“WebRTC is Popularizing in the Global Market and estimated to grow at a CAGR of 34.37% with rocketing the marketing value of about $ 300 Billion”
With remembering a couple of years ago, developing a video application on the web chat was relatively a solid strain for developers. Gradually, the days of utilizing the Flash technology and some licensing codec was fading and the real-time video call chatting landscape was thriving the chat experience to next level of technology.
What does MirrorFly bring Web Real-Time Communication (WebRTC) to you?
Browser & Mobile signaling in WebRTC
Chrome, Firefox, Opera
Android & iOS support
Before getting into the building of a real-time WebRTC video call app, let us be clear in identifying the functional dependencies to be used to build WebRTC Android/iOS and Web chat App.
Thorough WebRTC Library:
A Complete WebRTC native code package is required for every developer who wishes to integrate WebRTC for browsers and WebRTC API for Android/iOS chat app development.
jQuery is used to handle the event, manipulation and simpler the HTML with the easy to use of APIs that works on several browsers.
Semantic UI CSS:
A framework used to contrive responsive layouts through human-friendly HTML with an elegant CSS framework that works on embellishing the user experience of the Chat Platform.
A Compatible template provides the potential power to build semantic templates in HTML language that brings significant changes in the mobile devices to embed simple WebRTC messaging platform on Android and iOS platforms.
Apart from these dependencies, there are several frameworks and functionalities required to build a chat app for video and voice calling enabled with WebRTC.
How Video/Audio Transmission of Chat Application takes place?
Initially, both the video and voice call depends on the potentiality of streaming media between two client server connected to each other. Voice Over Internet Protocol (VoIP) is one of the most familiar and trusted standard technique for voice and video calling over the Web.
As we are very much aware that, WebRTC is the significant implementation for streaming of media content from one client-server to another.
- STUN Server
- TURN Server
Signaling – Peer to Peer Connection
Signaling Websockets are used to setup call connection between the client servers. The client connection must adapt to each other by sending messages, data and media over public IP address of both the clients webrtc signaling servers. To get the IP address, We use STUN Server.
STUN Server – Local IP Address
With the use of NAT (Network Address Translation) which provides the local IP address of the device to have peer to peer connection. To make this connection using WebRTC, we need to get the public IP address where STUN Server provides it.
TURN Server – Mediator
This server acts as a mediator to connect both the clients if case peer to peer fails. It sends data from one client to another over signaling process. This methodology works for webrtc video and Audio calls on android/iOS chat app and also for media to create support for the messaging applications.
If you are about to integrate WebRTC into your android/iOS chat app then probably you must have a clear idea on three WebRTC APIs which plays crucial in developing your WebRTC Video/Voice Chat on Android or iOS. Now, let us dig deeper into WebRTC APIs.
A Comprehensive Insight into WebRTC APIs for iOS/Android Chat Application
This allows you to obtain the access over the camera, screen or microphone of the device used by the user. It also provides additional layers as a security where it only allows access if the user is connecting from an (HTTP) secure connection which allows the user to stream their entire media library. This getUserMedia have three parameters namely:
- A Constraints object
- A Success Callback method
- A Failure Callback Method
This webrtc allows the user to communicate directly to have a peer-to-peer connection in order with the transcoding of the media files. It encodes and decodes the entire media content and voice/video chat that is sent to the remote server and from your local machine in receiving your media files.
It helps in transferring the data directly between the two peer users connected in a bi-directional data channel. This also helps in creating a secure connection to send data at a real-time function.
Some of the Highlights that includes in implementing the Interface:
This allows the user to communicate over text in a real-time chat experience on all the Webrtc audio and video enabled iOS/Android chat apps.
Based on Voice-over IP technology in real-time over the Internet through Chat Application. Low latency is carried to make 1 to 1 or 1 to many webrtc voice chat app connections across all the devices.
Video Connection helps in making quality webrtc enabled video and audio calls on android/iOS chat apps continuously at low latency. With adaptive bitrate which reduces the complexity of video/audio chat at a high pixel rate.
These methodologies are quite bewildering right! To make this more compatible and easier, MirrorFly is inbuilt with all these technologies and features embedded with WebRTC that in resultant to deliver a highly successful WebRTC enabled Voice/Video Chat App that runs seamlessly on iOS/Android and Web browsers.
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