WebRTC is an exciting new development in the physical and virtual world. It can provide so many capabilities and applications, especially when it comes to video conferencing and real-time communication by merging a browser with an IP camera. Not only is this a boon to application developers, but it’s also a great tool for embedded systems developers. It’s very small in size and data consumption, and it makes use of open computer technologies like the GStreamer framework for audio/video streams, trumping HLS and MPEG Dash in performance thanks to its IoT tilt.
This article will provide an overview of WebRTC, discuss its benefits, examine its viability in IoT, and define a WebRTC interconnected device. We’ll also go through how to use the GStreamer WebRTC framework to integrate this technology with smart devices.
WebRTC Technology Demystified
Web Real-time Communication (WebRTC) is a free-to-use ecosystem, created by Google and Ericsson. The technology is a collection of web APIs that lets you build communication tools for the web. It’s a framework that lets you add in-browser video, voice, and data channels to your website, so users can call each other directly—no third-party plug-in or software required. Applications built with WebRTC will just work, right out of the box.
WebRTC is already supported by Chrome and Firefox, and it’s coming soon to Safari (and possibly Edge).
In a nutshell, WebRTC establishes user-to-user communication between web or mobile browsers with no added plugins. It accesses a gadget’s microphone and camera, and is capable of streaming multimedia files with only a half-second delay. Worldwide, it’s considered the foremost real-time media file transfer innovation.
The emergence of remote work has ushered in a new phase of real-time application development with WebRTC:
- Audio/video conferencing applications (e.g. Google Hangouts, Zoom)
- Team productivity tools and collaboration platforms (e.g. Slack)
- Video streaming platforms (e.g. Netflix)
How WebRTC Works
WebRTCs don’t use the same communication commands that browsers follow to surf websites. The reason is that the users’ computers or smartdevices are safeguarded by firewalls. Unlike HTTPS sites, whose location is throughout the Internet, computers and smartdevices don’t have permanent web addresses. Hence, to establish a communication session between two people, browsers should find one other and request permission to swap media information in real time.
WebRTC also relies on the following signaling/communication protocols:
- SIP (Session Initiation Protocol) initiates, manages, and terminates real-time interactions.
- The Session Description Protocol (SDP) is in charge of media file transmission.
- The Interactive Connectivity Establishment (ICE) protocol generates and determines the shortest path for media transmission between devices.
How can WebRTC be Utilized?
WebRTC is a standard technology that’s built right into your browser, and can be used to replace some types of real-time communication. It provides peer-to-peer voice, video, and data transfer directly between browsers. No plug-ins or downloads are required; all you need is an HTML5-enabled browser on any operating system (Windows, Linux, Mac OS X, iOS, Android) and a webcam or microphone.
Benefits of WebRTC
WebRTC’s flexibility allows any company to improve their business communication tools using fast and secure web applications, say experts at the Microsoft Innovation Lab.
- The leading four browsers: Chrome, Safari, Firefox, and Edge, all support the technology.
- WebRTC is customizable and versatile: it allows for the transfer of data other than media content.
- WebRTC is designed for security: the technology encrypts data exchanged between devices using the Datagram Transport Layer Security (DTLS) and Secure Real-Time Protocol (SRTP), and notifies users before accessing a computer’s camera and microphone.
- The technology is web-based, which makes writing the client side of a program easier.
- Using the real-time communication engine, developers may “overlay” different forms of data on top of audiovisual material.
Limitations of WebRTC and IoT integration
For enterprises to adopt WebRTC and integrate it into their systems, a robust offering must be deployed across RFID-tags, antennae, video conferencing and other enterprise communication applications. A scalable infrastructure solution must be available to manage thousands of WebRTC users accessing the same server. An inter-protocol gateway allowing multiple protocols to interact with WebRTC is a critical prerequisite for large-scale deployments; in addition to TCP traffic, WebRTC requires support for UDP, HTTP, TCP/TLS and STUN traffic.
Consequently, no business or consumer-facing IoT solution provides WebRTC out-of-the-box currently.
Surmounting the Limitations of WebRTC using GStreamer
The original WebRTC Native APIs lack flexibility and can be inefficient. Here’s where GStreamer comes in handy. It’s an open-source network-based ecosystem for creating multimedia streaming applications for connected devices, desktops, and servers. It also has a native WebRTC API in its feature set.
The GStreamer architecture is analogous to a plumbing system, with water replacing media data and GStreamer pipelines serving as the pipes. These pipes have the potential to alter the quality and amount of water as it travels from the public water supply (device one) to a residential plumbing system (device two).
Supposing the source device is capable of reading video files. To divide outgoing traffic into audio and video data streams, we can construct a pipe bend (GStreamer demuxer). The data is decoded along the pipeline using h264 (video) and Opus (audio) and sent to the target device—specifically, its video and audio output components—or to the cloud, where it may be evaluated using machine learning programs.
Those functional pipe bends are referred to as elements in GStreamer. They are classified as source elements, which generate data, and sink elements, which take it. In turn, the components have pads—interfaces with the outside world that allow developers to link elements based on their capabilities.
Enterprise Applications of WebRTC through GStreamer
We can create smarter home automation and business security systems, among other things, using WebRTC and GStreamer. Let us examine three scenarios where companies from various segments can potentially deploy GStreamer in their embedded software development projects.
- Smart supervision: While traditional businesses and CCTV innovations are relatively familiar, they can’t initiate independent actions and require 24-hour monitoring.
- Connected Augmented reality: An offshore engineer can obtain real-time feeds via a smart headset through an onsite remote operator, who sets digital markers within a connected Augmented Reality.
- Smart transportation: In the automotive segment, WebRTC assists engineers to develop an elevated navigation experience through a mix of in-vehicle cameras and sensors, in high-frequency connection with a built-in computer.
All in all, WebRTC is here to stay, and it’s ready to become a powerful tool for IoT and embedded systems as it binds with GStreamer. Its prospects look bright, offering wholescale industrial applications for remote machine maintenance home devices, telemedicine devices, connected cars, and wearables.