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Technology progresses at breakneck speeds, and new buzzwords and acronyms are thrown around with abandon. One of the hottest topics among software development companies has been WebRTC vs. WebSocket. Both may sound familiar, but are you aware of how they work and which is better?
In this article, we'll delve into the key differences between the terms and help you decide which is best for your needs. So buckle up and get ready to explore the world of real-time communication on the web!
WebSockets are a mighty tool in the world of web development: they make real-time communication between the client and the server smooth and painless. Unlike traditional one-way HTTP requests, they allow for a two-way data flow, making them incredibly practical for apps that require constant updates and interactions.
WebSockets also boast excellent support across many modern web browsers, making them a viable tool for web developers everywhere. Furthermore, with the WebSocket protocol being standardized by the IETF, it offers a stable and secure communication method that can be leveraged for various apps.
WebRTC is a free, open-source tool that enables real-time communication without add-ons like plugins or other software installations. It was first introduced by Google in 2011 and has since gained traction due to the seamless communication between people that it allows for through their web browsers.
Image source: Business Wire
What makes WebRTC so striking is its ability to support various communication types - audio, video, and data sharing. This technology is ideal for web conferencing, online gaming, remote education, and telehealth services. It can be integrated into existing web applications and work on multiple web browsers and mobile devices, making it more versatile.
Both WebRTC and WebSockets enable real-time communication, but they accomplish it in different ways.
WebRTC establishes peer-to-peer connections between browsers bypassing the server.
WebRTC Triangle by GeeksforGeeks
It is mainly used for video and audio chat applications, file sharing, and screen sharing. WebRTC can work on any device with a web browser and an internet connection. It uses the Transport Layer Security (TLS) protocol to ensure secure communication between peers. WebRTC can also be utilized in a merge with other web technologies such as HTML, CSS, and JavaScript.
WebSockets, however, are used for bi-directional communication between a client and a server.
The protocol provides a persistent connection between the client and the server, enabling real-time communication in web apps. WebSockets use a single socket connection between a client and server to transmit data in both directions. They are more efficient than other real-time technologies like AJAX, which use multiple HTTP requests to transmit data. WebSockets are used for applications like online gaming, stock trading, and chat applications.
Let’s go through the difference between these two technologies in detail.
Both client-side and server-side development are required for WebSocket implementation. Developers must take care of creating and maintaining connections, transferring data, and controlling events on both sides. The process can be simplified through libraries and frameworks with abstractions and other functionality they offer.
To implement WebRTC, developers need WebRTC APIs and protocols. Establishing peer connections, controlling media streams, and putting signaling into practice for session establishment is also necessary. The implementation of a signaling server or the use of third-party services is often required as server-side infrastructure for signaling.
WebRTC utilizes a direct P2P route between users' browsers, while WebSockets establishes a client-server route between a browser and a server. While both of these approaches have their respective edges, it's worth noting that WebRTC's P2P connectivity is generally considered to be more efficient and reliable. What’s more, WebRTC's encryption protocols make it a more secure option for communication that involves sensitive data.
WebRTC is designed for low-latency, peer-to-peer communication suitable for applications that require real-time interaction between users. WebSockets, on the other hand, are better fitted for apps that need real-time communication between a client and server (such as updating live data or pushing notifications).
WebSocket can be scaled using strategies like load balancing and horizontal scaling of WebSocket servers. WebSocket apps can manage more concurrent connections by splitting the connections among several servers. However, expanding WebSocket to support large user populations could necessitate further thought and infrastructure planning.
With WebRTC, however, peer-to-peer scaling is possible without taxing the server infrastructure. WebRTC lessens reliance on centralized servers for media transmission by creating direct peer connections. However, additional infrastructure and load-balancing strategies become necessary to provide scalability when dealing with a high number of peers or when additional server-side functionality is required (for example, signaling and media processing).
One of the critical advantages of WebRTC is its built-in security features. WebRTC employs Datagram Transport Layer Security (DTLS) and Secure Real-Time Transport Protocol (SRTP) to encrypt voice and video communication. This ensures that only the communicating parties can access the data transmitted over the WebRTC connection. Additionally, WebRTC incorporates various security mechanisms, such as ICE (Interactive Connectivity Establishment) and STUN (Session Traversal Utilities for NAT), to protect against network attacks.
WebRTC protocol stack looks as follows:
Unlike WebRTC, WebSockets does not provide any built-in security features. This means developers must implement their own security tools to secure the data transmitted over the WebSockets connection. This is possible with the help of HTTPS, which provides TLS/SSL encryption. It can lead to additional overhead and latency, though. Alternatives include third-party security libraries like JSON Web Tokens (JWT) or OAuth to add security to WebSockets.
WebSocket is ideal for applications requiring real-time messaging, data streaming, gaming, and push notifications. It is widely used in chat applications, social media platforms, live sports scores, and real-time analytics dashboards, among others. It delivers a low-latency, bidirectional communication channel that allows for rapidly exchanging messages between servers and clients.
WebRTC is highly suitable for video conferencing, voice calling, live streaming, and remote assistance and support applications. It enables real-time audio and video communication between participants and facilitates remote collaboration, online education, and customer support chat. It also delivers a high-quality multimedia experience with low latency, enabling live events, webinars, and broadcasting platforms.
Applications that require real-time communication between IoT devices or streaming sensor data can also make use of WebRTC. Apps like home automation, remote monitoring, and industrial IoT are made possible by the efficient interchange of real-time data from linked devices made possible by this technology.
While both technologies come with their merits, they’re not devoid of drawbacks. Let’s explore both the advantages and drawbacks of using WebRTC and WebSockets.
WebRTC pros:
Cons:
Like any technology, WebSockets have their own set of pros and cons.
WebSockets pros:
Cons:
Which one is better - WebRTC or WebSocket? This question appears on the Internetverse quite often. Well, there is no clear winner between the two because various factors, like the hardware configuration and the number of concurrent users, affect how well WebRTC and WebSockets work. The comparison may be fruitless because WebSocket use cases and WebRTC use cases are distinct from one another.
Mainly when dealing with multimedia content and big user groups, WebRTC can be very CPU-demanding. As a result, developing and scaling WebRTC apps is expensive and challenging. Similarly, creating a WebSocket solution you can rely on to work at scale has several hurdles.
We know that selecting the proper protocol for your communication needs is crucial to ensuring successful audience interaction. In this regard, we've put together several tips to help you determine which protocol is best suited for your requirements.
Certainly, the choice between WebSocket and WebRTC primarily depends on your project-specific needs. So, assess each factor carefully before making a decision. Whatever choice you make, remember that your real-time ability to interact with your audience can significantly benefit your business and provide a better UX.
If you're wondering whether to use WebRTC and WebSocket together, the answer depends on your use case. By combining WebRTC and WebSocket, you can create high-quality, real-time, secure, scalable, and interoperable applications across different devices and platforms. Whether you're building an enterprise app, a social network, or an e-learning platform, WebRTC, and WebSocket offer a powerful combination that can meet your needs.
So, we already know that WebRTC is ideal for real-time, peer-to-peer communication, such as video calling or file sharing, without plugins or app downloads. It's handy for applications that require low latency, high-quality streaming, and end-to-end encryption. However, while WebRTC allows direct communication between peers, it still requires a signaling server to establish the connection. This is where WebSocket comes in, providing a reliable, bidirectional communication channel between the client and server.
WebSocket can be a signaling mechanism for WebRTC, enabling peers to exchange metadata and control messages in real time, such as codecs, media types, IP addresses, and port numbers. This is particularly useful for applications that involve multiple peers or require server intervention for signalings, such as group video conferencing or online gaming.
WebRTC and WebSockets differ in their functionality and intended use cases. While WebSockets are primarily used to establish a bi-directional communication channel between a client and server, WebRTC enables peer-to-peer communication between two web applications without passing through a central server. Moreover, WebRTC offers additional features such as audio and video streaming, screen sharing, and file transfer capabilities, making it a more comprehensive solution for real-time web communication needs.
That being said, WebRTC cannot entirely replace WebSockets, as they serve different purposes and are not mutually exclusive. Depending on the desired functionality and use case, either technology can be used to achieve efficient and steadfast real-time data streaming and communication between web applications. WebSockets remain the go-to solution for server-initiated communication and messaging applications requiring real-time bi-directional data exchange between clients and servers.
Both WebRTC and WebSockets are crucial technologies for real-time communication on the web. They complement each other rather than exclude. By leveraging the strengths of each technology, you can create seamless, immersive experiences that engage your users and drive your business forward.
Have any questions left? We’ll be happy to answer them. If you are creating your own solution and feel like you need help with it, don’t hesitate to contact us. We believe that software development should be easy, which is something we continue to prove on a daily basis.
