How to Integrate Video Calling and Chat API with WebRTC?
Basically, this means WebRTC supplies real-time communication capabilities and interacts with the other components of the application. It gives the ability to host video chat apps online, allows video call settings such as paired video conferencing, scheduling, etc. This integration provides users with a powerful platform for providing and receiving services and gives them an added value over using a third-party video conferencing tool.
Still have doubts? Let’s understand from the beginning, so what exactly is WebRTC?
What is WebRTC?
WebRTC stands for Web Real-Time Communication. It is a technology that allows web applications and websites to capture and stream audio or video as well as to exchange arbitrary data between browsers without requiring an intermediary.
WebRTC is a free and open-source project that enables web browsers and mobile devices to communicate with each other. It offers users the ability to communicate from within their primary web browser without the need for complicated plug-ins or additional hardware.
The WebRTC was announced by Google in May 2011 as a means of developing a shared set of protocols for enabling high-quality RTC applications within browsers, mobile platforms and IoT devices. During those times, Flash and plug-ins were widely used and were the only methods of offering real-time communication. Nonetheless, after a considerable amount of work was put in, the first cross-browser video call was established between Chrome and Firefox. Since then, support for WebRTC in the developer community has risen steeply as more and more organizations add their support for the specification. Currently, WebRTC is available within Chrome, Firefox, Safari, Edge, Android,, and iOS and is a widely popular tool for video calling.
Understanding the WebRTC Streaming Architecture:
Video calling is possible when two or more devices are connected using the WebRTC protocol. There are mainly two modes of connection
1. WebRTC Peer to Peer Audio/Video Connection
Peer-to-peer communication for WebRTC enables the exchange of video and audio content between browsers. Unfortunately, the purely direct exchange may not always be possible, as a browser may be located behind a symmetric Network Address Translator (NAT). NATs force WebRTC applications to use TURN servers located on the public Internet for forwarding media data between browsers. This type of architecture is simple to implement and has low application operating costs, because lesser backend infrastructure is required.
2. WebRTC Relayed Audio/Video Connection
Relying on peer-to-peer connections for browser video chat is not advisable while developing production apps. Normally, apps built on such ICE frameworks do not connect as users have advanced LAN security. Relatively, developers set their RTC Configuration to first attempt peer-to-peer, and then fall back to relayed connection if peer-to-peer fails. In such cases, developers do not need to decode/recode streams, they could just follow the below procedure, which is to –
· Use the Broadcaster to connect to the server (peer to peer)
· Allow the clients to connect to the server (peer to peer)
· Enable the server to relay the stream from the broadcaster to the clients.
This means when the STUN server fails, ICE uses TURN server, which is a STUN server with some extra relaying functionalities that can traverse symmetric NATs. Initially, ICE attempts to use STUN servers, but TURN relay servers are needed for very restrictive corporate networks. TURN relay servers can be bought by a service provider such as MirrorFly.
Creating Multi-User Video Chat with WebRTC
WebRTC needs signaling to set up peer-to-peer communication between two nodes. This means ultimately, if a client joins a room that has already been created by another client, the media exchange between peers will begin and will be managed by the following socket event callbacks or functions.
Do Not Build a WebRTC Signalling Server
WebRTC is a fascinating technology that goes beyond the common computer connection protocols and offers excellent communication pathways between a client and a server. However, building WebRTC signaling servers that function aptly is extensive and expensive. However, it can be full-fledged with the right service provider – someone like MirrorFly. MirrorFly offers Signalling WebSockets to developers so that they can set up call connections between the client servers. The client connection must adapt to each other by sending messages, data, and media over the public IP address of both the client’s WebRTC signaling servers. Moreover, with an open-source WebRTC library offered by MirrorFly developers can swiftly build and competently integrate video calling and chat APIs with WebRTC. So, it’s best not to waste time and money in building a WebRTC signaling server.
Engage In Open Source WebRTC Video Chat Systems
Engaging in an open-source WebRTC software project provides developers a platform suitable for creating modular applications with advanced real-time communication capabilities. WebRTC media server and a set of client APIs make it simple for the development team to build advanced video applications for WWW and smartphone platforms. From group communications, transcoding, recording, mixing, broadcasting, and routing of audio-visual flows – everything is possible with an open-source WebRTC video system. Look for WebRTC video systems that also provide advanced media processing capabilities involving computer vision, video indexing, augmented reality, and speech analysis to provide clients with the ultimate experience. Concludingly, WebRTC streaming architecture should allow integration of third-party media processing algorithms (i.e., speech recognition, sentiment analysis, face recognition, etc.), which can be transparently used by application developers along with the built-in features.