WebRTC protocol has revolutionised the way real-time communications are designed, developed and deployed. Most real-time communications solutions are being created on top of the WebRTC framework.
WebRTC-based audio/video solutions can be challenging to build. But there is good news for you. It lets you create these solutions without coding them.
In this post, you’ll know various possibilities for developing such a solution to build a feature-rich solution quickly.
What Is WebRTC And How Does This Work?
WebRTC is a free, open-source framework developed in 2011 by Google to facilitate peer-to-peer data transfer without plugins, native applications or third-party proprietary software. It allows adding real-time communications capabilities to your application and is supported by all major browsers.
Also read: The Most Comprehensive Guide on WebRTC
When developing a WebRTC-based application, a developer needs to consider the entire workflow of the application when connecting to remote users, and this is where things can be complicated.
Here are some broad scenarios that need to be taken into account while embarking on the path of building a power-packed audio-video real-time communication:
|UI/UX||Standard Features||Functional work flow / Scalability/Stability/Reporting|
|Visually appealing UI/UX||Basic audio/video sessions||Call workflow: app– app calling/ in–room calling|
|Covering all possible scenarios: low bandwidth, user disconnecting unexpectedly because of browser termination, remote user network failure.||Security, Lobby, moderated entry, restricted access/ forceful dropping, remote mute-unmute
Scalability: horizontal scaling in terms of the number of participants. For example, if you need 250 + participants in a single video session or 1000+ participants in a single webinar session, you require good horizontal scaling capability.
Vertical scaling: the number of meeting rooms is a good example of vertical scaling.
|No UI/UX interference with an existing applications.
What to do in the case a call is received during a WebRTC session.
|Screen sharing Screen share overriding, multiple screen sharing||Stability: no downtime / Future and backward compatibility with browser and mobile apps.|
|UI/UX customisation and personalisation: language selection.
|Reporting: detailed auditing and quality logs for the audio/video sessions.|
|Device compatibility and responsiveness: browsers on different machines/mobile apps/ mobile browsers.|
EnableX Low Code Video Embed And Visual Builder
Developing applications that run on multiple platforms requires knowledge of the skills involved in cross-platform development, such as web, mobile, and WebRTC. The reason is building such applications can be difficult, platform-as-a-service (PaaS) providers can provide low-code or no-code solutions to facilitate the process.
EnableX Low Code Video Embed solution and Visual Builder is currently the most advanced video/audio WebRTC based communication platform.
Some of the salient features are:
- Screen sharing/annotation allows participants to annotate over a screen share.
- Two-way annotation during a live video session
- Audio join in / audio calling to a phone number from the video room
- UIkit for mobile frameworks allows you to develop video-based apps in no time for mobile frameworks
- Picture-in-Picture mode for screen sharing allows you to view other participants while you are surfing other websites
- Multiple layouts for video meeting rooms
- Customisation/personalisation of audio/video rooms
- Visual builder –configure your UI
- Customized backgrounds
- Multiple layouts for recording
- Live recording/ Live streaming/ Multiple live streaming/ Transcoding/Water marking/ encrypted recording
- Language personalisation
- IP zoning
I hope you’ll have a fair idea of the different scenarios by now. To gain hands-on experience on how to use Low Code Video Embed solution to build a real-time audio/video solution, try out EnableX Low Code Video Embed.
Build something exciting!