WebRTC stands for Web Real-Time Communication and as the term suggests, it’s an exciting new technology which takes unified communications to the next level. At its core, WebRTC live streaming is essentially an HTML5 JavaScript specification. It contains communication protocols and a set of web APIs to establish uninterrupted, bidirectional audio and video communication between browsers – with all major web browsers being supported, including Mozilla and Chrome. This communication includes data streams, signalling, STUN/TURN servers, SDP, ICE, SIP, NAT, JSEP, network sockets, and more. The best part is all the APIs are completely plugin-free, so no tedious installations or downloads are required to start using live streaming WebRTC from your browser of choice.
For business purposes, this can especially prove useful, as anyone with a reasonably good internet connection and web browser can login right away and join the team.
WebRTC live streaming – Why it’s so important in this day and age
Since WebRTC live streaming is HTML5 compatible and it can be used to easily add real-time media communications between browsers and devices without any plugins, it has proven to be a valuable communication tools for internal and external company communications.
A WebRTC live stream enables very low latency communication between browser/device pairs over the internet and supports all popular browsers including Safari, Opera, Edge, Firefox, Chrome, etc. And, the fact that it’s an open-source project means that there’s a vibrant and dynamic ecosystem surrounding WebRTC – one where you can build some very unique products by gaining external help from companies such as Quickchannel. More on that towards the end of the article.
Perhaps one of the best parts about live streaming with WebRTC is that it is available via JavaScript in Opera, Firefox and Chrome, on desktop as well as iOS and Android.
WebRTC live streaming for Android and iOS is also available through native apps, along with browser and mobile APIs providing a number of integrated features such as camera/mic capture and playback to make live streams and webinars from the browser. In addition, WebRTC streams are also capable of handling data streams such as game data, so clearly, there are many applications of interest.
Even though WebRTC live streaming has the necessary features for peer2peer communication, additional features are needed for more complex functionality. For instance, a work from home webinar host wanting to reach out with a product presentation to a global audience – with nothing more than their computer and a normal internet connection.
Applications like these require the use of a WebRTC solution implemented in a streaming platform and a good one needs to provide the following at the minimum:
- Establish a robust connection between the host camera, microphone and screen sharing and the streaming platform – known as ‘signalling’ in WebRTC live stream language.
- Allow for one or two camera streaming with WebRTC – or one camera and a screen share to allow for presentations and showing windows from the computer to the audience.
- When streaming with WebRTC to large audiences the platform should provide engagement tools such as chat, moderated chat and polling.
- Enriched host experience with speaker name presentation, interactive agendas and overview of live broadcast and viewer count.
- Scale the number of participants or players to as many as required and deliver on a global scale.
- You should also look for a platform with additional features, such as, possibility to save your video stream and webinar presentation features.
If we stop to think for a moment, live streaming using WebRTC has become quite important in this day and age, namely because:
It’s an open source project
Making it highly suitable for developing or building applications for those looking to use the technology for live streaming.
It’s continuously improving and evolving
With each release WebRTC becomes more reliable, user-centric, and packed with additional features. The developer listens to community feedback and addresses issues each update.
Simple and straightforward peer-peer communication
Enabling developers to utilize the WebRTC live framework for building web apps which deliver a more complete user experience by facilitating video calling and video chats, as well as peer-peer sharing of files between same or different browsers.
Low latency and bandwidth consumption
The peer-peer communication framework used for WebRTC live streaming does not place a heavy demand on connection bandwidth and keeps latency low, so that users don’t face issues during file sharing, no matter what kind of connection bandwidth they have.
Embedded audio & video communication
The wide-ranging feature set delivered through live streaming WebRTC enables developers to easily embed audio and/or video communication within their web apps.
Secure data exchange
WebRTC secures every piece of data from the source to the destination, where the latest encryption protocols are used to secure the respective components within each data stream. For added security, JavaScript apps are accessed only through local hosts or https.
Support for a broad range of mobile devices and web browsers
Helping developers embed audio/video with ease through HTML5 tags, because WebRTC enjoys support from all major browsers currently available and works seamlessly with iOS, Android and desktop devices.
