How WebRTC Is Driving the Success of Video Conferencing
Over the last few years, video conferencing has changed the way businesses connect with their customers, suppliers and employees. It is used to facilitate the most effective and efficient communications in a world where swift and secure dissemination of information is critical to success.
WebRTC is one of the main forces driving the widespread adoption of video conferencing technology in today’s world. Read on below to find out how one-to-many video conferencing with WebRTC is facilitating easy, affordable, and effective online video meetings.
What Is WebRTC?
WebRTC stands for Web Real Time Communications. This is a recently developed standard for audio and video communications that is changing how businesses and individuals communicate. The open-source protocol, developed by some of the biggest names in the IT world including Google, Opera, Cisco, Microsoft, and Ericson among others, enables the sharing of voice, video, and file data through a normal standard web browser.
This type of capability helps eliminate the need for dedicated solutions and plugins that usually result in vendor locking, thus reducing the cost of video conferencing for users while also making it much easier to implement and use.
It is clear to see how WebRTC makes it easier and cheaper for businesses and individuals to meet and interact with each other online, face-to-face.
Read on to find out more on how you can use simple WebRTC-based video conferencing to create more powerful online meetings.
WebRTC Is Safe and Secure
When it comes to hosting successful online meetings, security and reliability of the solution in use are essential. WebRTC is designed with a number of impressive and highly effective security features meant to ensure safe and reliable communication between all users at all times.
With the number of RTC end points expected to go over the six billion mark in 2019 and to continue growing exponentially into the future, many existing and potential users expect the protocol’s security and reliability capabilities to be pushed to the limit. However, with the built-in security protocols described below, WebRTC is more than capable of supporting the growing number of users, facilitating safe and secure communication during online meetings.
For starters, choosing to use WebRTC saves users from downloading malicious software from the internet, thereby protecting them from related security threats. To use this protocol, users are not required to download any third-party solutions; everything that is needed to support web real time communication is already built into your browser, whether you use Chrome, Firefox, Opera, Edge, or other major browsers.
In addition to the above safety measures, WebRTC developers also added a few important security features to the protocol.
- • For instance, to prevent eavesdropping, users must activate their camera and audio access through a popup as part of the camera and microphone security protection feature.
- • Regardless of which browser you are using, you get to enjoy end-to-end encryption between all endpoints.
- • Datagram Transport Layer Security encryption is used for all the data sent through the technology at all times.
- • To protect users from hacking attempts, web real time communication standards also employ the use of Secure Real Time Protocol.
As you can see from the above, WebRTC-based video conferencing is safe and reliable when it comes to conducting the most secure online meetings. All voice and visual data are secure from the point of origin, during transmission and up to when it reaches the right recipient. MegaMeeting also employs several additional security protocols when it comes to securing our online meetings, such as utilizing individually created unique meeting invites and locking down the recording of meetings.
High Quality Video and Audio
To be as close as possible to the real thing, online video meetings are supposed to capture all the nuances of real-life face-to-face meetings. Changes in tone of voice and facial expressions all play a huge role in facilitating effective communication. While online video meetings make it possible for individuals, including business representatives or employees, to interact face-to-face, albeit virtually, WebRTC ensures that only the highest quality video and audio is captured and transmitted to facilitate the most effective communication.
In the past, before the introduction of WebRTC, users in need of video communications relied on solutions designed by specific companies. These standards usually required all users to be on the same platform or even using the same proprietary hardware. WebRTC, a revolutionary protocol, changed all this, introducing a new and more efficient norm.
WebRTC standards have removed platform and device specificity from the equation. This simply means that any obstacles that were initially hindering online video communications have now been eliminated in one fell swoop as attendees can now use the device of their choice (desktop, laptop, tablet, or phone).
WebRTC is generally designed to make it possible for users to make audio and video calls in real time using their device’s browser. When it comes to making multimedia communications via the internet in real time, these new standards do not rely on a specific platform or device. This means that no one is left out, provided that they have an internet-enabled device and active access to the internet.
All in all, WebRTC is not platform or device specific.
Online WebRTC video conferencing is designed to enable individuals from different locations, whether in the same geographical region or across oceans and continents, to connect with each other in real time via an internet connection. However, this brings in the question of signal strength. Web real-time communications standards strive to give every user on each endpoint the best possible video and audio quality given the constraints of their network.
While it is true that WebRTC is designed to offer high-quality video and audio transmission, as previously discussed, this might not be standard across all endpoints. To avoid disconnections or any unnecessary interruptions or lag in the real-time feed, WebRTC automatically adjusts the audio and video profile to match the capacity of each endpoint’s network connection. The receiving browser’s network conditions are sent to the sending browser via the receiving browser’s Secure Audio Video Profile with feedback and Multiplexed Real Time Protocol Control Protocol embedded.
Simply put, WebRTC dynamically adjusts the quality of video and audio sent to each endpoint to match the specific network conditions, resulting in a higher percentage of meeting attendees having a better overall experience in their online meetings.
An Investment into The Future
One of the best ways to guarantee the success and longevity of your business is through the adoption of the most relevant and reliable solutions and processes. Even though RTC only came onto the scene a few years ago, rapid development and widespread adoption are working to make it an industry standard, preferred over other available alternatives in the video communication field.
WebRTC came into being due to the focused efforts of some of the biggest and most successful brands in the IT world. This means that with such strong backing, this standard is set to grow more quickly and outlast other available video communication solutions in the market right now.
By adopting this new and highly effective communication protocol, businesses can rest assured that they are relying on a product that enjoys the favor and unwavering focus of the best brands in the area of online communication.