As communication technology develops, media streaming becomes more and more common on the web. Anyone who is interested in media streaming has definitely heard of WebRTC technology. In most cases, as well as WebRTC, the WebRTC server will be a major concern. In this post, we will introduce WebRTC Technology and WebRTC Servers without any technical details.
WebRTC is a free, open project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. Namely, WebRTC enables for voices and video communication to work inside web pages. The WebRTC components have been optimized to best serve this purpose.
WebRTC is HTML5 compatible and you can use it to add real-time media communications directly between browser and devices. And you can do that without the need of any prerequisite of plugins to be installed in the browser.
WebRTC presents a protocol that enables ultra-low latency communication between pairs over the Web by using web browsers without plugins. Almost all of the popular browsers (Chrome, Firefox, Safari, Edge, Opera, etc.) comes with WebRTC support without any plugin dependency.
As it is an open-source project, this has created a vibrant and dynamic ecosystem around WebRTC with various open source projects and frameworks as well as commercial offers from companies to help you build your products.
There are several other protocols for media streaming, like RTMP (Real-Time Messaging Protocol), HLS (HTTP Live Streaming) and MPEG-DASH (Dynamic Adaptive Streaming over HTTP). However, WebRTC has some key features that make it a promising technology such as:
Although WebRTC has adequate and necessary features for peer to peer communication, further features are required for more complex applications. For example, an online education application that requires a classroom and one-to-many communication capability. Applications having such requirements have to use WebRTC server to use WebRTC. A good WebRTC server must provide such features:
- Establishing the connection between the caller and callee. (which is called signaling in WebRTC literature)
- To keep the call alive without any interruption under the low quality of the connection. Adaptive bit-rate is the solution to this problem.
- Establishing one-to-many or many-to-many communication.
- Scaling the number of players up to high numbers.
- Having extra features like saving the video call.
Ant Media Server supports most of the common media streaming protocols like RTMP, HLS and of course WebRTC. Actually, Ant Media Server is one of the best WebRTC servers on the planet. Ant Media Server provides all of the features listed in above. Ant Media Server is able to provide WebRTC publishing with ~0.5 seconds latency. If you like to experience Real-Time Video Streaming, then you need to arrange a demo from Ultra-Low Latency Adaptive Live Streaming Server from Ant Media.