio communications that require less battery, bandwidth, and network resources.
WebRTC’s unique position in the market stems from the fact that thousands of software developers are contributing to it in concert, thereby standardizing web conference protocols and making interoperability less of a challenge. In essence, WebRTC has leveled the technological playing field among conferencing providers to truly drive enterprise communication and collaboration, allowing multiple devices to smoothly share media and data among one another on a common platform.
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The three different types of WebRTC architecture are:
1. Peer-to-peer (P2P)
Peer-to-peer communication for WebRTC allows a direct exchange of media content between two browsers. Every public or private computer has a network access translation device (NAT) along with the firewall, which translates private IP addresses from inside a firewall to public-facing IP addresses for added security. This is where STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relays around NAT) servers come into play. A request for accessing the public-facing IP address of the computers is made to a STUN or TURN server. Once this is done, peers can immediately start real-time audio/video conferences of media sharing.
Implementation simplicity, low operating costs, and guaranteed end-to-end security between participants are the key advantages that this WebRTC architecture offers. However, it is crucial to remember that this architecture works best when the number of participants is less because multiparty call scenarios often require significant uplink bandwidths and elegant WebRTC leak tests to prevent IP leaks.
2. Multipoint Conferencing Units (MCUs)
Multipoint conferencing units (MCUs) is a frequently used WebRTC architecture that has supported a range of successful applications in legacy conferencing systems for years. This architecture assumes that the conference participants send each of their streams to the MCU, which decodes each received stream, rescales it, devises an entirely new stream, encodes, and sends it to all the participants.
Such an approach calls for very little intelligence in device endpoints, as the majority of the logic is located in the MCU itself. Besides, this WebRTC architecture can generate custom quality output streams for different participants depending on their specific downlink conditions. In short, MCU serves as a robust solution for driving interoperability in low capacity networks.
3. Selective Forwarding Units (SFUs)
One of the most popular choices for modern times, SFU architecture empowers participants to send and receive media streams or data via a centralized SFU server. During an audio/video conference or data sharing, the participants send multiple media streams to the server, and it is the SFU server that selects the media streams to forward among the other participants.
Unlike the MCU, there is no need to decode and re-encode received streams in SFU architecture as it merely serves as a forwarder of media streams. However, it requires more intelligent device endpoints and computing power than the MCU architecture. SFU architecture’s primary upshot is the ability to work with asymmetric bandwidth—both uplink and downlink, rendering it suitable for data communications utilizing asymmetric digital subscriber line (ADSL) networks. Another key advantage is the immense scalability that this architecture offers, users can add more streams or participants to the SFU painlessly.
Added to this, the scope of implementing WebRTC for any developer revolves around three primary WebRTC APIs.
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The three primary WebRTC APIs are:
1. MediaStream: This API allows users to establish the mode of communication for sharing media streams among peers. Here, the local media stream grants the browser access to stream devices such as the camera, microphone, disc, or from arbitrary data sources such as sensors and other inputs to capture video, audio, and data streams.
2. RTCPeerConnection: Once the user decides on the stream of communication, the next step is to connect it with the other participants’ system. This is where RTCPeerConnection steps in. By leveraging STUN and TURN servers, RTCPeerConnection creates and negotiates peer-to-peer connections for facilitating direct data exchange among partner browsers for voice and video calls. Moreover, the various WebRTC codecs and protocols do most of the labor to initiate real-time communication even over unreliable networks, thereby saving developers from the complexities that lurk beneath.
3. RTCDataChannel: A network channel that equips browsers with the capability to share media synchronously among peers, with low latency and high throughput. It is an essential interface that enables bidirectional peer-to-peer transfers of arbitrary data. For developers, RTCDataChannel offers cutting-edge features to make the most out of the RTCPeerConnection and enable robust and flexible peer-to-peer communication.
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Examples of WebRTC
The first-ever web browser to provide embedded support for WebRTC functionality right out of the box was Google Chrome in 2012. This was soon followed by Firefox, Opera, and most recently by Microsoft Edge, and Apple’s Safari browser. WebRTC has emerged to become an integral part of the communication landscape today, with support from all the popular web browsers. At the moment, supporting WebRTC in these browsers comes in the form of 3rd party extensions and add-ons. Let’s briefly look into some of them:
1. Google Chrome: Chrome allows users to have control over the WebRTC API in their browser through its WebRTC Control extension. In addition, Chrome also offers WebRTC Leak Prevent and WebRTC Leak Shield extension for preventing WebRTC leaks. Developers utilizing Google’s WebRTC API can gain access to a broader array of voice codecs, which is a bonus for those who want to integrate their WebRTC applications with VoIP networks. For those users who want to prevent WebRTC leaks that occur during peer-to-peer communication, Chrome also provides a Disable WebRTC Chrome extension.
2. Mozilla Firefox: As an open-source project since 2004, Firefox realized the scope of the WebRTC project and took no time to extend support to it early on in the project’s initial stages itself. Firefox began introducing the three WebRTC APIs into its official source code in 2012 itself and now has fully integrated it into the browser. Today, WebRTC Chrome and WebRTC Firefox are interoperable, and also extend similar extensions, including WebRTC Control, WebRTC Leak Shield, and WebRTC Protect, along with a Disable WebRTC Firefox add-on to nullify the possibility for IP leaks.
3. Microsoft Edge: Microsoft announced the integration of WebRTC 1.0 API that supports H.264/AVC and VP8 video codecs for real-time communications in Microsoft Edge, enabling plugin-free, interoperable communications across browsers and mobile applications. The WebRTC 1.0 API aims to provide interoperability with legacy implementations on existing websites and other applications to initiate peer-to-peer audio and video streaming. Since it has already introduced support for object-real time communications (ORTC) in the Windows 10 version 1511 of Edge, the company holds no immediate plans to update the native WebRTC API further.
4. Apple Safari: In 2017, Apple became the last major vendor to release support for WebRTC on its Safari 11 browser, paving the road for platform interoperability in real-time communication. This proved to be a significant step towards WebRTC compatibility: gone are the unnecessary plugins on the desktop, and the need to send users towards Chrome or Firefox. With the addition of the Safari desktop browser to Chrome and Firefox, WebRTC is now accessible to more users for enhancing their enterprise communications fabric.
With the official support from all the major browsers and scores of applications, WebRTC looks all set to evolve continuously to satisfy the demand for efficient and painless real-time communication in this fast-paced age. By integrating the concepts of Object Real-time Communications (ORTC), new video codecs, and other changes, the future holds a wide range of applications for WebRTC not just in browser communications but also in non-browser devices, including mobile platforms and IoT devices.
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