In simple terms, WebRTC is a technology that provides peer-to-peer communication between web browsers and mobile applications. Known as Real-Time Web Communication, which refers to open source projects and allows for the transfer of audio, video, and data.
The top WebRTC development company claims that the technology is simple but sophisticated. The theme of simplicity comes down to ease of use. It is possible to use five to ten lines of code to organize peer-to-peer video connections between two browsers. At the same time, there is always a “but” where we work with technology.
In the case of WebRTC, the biggest challenge is the backend: developers have to make sure the solution works on different networks. For more information on the complexities of technology and how to overcome them, read our article, which also explains the conditions for using WebRTC and the emergence of technologies in the future.
What is WebRTC?
An open-source project released by Google in 2011, WebRTC provides API-based communication between web browsers and mobile applications, including audio, video, and data transfers. Eliminating the need for native plugins and installing apps makes this connection easy to use and supported by all major browsers and mobile applications.
WebRTC’s popularity in the technology community has grown significantly over the past few years. Facebook, Amazon, and Google are among the key technology companies that have used WebRTC to make their web applications faster, more reliable, and more secure.
A good example is OpenTok, PaaS for live communication, courtesy of former TokBox business partners (now Vonage). We have successfully implemented many solutions for our customers, including an improved authentication service based on biometric techniques.
As already mentioned in the summary, an important feature of WebRTC is that the technology is simple but sophisticated. The theme of simplicity comes down to ease of use. It is enough to use five to ten lines of code to organize peer-to-peer video connections between two browsers. The complexity of the technology is related to WebRTC specifications, which must be compatible with different browsers, and it is difficult to fix if it does not work properly. Also, to get the result you want, you should be aware of STUN, TURN, and NAT.
STUN is a set of standardized methods, including network protocol, for crossing the network address translator gateway (NAT) in real-time voice, video, messaging, and other interactive communications. Why do we need it?
How does WebRTC work?
WebRTC’s main focus is to provide real-time audio and video communication between participants, who use web browsers to initiate conversations, locate solitude, and pass fire shortcuts.
- Send and receive audio and video streams.
- Retrieve network configuration data, e.g., IP addresses, application ports, firewalls, and NAT (network address translators), required to send and receive data to another client using the WebRTC API
- Turn on / off the connection and report errors.
- Transfer media data, e.g., image editing and video codecs
- To send and receive data streams, WebRTC provides the following APIs that can be used in web applications:
- RTCPeerConnection for audio and video transmission, encryption, and bandwidth configuration
- RTCDataChannel standard data transfer
- MediaStream to access multimedia data streaming from devices such as digital cameras, webcams, microphones, or shared desktops
- A set of WebRTC application standards for software is currently being developed by the Internet Engineering Task Force and the Web Real-Time Communications Working Group.
WebRTC Under the Hood
WebRTC is primarily a way to send and receive UDP packages within browsers. Also, WebRTC development is aware of media transfers – both audio and video, and can connect two clients directly – peer-to-peer. The developers agree that under the hood, WebRTC is much simpler: open a UDP port, know the IP address of partners, bind traffic to RTP.
Let’s talk about what happens between a camera and a video game on the screen. This process consists of 7 basic steps:
1. Camera capture
The browser has an API that allows users to access the camera or microphone – navigator.getUserMedia => MediaStream. The main difficulty is that we cannot immediately send media streams to the interlocutor because it has a lot of weight without compression. For example, a single 640 × 480 image in BMP format weighs 1.2 Mb. The number of such images per second is 30 seconds. That means one second of video weighs 36 Mb. Therefore, the minimum rate will be 288 Mbps. Data must be compressed for transfer. So the next step – coding – is mandatory.
In simple terms, codecs allow for the compression of audio and video streaming. Let’s take VP9 as an example. It can transfer images with 1280 × 720 resolution and compress them so that 30 frames weigh 1.5 Mbps. How can VP9 do this?
Instead of constantly sending information about images, VP9 distinguishes between two images. We get a great deal out of the exit, while other independent frames represent differences in the mainframes. Additional actions in the frame mean more image weight.
If we lose a series of interframes at least once, we can’t draw some frames.
3. Packing on RTP
We need a number of packages to reproduce them properly. Also, RTP stores information about the time it allows for synchronizing audio and video tracks. Additional RTP details have a small overhead of about 5%.
There is an extension of a core protocol called RTCP. It works to exchange information about lost packages and recovery statistics.
4. UDP network transfer
Data is exported as a built-in UDP package. If we compare UDP to TCP, the biggest benefit will be the small interval between packages.
5. Disassemble the RTP
The package order is restored at this stage. Video traffic is received and forwarded to the screen.
The data is sent in the correct order, and upon its release, we receive a pure video stream – MediaStream.
7. Drawing on the screen
We attach the stream to the video section and get the picture.
During peer-to-peer interaction between two browsers, you will sometimes notice that the video is covered with squares or ice. Reason for loss of packages caused by various problems:
- Random Loss of Network Loss (in simple terms, part of the packages are left on the walls of the house).
- Network congestion.
In order to achieve stable video connectivity, we need to avoid package losses. Four main solutions help you to use it:
1. Jitter Bath
We offer one RTT later. We may request a package that does not exist. In the event of a major loss, the frieze is short because there is plenty of time to request a keyframe. The main drawback of such a method is a further continuous delay.
2. Reduce bitrate
Bitrate = FPS * quality * adjustment
We can cheat bitrate by changing any of these parameters.
3. Forward Error Repair
Codec repeats some data. This may increase network congestion, but we have a high chance of delivering content for the first time.
4. Network tuning
– Best network modes (we can design networks to make routes better, and the media server is selected according to the minimum ping value system).
– Setting up servers and routers.
Pros and Cons of WebRTC Technology
The main advantages of WebRTC are:
- Using modern audio and video codecs enhances high-quality communication.
- Protect and encrypt the DTLS connection with SRTP.
- There is a built-in way to capture content (desktop sharing).
- P2P = End-to-end encryption.
- Browsers agree directly.
- Versatility: a standard-based operating system works well on any OS as long as the browser supports WebRTC app development.
The conditional disadvantage of WebRTC is the high value of its storage, which is linked to the need for dynamic servers.
WebRTC Use Claims for Business Applications
As mentioned in the article, the basis for Web Real-Time Communication is video chat. Services with audio and video calls and data sharing are the main types of applications that include WebRTC technology, the most popular examples are WhatsApp, Google Hangouts, and Facebook Messenger. But if we put all the business cases and WebRTC models together, we can find that there are many areas to use.
MobiDev has repeatedly faced the need to use WebRTC in different areas. One of the most notable usage situations is remote assistance with shared AR and WebRTC. Used for peer-to-peer communication and helps prevent server overload. The essence of the case itself rises to the fact that two-way real-time communication with AR helps resolve tasks with assistance in many areas.
A simple example is the repair and maintenance of any machine.
The Future of WebRTC App Development: Trends and Predictions
Let us remember that in 2016 the global market value of products using WebRTC was $ 10.7 billion. WebRTC’s change came in 2017 when Microsoft Edge and iOS Safari 11 began to support it.
In terms of globalization, the WebRTC market covers North America, Europe, Asia, the Middle East, South America, and Africa.
Today, Google is making great strides in improving Real-time Web Communication. Therefore, the future of WebRTC app development may be uncertain. It’s easy to confirm this by checking Google’s investment in technology. They are all geared toward code development and enhancement or the development of a feature set.
The major trends related to WebRTC for 2021-2022 are:
- WebRTC, known as the W3C standard, will grow rapidly.
- The meeting sizes provided by WebRTC will increase, and that contributes to the complexity of the solutions. Significantly, the 1000 users at the conference is a real challenge that requires a new architecture.
- The pandemic caused their growth.
Moreover, as long as WebRTC development is a W3C standard, the top mobile app development company in USA contributes to WebRTC app development, which means good prospects.
If you would like to upgrade the WebRTC app not in the future, but in the meantime, rest assured that MobiDev will use any of your projects.