The evolution of technology has simplified the communication and one important breakthrough is the real-time communication. In these times of pandemics where remote workflow is the new normal, the communication is of utmost importance for the coordination within teams and with prospects.
WebRTC is an acronym for Web Real-Time Communication. Since its adoption, the technology took communication to new heights. WebRTC app development is a solution to achieve real-time communication in the web browser and it was initiated by Google and Mozilla as an open protocol to channelize the open communication between two users.
These comprise of data streams, STUN/TURN servers, signaling, JSEP, ICE, NAT, UDP/TCP, SDP, SIP, network sockets, and more. These APIs are plugin-free and don’t require any tedious installations and downloads. WebRTC can prove to be beneficial for businesses for video-conferencing, click-to-call, peer-to-peer streaming, and instant messaging.
In WebRTC, communication takes place between browsers. It employs three HTML5 APIs that allow users browsers to capture, encode, and transmit live streams between one another. Therefore, it enables two-way communication. And due to this, WebRTC is known as peer-to-peer technology.
During these message exchanges, there is no need for any intermediary web servers, additional equipment, or software. The URL-based meeting rooms are the top-notch example of the convenience and real-time communication delivered by WebRTC.
Why Getting on the WebRTC Bandwagon Important?
To be precise; real-time communication is not a new concept, there are established proprietary closed-source services like Skype and Apple’s Face Time.
We are on the verge of the combination of the trio; namely voice, video, and data for unified, open-standards communication within the browser.
WebRTC… Working and Technical Dimensions
As we have discussed that WebRTC leverages multiple standards and protocols, so here we will discuss various technical aspects and working of this emerging technology.
Exploring the Peer-to-Peer Communication Channel
- The Primary benefit of WebRTC is peer-to-peer audio and video (i.e., multimedia) communications. This communication with another person (i.e., peer) is via a web browser, so here each person’s web browser must agree to begin communication.
- One of the challenges with browser-based peer-to-peer communication is to gather information about how to locate and establish a network socket connection concerning another computer’s web browser for transmitting the media.
- So, here you make the HTTP request to a server that is known and easily locatable (via DNS) and that reverts a response (i.e., the web page).
Image Credit: Sam Dutton (HTML5 Rocks)
Exploring the Firewalls and NAT Traversal
- The internet accessed by us is not assigned a static public IP address. Our system sits behind the firewall and network access translation device (or NAT).
- In a broader sense, the NAT device translates private IP addresses from inside a firewall to public-facing IP addresses.
- Now if you are thinking about how would you know another person’s address to send audio and video data to, or how would he know what IP address to send the audio and video back to, this is where the role of STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relays around NAT) servers come.
Image Credit: Sam Dutton (HTML5 Rocks)
- So, for the working of WebRTC, your public-facing IP address requests the STUN server. The server then responds with the IP address.
- Now, suppose this process works smoothly and you get your public-facing IP address and port, then from these, you can tell other peers how to contact you directly.
Exploring the Signaling, Sessions, and Protocols
- In the broader sense signaling involves communication security, session creation and management, error handling, media capability, metadata and coordination, network discovery, and NAT traversal.
- Now, if the WebRTC browser-based application has figured out its public-facing IP address using the STUN server, the next step would be to establish the network session connection with your peer.
- Initially, this session establishment takes place using signaling/communication protocols that are especially for multimedia communication.
- Any WebRTC leveraging application (or peer) that attempts to communicate with another peer generates a set of ICE candidates; here ICE is an acronym for Interactive Connectivity Establishment protocol.
- In the case of ICE, candidates represent a combination of IP address, transport protocol, and port to be used.
WebRTC Application Development… USE CASES
The use cases of WebRTC development are spread in diverse verticals; hence we would restrict our discussion to the healthcare segment.
WebRTC in the Healthcare Sector
The telehealth service or telemedicine app development service market has emerged as a major sector for WebRTC. The healthcare providers are attracted to it because of the data encryption provided that ensures the safety of the personal health data of patients. At StarTele Logic, we strictly adhere to the HIPAA protocols to safeguard the data being exchanged.
The telemedicine app development services and solutions provided by us encapsulate real-time video feeds, appointments with the doctor, clinical meetings, teleconferences, and remote observation of operating rooms.
WebRTC and… Trillup, Telemo
At StarTele Logic, WebRTC based application development makes an organization work flawlessly. It is encapsulated with voice, text chat, seamless video streaming, and much more off-center features.
WebRTC is a basic building block for Trillup, and Telemo’s audio video chat, or conferencing calling features using its peer to peer technology. To bring-in the real-time communication to life, we have integrated both Trillup and Telemo on WebRTC audio chat, conference, and video streaming technology.
This WebRTC development enabled real-time video streaming, audio embedding has made Trillup and Telemo a full-fledged application with soaring advantages and benefits.
Some of the robust hallmarks of Trillup and Telemo are:
- Text messaging in the Real-Time environment: It allows you to exchange messages in the real-time frame.
- Voice and Video calling and conferences: Prospects can use these features to have a face-to-face conversation.
- One-on-One Chat: This feature can be used as private meeting rooms for various users.
- Group Chat: For people who want to form a special interest group can use this feature. Users in the group can either make it open for all or make them invitation-only groups.
- File-Sharing: It allows users to exchange files on the go!
- Collaboration: It also encapsulates collaborative whiteboard and collaborative documents respectively. The whiteboard allows users to put their ideas on ii instantly! On the other hand collaborative document allows multiple stakeholders to carry out work on a single document in real-time.
- Multi-language Support: To enhance their communication, users can interact in the language of their preference; they can modify the UI of Trillup and Telemo as per their convenience.
Read also: “Telemo: Welcome to a Virtual World for F2F Meetings“
Why Collaborate With Us to Build Your Own WebRTC Based App Development?
As there exist many applications that provide real-time communication, but building an application based out of WebRTC development requires expertise as well as hands-on experience over various tech spheres.
- With us, you don’t need to build from level 0!
- Provide swift integration
- Optimizes Development Cost and Time
- Next-Gen Collaborative Features
Whether you are a budding startup or a developer, you surely do not need to work from scratch to build a WebRTC voice/video calling chat app. This will save time and effort at your end, otherwise, you have to go through all the stages of the software development life cycle model.
The applications developed by our experienced developers won’t take much time to get integrated with your app or website. Collaborating Trillup or Telemo with your existing application or website would make the process run faster.
Well, going with Trillup and Telemo, instead of developing a new application will save a lot on time and cost budget. The integration will lead to the development of the project with the latest technologies. You would surely reduce project overheads and time if you choose to integrate Trillup or Telemo with your application.
We at StarTele Logic provide you with wholesome of great features that would not only prove to be efficient at present but also a milestone in the longer run.
WebRTC… Key Takeaway
To summarize, we can say that WebRTC app development has become a real-life synonym of on-the-fly communication. It provides users the facility to exchange voice and video chats over the P2P network using web browsers.
For those out there looking for high-definition digital communication, mobile-to-mobile communication, messaging and file sharing, smartphones-to-browser connection, WebRTC has come out as an ultimate solution to meet all your needs!
StarTele Logic leverages these upscale technologies to deliver reliable and top-notch cloud telephony solutions and services to its global clients. Our developers hold commendable experience in not only developing WebRTC but also building quality native as well as cross-platform apps development for iOS and Android devices.
Sounds Amazing! Get in touch with us to know more about our services, and please enlighten yourselves by shooting your queries in our mailbox!