Since its evolution in 2011, the WebRTC protocol, or name it technology, is in full swing aiding millions of developers to build WebRTC apps and web applications flawlessly and seamlessly. Though the tech was a bit hidden behind the bushes, it came to rise post the pandemic when video calling and conferencing were in great demand for rendering low-latency real-time chats. The protocol does this work in a well-fashioned way by integrating into any browser without the need for standalone apps.
As per a recent report, the demand of WebRTC is predicted to rise to $16,570 by 2026 because right from the development of WebRTC telehealth apps to WebRTC app development to android to WebRTC iOS apps, every VoIP stone is turned with this protocol.
Therefore, for all knowledge-seekers out there wanting to grasp the benefits of this technology and web service examples, this blog is the right spot.
Let’s Start With WebRTC In A Quick Glance
Future Trends in WebRTC
- They evolve and develop rapidly
- Connecting users using this protocol will double in size
- Noise suppression and blurs are ought to be removed in the latest version of WebRTC
- Tendency to modernize codec VP9 and AV1
With WebRTC acting as a trendsetter in the communication domain, our next focus has to be on,
What is the Working Functionality of WebRTC?
Here is how the WebRTC Security Architecture Works:
- A user sends text messages or audio or video streams to the recipient where a new connection is established.
- While the data is transmitted, the WebRTC API retrieves network configuration data like IP addresses, firewalls, NATs, and application ports to send it to another client.
- Upon successful transmission of data, all the connections are closed and errors if exist are notified.
- As a next step, metadata like video codecs, device info, network info, and other likewise info is also transmitted to the clients.
From what we discussed, we can draw a point that WebRTC APIs are the backbone for establishing any real-time communication. Well, with this API, the protocol makes use of other APIs in web applications for connection, communication, and data transfer. Below are those APIs.
- RTCPeerConnection – for audio and video transmission
- RTCDataChannel – for transmission of data
- MediaStream – for granting access to devices like digital cameras, webcams, shared desktops, and microphones.
Knowledge Fact: Get a full insight on video calling APIs for launching features through this post.
Next up under the WebRTC cloak are its benefits.
What are the Benefits of the WebRTC App?
Due to bags of easiness, reliability, and affordability level, most of the IT giants like Amazon, Google, FB, and WhatsApp are using this technology to develop WebRTC native apps. Their exponential growth is an example of quickly, securely, and seamlessly building real-time chat system on all consumer platforms.
Coming in full compatibility with the latest devices and browsers like iOS, Android, Chrome, Safari, BackBerry, Vivaldi, Brave, Firefox, and Edge, the protocol gave routes to multiple IT industries, financial services, the healthcare sector, and telecom units. Other benefits of WebRTC reference app include,
- The SRTP protocol used in WebRTC to encrypt voice and video calls has given way to endless security.
- Supports streaming content from AR and VR-enabled devices too.
Post saying these, you might have the pulse on finding out how to make a WebRTC app or build a WebRTC app. Check this article posted by our expert for a full-fledged understanding of the topic.
Use Cases Of WebRTC Application
As of the calendar year 2022, WebRTC is used in thousands of solutions because of the die-hard convenience and sustainability reviewed and understood by tech giants of this era. Some of the notable WebRTC web and app development examples are:
Who doesn’t know Discord as an online gaming community app? But, not many know it as the earliest VoIP application that was started as a means to establish communication. But, seeing its growth in the field of gaming, the enterprise had to bring in technologies like WebRTC to add group video calls and HD-quality audio chats to the application so that users can seamlessly connect with their fellow gamers over an event.
- Amazon’s Chime
Initially developed as a means for internal communication, Chime- a video conferencing hub later was made available to customers. This is the prime solution or product of Amazon that uses WebRTC technology. Another broad usage of this protocol can be seen in Amazon Kinesis Video Streams which helps in streaming content to AWS cloud and Alexa assistant for integrating to smart home cameras and doorbells.
- Google Meet
The mega bond tech giant, Google Meet is the newest version of them that is into offering buffer-free video streams. Earlier, Google had an in-built video conferencing solution in Gmail but since it required the use of a plugin and live videos were not available in the browser, they migrated to using a WebRTC-enabled solution – Google Meet.
We are halfway through this article, and till now, we saw how to easily create WebRTC apps and their use cases across different domains. And, much similar to what we discussed before, our next topic is developing web and hybrid apps using WebRTC.
WebRTC App Development For All Web & Hybrid Apps
WebRTC has been the center of magnetism for businesses of all sizes because of its attractive features like free and open source code, the ability to work on any user-friendly interfaces, HD quality audio and video calls, great server-side support, and bulk savings on telephony.
Amongst the different features that we saw, it is the free source code that is drawing the attention of many, as having that in hand, developers can easily create multimedia support for streamlining presentations, conducting meetings, and streaming secure and buffer-free video content. Now this mega benefit of being an open source can indeed become a little knotty as it requires a considerable amount of effort for developing both the front-end and back-end sides of the application.
Hence, it is required to have Android and iOS developers in hand depending on the platform you are building the application. Now, you might think, WebRTC does not require client-side programming, so what is the need for front-end development? Well, you would still need tools and other dependencies like
- WebRTC Libraries
- CSS Framework
- jQueries for adding elements
- Nodejs Server
to create a WebRTC mobile app. Once you have all these libraries in hand, make sure to install third-party tools for supporting WebRTC mobile solutions. They are;
- Telefonica Tokbox – A video calling API plug-in
- MirrorFly – WebRTC SDKs and APIs for mobile apps
- Hookflash – An additional support for web and mobile app development.
App Development For WebRTC Native Apps
Initially, WebRTCs don’t cater their services to native app development for multiple systems because of its complication than creating a web service. However, as time passed by, people started to recognize its benefits even though WebRTC native apps require a considerable amount of knowledge, time, and resources. You would not need a native app for creating basic audio or video chats. Rather, their use cases are spread across domains like
- Audio and video conference calling options
- Creating sensitive applications like delay-sensitive, video, and sound quality sensitive
- Embedding multiple video streams, encoding, and deciding video contents
These may seem to be tough, but with the help of WebRTC APIs, you will have complete control over the native app.
- Build device and platform-independent WebRTC apps
- Deliver smoother and faster video calling experiences
- Integrate WebRTC technology into any existing application
- Eliminate any browser incompatibilities, and
- Avail full access to all OS and hardware resources
To wrap up this topic, creating a WebRTC native app has the same difficulty as that of any web application. Nevertheless, building a native app using WebRTC can become easy if you have checked on the following common mistakes.
Most Common Mistakes Developers Make
Here is a list of some of the common mistakes developers make while building a WebRTC Android app or iOS despite its ease of use and implementation.
- Hire a dedicated tester or test updates often
Though WebRTC is compatible across different platforms and devices, when you make an update, it is required that you test the updated code on multiple browsers using different roles. This part is required because a media stream layer of WebRTC may throw an error in certain browsers that may change the roles, video codecs, and other settings.
- Get to know browser updates
We are well aware of the fact that WebRTC is constantly developing and improving very often. Therefore, it is required to keep an eye on the browser updates, releases, and changelogs, and test frequently to find out bugs that could hamper the development process.
- Do not miss to test calling functionality
The calling functionality is very important in the implementation of video calls to web and mobile devices. If you do a test call on a single browser, only the caller will have access to the camera or microphone of that particular browser. So, when you do this step, make sure to test the whole application using the Signaling Layer or VPN.
The use of WebRTC is not going to stop any time soon because of the continuing pandemic and the growth of several conferencing options, their usage will only spike in the mere future. Therefore, the only option left in your pocket is to use a dedicated platform like MirroFly to handle all multimedia needs, stream content with low latency, and ensure zero intrusions on data.
If you agree with us and are ready to create apps that use WebRTC, drop us a line, and we will help in building a customizable video calling app using WebRTC APIs.
Frequently Asked Questions (FAQ)
What Language Does WebRTC Use?
WebRTC (web real-time communications) itself does not have any specific programming language as it is a set of APIs (application programming interfaces) that can be utilized by multiple programming languages.
Is the WebRTC API Free?
Yes, the WebRTC API is free to use. WebRTC is a real-time communication API available to developers without any cost. It is an open-source project supported by major web browsers such as Google, Apple, and Firefox.
But bandwidth charges from your network provider are applicable for using WebRTC to transmit audio or video streams.
Can WebRTC Work Without a Server?
A signaling server is required to establish and coordinate communication between peers using WebRTC. Which means without a server, WebRTC won’t work. WebRTC can facilitate direct peer-to-peer communication after initial setup, but it still requires a signaling server to initiate and coordinate the communication among peers.
What Encryption is Used in WebRTC?
Media encryption and signaling encryption are the two types of encryption used in WebRTC. WebRTC uses encryption mechanisms to ensure the safety and privacy of transmitted data. These encryption mechanisms generally ensure the security and privacy of WebRTC communication.
What is the Minimum Bandwidth for WebRTC?
WebRTC is developed to adapt to different network conditions and can work with limited bandwidth. Depending on specific use cases and the desired quality of audio and video communication, there might be a minimum bandwidth of 100 kbps or less to 8 mbps or higher, depending on the quality.
Is WebRTC VoIP?
Yes, WebRTC is primarily used for voice over Internet protocol (VoIP) applications. VoIP refers to the technology that provides voice communication using the internet rather than traditional phone communications. WebRTC provides the necessary protocols and APIs that can be used inside web browsers, making it viable for VoIP.