WebRTC for Mobile Application Developer
In this article we will learn about WebRTC and we will learn about its key features, pros and cons, and why we need it, where can we use it, and how it works.
so let’s start it so what is WebRTC,
WebRTC is an application layer technology that allows web browsers to stream audio and video media and to extend useful data between browsers and mobile platforms or IOT devices.
In webRTC, you can use plugins for audio video communication on both desktop and mobile browsers, so there is no native app required for audio video communication. Additionally, it allows peer-to-peer communication on your web page. Therefore, webRTC has become an essential tool for communication and data sharing, as it encourages the exchange of information between two different sources.
so let’s understand the key feature for that webrtc,
WebRTC uses three primary components to initiate a peer-to-peer interaction:
Firstly, we will use the media stream, also known as get user media, to access and control the user’s camera and microphone.
Using peer-to-peer connections facilitates direct communication without the need for intermediaries.
Secondly, we have the data channels, which are basically peer-to-peer exchanges.
Let’s understand the pros and cons of our webrtc,
Let’s start with the pros first. The first one is better sound quality because webRTC provides built-in noise reduction and eco cancellation, as well as automatic microphone sensitivity adjustment. Due to this feature, webRTC calls are exceptionally clear as compared to other technologies we use and webRTC also uses modern audio video codecs to deliver outstanding call quality even on slow connections,
As for open source technology, open source code gets reviewed and quality controlled quickly by the webrtc community, but it has also resulted in widespread adoption,
Despite the fact that webRTC is based on the C++ architecture, developers can use the built-in Java script API layer to develop webRTC solutions quickly.
Basically, webRTC is protected by few mandatory encryption specifications and this provides end-to-end encryption for any data that is sent through a webRTC peer-to-peer connection,
Then we have no plugins that need to be installed on the end user, unlike most real-time communication technologies that require a plugin to make calls through the browser; however, with the use of this webRTC, we do not need any plugins at all, we can simply use the simple javascript API and all.
Now let’s understand the cons also.
Because webRTC is a new technology, it is currently under development, and the current version is a working draft. This means the webRTC source code may undergo significant changes in the future. Whether or not the future version will be backward compatible with the previous version remains to be seen. Additionally, the documentation for this webRTC is very fragmented, so questions may arise.
Do we really need webrtc,
The answer for this question is yes, we need webrtc because most people have the browser that supports webrtc and the webrtc works without any plugin or browser extension. From an end user point of view, webrtc is a click-and-play experience that is simple, easy, and quick. No plugins are needed, you just need to have a browser or mobile application to use it.
Use Cases for WebRTC
In addition to using that web RTC in our online education, we can also include it in our online education portal, so the platform is always available and the instructor and students can interact in real time regardless of how busy that RTC is.
Our telemedicine software can also use the video calling feature and we can give that to doctors and patients there also.
As we have a team communication and collaboration solution, we are also able to provide that for any teamwork we have or anything like that. Instead of using any ready-made software like Zoom Mating or Microsoft Team, we can create our own solution with that webRTC for team handling, communication and collaboration.
also we can use that webrtc in our gaming entertainment purposes.
How does WebRTC Work
so here you can see we have client a and plan b and they want to communicate with each other for that we have to use something like signal server and stun server and everything suppose client they want to connect to client b and client a first need to determine all the possible way that public can connect to it in the same way client b also need to determine how the public can connect to it client a and client b signal this session information sdp that we will discuss later on to each other with the help of that websocket and also they can use http also because that sending information doesn’t matter how we are sending it then client a connect to client b via most optimal path that is ic that also we will discuss later on then a and b also extend their supported media types and the security information after that an agreement will be made between the client a and client b and the connection will be open so now let’s understand what all this information is like signal server sdp stunt server then we have our ic and all.
To build up the framework, there are several steps involved as webrtc protocol is a combination of server technologies.
Our first step is signaling, then we connect, then we secure, and then we communicate.
so our first step is Signalling so it’s referred to setting up and controlling a communication session the peers connecting to a real-time communication send their stream to the server with the server in quotes and delivered to the receiving peer so basically signalling protocols are a way to coordinate and control the communication between the peers it’s user sdp that is session description protocol which is in a plain text format and containing the list of media sections following information are shipped by the signalling the location of the peer which is the IP address of both the agents client a and client p and whatever the agents which we are connecting to each other then we have our consuming audio and video track that an agent comes across then we have the producing audio video track which agent transfer then we have the data channels that determine the media type with a resolution exchange.
It is then time to get to Connecting. Connecting refers to securing a bidirectional communication between two peers in WebRTC. WebRTC communication occurs on a peer-to-peer basis rather than between clients and servers. In essence, it’s a framework for connecting peers, such as two article chat clients.
We have an efficient method for solving these difficulties that ensures that the best possible connection is maintained between the two peers, even if the locations are difficult to connect to. In this case, two servers are used, stun and turn. Stun is basically a set of network protocols and methods that allow an end host to find its public IP address, since it is a session traversal utility for the internet.
Whenever it is behind the network, we have the turn server that traverses the network using relay, and these servers help us establish a connection between the two agents when a direct connection is not possible due to firewall restrictions. As a proxy between the two clients, they created a temporary IP for the agent to generate traffic to and from.
then we have the next step which involved with the Securing and WebRTC ensure the security and it’s ensure that the communication share between the two agents are encrypted and remain confidential with with any third party and we have two types of protocols used over there first one is a DTLS that is data-gram transport layer security and which allow webrtc to establish the secure and encrypted communication between the two peers client and the server to communicate need an agreement on certain values known as cpr in a dtls handshake and it’s secure the data stream of the ps and dtls is required.
then we have SRTP that is cure real-time transport protocol and it ensure and encrypt a media stream between the two connecting peers and it initialized by using the key generated by the dtls and this protocol is specifically designed for encrypting the rtp package.
and the next one and the last one is that is Communication and basically webrtc is developed for transferring data audio and article over the web and this technology allow sharing of unlimited data over the web it’s allow a user to make activities such as adding or removing the stream anytime over a call and this stream could be bundled together with two core protocol of webrtc communication and we have two types of protocol over there first one is RTC (real time transfer protocol and this protocol is designed to carry real-time delivery of the article and let’s give the agent stream which can be run over the multiple media feeds from one connection and this protocol does not ensure the media transfer cover latency and a reliability but it’s provided tool to implement them then we have RTCP(real-time control protocol) and its allow administrator to monitor the quality of the calls from the collected metadata and this protocol allow an agent to add any metadata they want to communicate no statistic of the cause this protocols also track the package loss latency and the other yp concern so if there is any connection issue there is any article delivery issue anything like that that can also be track over there but these all the steps which involve in the webrtc for having connection everything.
As a result, we will only cover the basic information about WebRTC regarding mobile applications, its features and pros and cons, as well as how and where it works. That’s all I have to say about WebRTC regarding mobile applications. The information was taken from a variety of websites as some research. If you find any errors or misdirections, please point them out or comment below.
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I am Shirsh Shukla, a creative Developer, and a Technology lover. You can find me on LinkedIn or maybe follow me on Twitter or just walk over my portfolio for more details. And of course, you can follow me on GitHub as well.
Have a nice day!🙂
