During the covid epidemic, physical interaction between people reduced, and video conferencing was considered as a potential tool for maintaining excellent communication. As a result, people found that having online communication among themselves was a more convenient medium than physically seeing someone. Video conferencing had stretched its reach across numerous demands, from online coaching to birthday celebrations.
WebRTC is one such protocol that allows users to communicate visually and audibly without the need for a server or third-party software.
What is WebRTC
WebRTC stands for Web-real time communication.Google released WebRTC, an open-source project for browser-based real-time communication, in May 2011. WebRTC is a free and open source software that enables an open framework for the web that enables Real Time Communication between different browsers which includes fundamental building blocks for high-quality communication on the web. It enables voice, video components, and peer-to-peer file sharing across browser and mobile applications without the use of third-party plugins or software. When these components are implemented in a browser, it can be accessed through java script APIs, enabling developers to easily implement their own RTC web app.
It uses multiple protocols and HTML5. It is being standardized on an API level at the W3C and at the protocol level at the IETF. It is supported by browsers like: Google, Mozilla and opera.
Detail explanation of its working
The WebRTC communication is held by combination of several technologies and steps.
Let us know what are the steps and processes that are being followed in order to set a secured communication and build up the framework. But before that keep in mind that no steps can be bypassed i.e. to make this framework work, preceding steps should be completed.
It is bi-idirectional communication in which a client sends its information to another client through a signaling server. The information is transferred using SDP protocol, which is in plain text format and contains a bunch of media sections. It contains information such as:
- Peers location i.e. IP address of both the clients.
- Audio and Video tracks of the clients
- Data channels, which determines the media type.
There are two parts of the signal server that are STUN and TURN servers, in order to have this peer to peer communication.
Step#2 Peer Connection
It refers to securing a bi-directional communication between two peers. Here no client-server is required to establish the communication. It is a difficult task as there are different network protocols and transport addresses of the peers. But it is solved by the use of ICE and NAT protocol servers. To ensure best possible ICE uses STUN and TURN servers for the connection
WebRTC takes care of security of the communication that takes place between two peers.It remains confidential to any third party. To ensure confidentiality it consists of three encryption specifications: Secure Real Time Protocol (SRTP), secure encryption key exchange, and secure signaling.
SRTP encrypts the data sent through real time communication.DTLS-SRTP (Datagram Transport Layer Security ) is a safe encryption key exchange protocol that needs encryption keys to be sent directly between peers on the media plane. Other encryption key exchange techniques are explicitly prohibited in WebRTC interactions because they are insufficiently secure.
Now the connection is established securely. The clients can communicate by the exchange of various Video and audio files. It also enables users to stream inbetween the call. To make this work it uses two set of protocols with different usability.
The Real-time Transport Protocol (RTP) is a network standard for transporting audio or video data that is geared for delivering live data in a consistent manner. It’s used for internet telephony, Voice over IP, and video telephony.
Voice over IP (VoIP) and Internet Protocol Television (IPTV), as well as streaming media and video conferencing, all employ RTCP. The statistical and control data is carried via RTCP, while the data is delivered by RTP. The number of bytes sent, packets sent, missed packets, and round trip duration between endpoints are all common RTCP statistics.
Applications using WebRTC
1. Google Hangouts, Google Meet, Google Duo
Since 2011, Google has been using Web Real-Time Communications and has developed multiple communication apps for personal and business use. These apps are: Google Hangouts, Google Meet, and Google Duo.
2. Facebook Messenger
WebRTC is used by Facebook’s mobile app and web client (which can be accessed via a web browser). Messenger has provided phone and video chats to its users via Web Real-Time Communications, and more recently, co-broadcasting via Facebook Live. WebRTC has also been included into Facebook’s VR Chat for video calls in Oculus, Workplace by Facebook, and IG Live Video Chat.
WhatsApp began as a simple messaging service and has since evolved into a worldwide messaging network that swiftly connects people from all over the world. WebRTC and SIP calling are heavily used in WhatsApp’s Android and iOS apps for fast and reliable virtual communication.
In their web client video conferencing, GoToMeeting had used numerous VoIP technology and WebRTC functionalities. The desktop client has been used by the majority of their clients and users (non-WebRTC). However, the easy-to-use web client’s growing popularity is attracting more consumers to use the browser tool.
Discord, which was originally created for the online gaming community, blends Web Real-Time Communications and VoIP to provide users with voice conversations and in-app chat. Discord’s engineering blog explains how they employed WebRTC to simultaneously serve over two million customers. They have more than 87 million registered members and 14 million daily active users.
There are thousands of third-party data-sharing servers available on the internet marketplace, but finding the right one for your OTT business can be tricky. If you are a small business organization and looking to launch your OTT platform on a shoestring budget, Muvi One is the best solution for you.
Muvi’s all-inclusive WebRTC integrated OTT platform development services are budget friendly. They allow you to stream high-quality video and audio with a minimum latency of 10 seconds. With adaptive multi-bitrate streaming, you offer a zero-lag streaming experience to your audiences.
To learn more about how MUVI One can help you start with your OTT content marketing strategy: Take a 14-day trial of Muvi One and explore the exclusive OTT platform features supported by Muvi.