Digital evolution and transformation are unavoidable in the current age and time. Now that covid-19 has changed the way many things work it has made digital inclusion a necessity. Reduced physical contact called for immersive video calling experiences and its convenience is here to stay; from e-learning, business video conferencing to live streaming and real-time gaming applications the requisites of communication have greatly scaled.
To make communication work better and efficiently, any video application should have low latency and reliability, that is when WebRTC comes into the picture.
WebRTC is an open-source framework that enables mobile applications and web browsers with real-time communication via simple APIs; hence the name WebRTC – Web Real Time Communication. With the help of real-time protocols, WebRTC facilitates peer-to-peer real-time communication, media & application data transfer between web browsers to stream live video, audio and data streams over a network. These sets of protocols or specifications ensure encrypted communication and deal with a bunch of issues like data loss, connection dropping, delays/buffers, bandwidth incompatibility, noise, etc.
Most modern desktop and mobile browsers such as Google Chrome, Mozilla Firefox, Safari and Opera have already implemented this technology natively; even otherwise it is an easy to set up open-source protocol.
Why do we need WebRTC?
Because, WebRTC requires no additional plugins or native apps and establishes connection between peers with no servers in the middle. All you need is a WebRTC compatible browser/ application. This puts zero burden at the user end, making them enjoy seamless video conferencing, screen/ file sharing with just their browser or app.
Some of the highlights of this technology:
- Secure peer to peer connection
- Bypass paying for any bandwidth between the peers
- High performance & low latency
- No servers or infrastructure required
- Officially standardized and constantly evolving
- And, it is completely free
WebRTC consists of several interrelated APIs and protocols which work together to achieve Real Time Communication. Here are some of the important terms to understand.
A set of initial communication setup has to be initiated, before establishing a connection between two browsers. The mechanism that takes care of this information exchange is called Signaling. To set up, control and terminate a connection, three types of information must be exchanged:
- Session control information – manages when to initiate, end and modify sessions. Also used in error reporting
- Network data – determines & reveals IP location of endpoints to connect callers and callees
- Media data – determines codecs and media types that are common between the peers
This set of information also called metadata is extremely important for a successful communication establishment; Signaling mechanism completely takes care of this metadata exchange between peers to transfer media, audio and video data.
A STUN (Session Traversal Utilities for NAT) server allows Network Address Translator (NAT) clients to set up phone calls to a VoIP provider hosted outside of the local network. The STUN server allows clients to find out their public address, the type of NAT they are behind, and the Internet side port associated by the NAT with a particular local port. This information is used to set up UDP (User data protocol) communication between the client and the VoIP provider to establish a call.
A TURN (Traversal Using Relay around NAT) server is a media relay/proxy that allows peers to exchange UDP or TCP media traffic whenever one or both parties are behind NAT. When you want to make a call between different networks or when NAT won’t allow direct access to a host, WebRTC uses the TURN server. WebRTC protocol automatically decides when to use STUN or TURN servers.
How does it work?
WebRTC can send real-time audio, video, or data directly across browsers, thus it is referred to as a P2P technology. How exactly the communication takes place can be explained with the following concepts.
WebRTC’s RTCPeerConnection API is used to stream media from one browser to another. This is executed by identifying the correct IP address and port number. The API’s signaling mechanism, after detecting the location of both the ends, begins transmission of multimedia in real-time.
Firewalls and NAT Traversal
Each device has a public and a private IP address with a firewall protecting it. NAT (Network Access Translation) device translates private IP addresses from inside a firewall to public-facing IP addresses. Now the STUN and TURN servers help in detecting the IP information and facilitate the connection.
Signaling, Sessions, and Protocols
Signaling involves detection of networks and NAT traversal, session creation & management, security, metadata transmission and error handling. SIP (Session Initiation Protocol) and SDP (Session Description Protocols) take care of the signaling mechanism of passing digital information back and forth.
Finally, a two-way communication is established and data is transmitted with the help of ICE (Interactive Connectivity Establishment) protocol.
To acquire and communicate streaming data, WebRTC implements the following APIs:
- Media stream API helps in getting access to data streams, by allowing you to access the input devices at both ends like camera and microphone.
- RTC peer connection API enables audio or video calling in real-time with facilities for encryption and bandwidth management. It helps to initiate a connection with your remote peer, manage & control it and securely close the connection.
- RTC data channel API enables peer-to-peer communication of generic data.
WebRTC’s security protocols and encryptions help in maintaining a reliable & protected real-time communication. Secure protocols like DTLS & SRTP, mandated encryption and permission-based access to media devices ensure protecting the digital communication from vulnerabilities like data theft, malware and uninformed video distribution.
With the current boom in technology and the need to be digitally available, the APIs and standards of WebRTC are extremely accessible and resourceful. They offer tools and services that make real-time communication convenient across widespread streams.
Driven by innovation and focused on uniqueness, Innoinstant is the best choice as your video calling solution provider. The right product is achieved with the right start. Innoinstant is here to take you towards the completion of a great product. Our customizable approach helps in creating a rich and engaging real-time video calling & conferencing application for a greater business reach.