We are past the age of video calls being nothing more than just a means of connecting with friends and relatives. Today, video conferencing is becoming increasingly popular with enterprises, and is used as an integral part of workplace collaboration. While video calling platforms meant for consumers, such as Skype and Facetime, have shown how video calling can easily be arranged in a matter of minutes, we can’t say the same with regard to enterprise-grade video calling solutions. Until now. Web real-time communications (WebRTC), allowing users to make video calls from their web browsers, has changed the scene. WebRTC offers video calling capabilities to anyone with a webcam and browser–its as easy as clicking on a link to launch a session.
Microsoft’s recent announcement of the launch of Skype for Business is timely, and browser-based video is top-of-mind for businesses today. We are all excited about how plugin-free video conferencing would feel – without the hassle of downloading several apps, without the frustration of failed logins, with easy-to-launch sessions, with smoother operations, and so on. But perhaps, the real concern lies elsewhere – with network support, call quality, security, and video clarity.
In order for video conferencing to really meet our expectations in terms of improving collaboration and reducing the need for expensive in-person meetings, there are factors we must consider: video and sound quality, and network reliability top the list of concerns that must be met before we assume that new solutions like Skype for Business, or any other browser based video, will be sufficient to meet our needs.
WebRTC has different network requirements
As opposed to conventional video conferencing systems, WebRTC comes with a different set of bandwidth requirements. While WebRTC-based video uses more bandwidth than IP telephony or real-time collaboration, the network source for such types of video is Internet bandwidth, and not LAN or private WAN bandwidth that traditional video services use. A rock-steady Internet connection is a prerequisite for WebRTC-based video sessions, no question about it. In addition, you need to have sufficient bandwidth available to carry out the number of video calls necessary to meet enterprise-level requirements.
The brighter side of network usage in WebRTC video is that it is able to withstand changing network conditions. As a result, users can enjoy steadier video conferencing experience with fewer call drops. In addition to bandwidth efficiency, WebRTC-based video also consumes less processing power than conventional desktop video.
Audio and video quality offered by WebRTC
WebRTC makes for better and more effortless communication, and has video quality that is much better than Flash, though not to the level of high-definition videos. When using conventional desktop video conferencing, launching or joining a session on an older endpoint would invariably end with a poor quality video. WebRTC-based video has reduced that risk to a great extent because of the technology it employs.
WebRTC includes a whole set of voice communications tools, including software based acoustic echo cancellation (AEC), automatic gain control (AGC), GIPS (Global IP Sound), noise reduction, noise suppression, and hardware access and control across multiple platforms. Not only will WebRTC mean more audio clarity, but its video capabilities will be at par, as well.
Many of the enterprises that have already adopted WebRTC think that the real benefit of using this video service is in its simplicity. With its one-click nature, WebRTC-based videos take the hassle out of setting up and launching sessions, unlike other video conferencing services like WebEx and Miscosoft’s rebranded Skype for Business (a/k/a “Lync”). How well can WebTRC compete against these services with regard to call quality and network-usage? We shall soon see.