WebRTC vs WebSockets: What Are the Differences?

From gaming to messaging and video conferencing, real-time communication is at the forefront of most apps.

WebRTC and WebSockets are among the standout real-time technologies today. They enable instant, two-way information exchange to facilitate online interactions and make communication more efficient.

While they both offer powerful communication features, they have important differences. By understanding their strengths and limitations, app developers can tailor solutions to specific project demands and performance goals.

What Is WebRTC?

WebRTC is an open-source protocol for voice, video, and data sharing. It lets developers build robust peer-to-peer communication solutions for the web and mobile applications without worrying about compatibility. It's popular in video streaming, real-time data sharing, and interactive applications.

WebRTC's main components include:

RTCPeerConnection: The RTCPeerConnection is at the core of WebRTC. It establishes peer-to-peer connections between devices and facilitates data exchange without intermediary servers.

MediaStream: The MediaStream captures input from devices like cameras and microphones to record, resize, and display video and audio content. With the getUserMedia() function, users can specify the type of media they want.

Data Channel: The RTCDataChannel is a data exchange component that allows bidirectional transfer using the Stream Control Transmission Protocol (SCTP) to provide reliable file transfer with low latency and direct data sharing.

How Does WebRTC Work?

WebRTC uses JavaScript APIs and HTML to embed real-time communication technologies on web browsers. It combines several components to exchange media capabilities and network information:

• Session Description Protocol (SDP) to characterize sessions and capabilities.
• ICE (Interactive Connectivity Establishment) to find the best peer connection path.
• Real-time Transport (RTP) and RTP Control (RTCP) to manage transmission and stream quality.
• Stream Control Transmission (SCTP) Protocol for managing data channels.

The transfer process begins by accessing the media data, with the RTCPeerConnection establishing a peer-to-peer (P2P) connection between devices and browsers. You can also add media channels (e.g., MediaStream) and data channels to the open connection.

When users aren't on the same IP network or Network Address Translation (NAT) firewalls prevent peer connections, WebRTC is used alongside Session Traversal Uti+lities for NAT (STUN) servers to enable IP access.

When the peer-to-peer connection is complete, the peers communicate directly and transmit streams. First, the media files are compressed to reduce their sizes and latency. After transmission, they are decompressed using a codec. Data exchange in WebRTC occurs in real time without any extra plugins or configurations.

Pros 

The pros of using WebRTC are:

• Data Encryption for Secure Transmission: Every software component in WebRTC is encrypted to secure data during transmission. Besides, the WebRTC API requires secure data origins via HTTPS.
• Multi-Device Compatibility: WebRTC works on all major operating systems without additional configurations. It doesn't require third-party components as long as the browser supports WebRTC.
• Ease of Integration: The WebRTC API integrates seamlessly with the native browser APIs to simplify integration.
• Compatibility: Web browsers like Google Chrome, Safari, Mozilla Firefox, etc, support WebRTC, ensuring compatibility and accessibility.

Cons  

Some of the downsides of WebRTC include:

• Bitrate Adjustment: WebRTC has an automatic bandwidth adjustment feature that can downgrade audio or video quality.
• Signaling: WebRTC lacks a standard signaling protocol, which requires developers to implement custom solutions such as session management. This process can be complex and costly, adding to development overhead.
• Implementation: To implement WebRTC, you need a compatible browser and a signaling server to create peer connections. You also need a TURN/STUN server to bypass network limitations and a client-side code to handle the APIs for media capture.

What Is WebSockets?

WebSockets is a bidirectional communication technology that provides a persistent client-server connection for instant data exchange, allowing developers to build real-time applications with minimal overhead.

WebSockets has two main building blocks:

• The WebSocket protocol provides real-time client-server communication.
• The WebSockets API is an interface for creating two-way communication between the client and server in a web app. It allows developers to integrate the WebSocket protocol into their applications easily.

WebSockets' main components include:

TCP Connection: WebSockets operates on a single, persistent TCP connection. It stays alive after the client establishes a connection for bidirectional communication.

Protocol: WebSockets uses a specific data exchange structure defined by the WebSocket protocol. This includes the handshake process, headers involved, and procedures for closing connections. Full-duplex communication means messages can be composed of one or more frames. A frame is a binary-formatted unit that contains metadata and payload data.

Prefixes: Prefixes in WebSockets indicate the protocol and security status. 

• wss:// indicates a secure connection. Transmitted data is encrypted using TLS (Transport Layer Security).
• ws:// indicates an unsecured connection. The server doesn't provide encryption and requires a valid SSL/TSL certificate for secure transmission.

Masking: Frames sent from the client to the server are usually masked using the 32-bit masking key value to prevent cache poisoning attacks during transit. The frames are unmasked before processing. Frames sent from the server to the client should also be unmasked.

How Does WebSockets Work?

Communication starts when the client connects to the server through a WebSocket handshake. With the connection on, the client can send messages over the persistent connection without additional HTTP requests.

WebSockets is a stateful protocol, meaning the server and client remain active until one party terminates the connection. If either party terminates the connection, a new one must be established.

Pros  

WebSockets' benefits in real-time communication include:

• Full-Duplex Communication: WebSockets provides full-duplex communication, which means data transmission can occur between servers and clients simultaneously. HTTP, in contrast, only allows for one-way communication.
• Low Latency: WebSockets establishes a persistent connection, which minimizes latency. It allows users to send messages with minimal overhead, which makes it ideal for live chat and real-time gaming.
• Efficiency: The WebSocket protocol maintains a single connection and doesn't need to send headers to establish a new connection for every interaction, reducing the overhead that comes with frequent HTTP calls.
• Reduced Bandwidth and Server Load: WebSockets uses less bandwidth, which reduces loads on both the client and server sides.

Cons 

Here are some of WebSockets' downsides:

• Compatibility: Although WebSockets works on modern browsers, older ones may not fully support the protocol.
• Memory Consumption: Running numerous connections leads to high consumption, as each one consumes RAM. This high resource consumption requires proper management to avoid causing communication bottlenecks.
• Server Connections and Proxy Limits: Network configurations such as proxy servers and firewalls can block WebSocket connections. In this case, you must consider supporting fallback transports, which adds extra complexity.

WebSockets vs WebRTC: The Differences

While both WebRTC and WebSockets support real-time communication, they serve different purposes and have several key differences.

When To Use WebRTC vs. WebSockets 

WebRTC is essential in apps that need secure, high-quality, low-latency peer-to-peer connections, such as video, audio, and data streaming.

Live Streaming: WebRTC is used in live streaming applications such as video conferencing, live event broadcasts, webinars, and online gaming. It delivers high-quality audio and video with minimal delay, making it ideal for virtual events and platforms with live audiences.

Data Transfers: WebRTC excels in data transfer apps as it can facilitate sharing large files with minimal overhead. It's efficient for remote desktop access and file sharing, where large files must be transmitted instantly. It's also useful for distributed systems, online gaming, and collaborative tools.

Privacy: Since WebRTC provides encrypted connections, they are ideal for apps where data privacy is paramount, such as secure communications platforms, financial services, telehealth, etc.

When To Use WebSockets vs WebRTC 

The WebSocket protocol is ideal for instant data exchange in full-duplex communications. Situations where you might use WebSockets include:

Real-Time Chat Applications: WebSockets is excellent for live chat systems, like social media platforms, gaming chat apps, and customer success chatbots. It maintains persistent connections without the delays of HTTP pooling for seamless interactions.

Team Collaboration: It's also ideal for project management and field operations apps where users need to communicate in real time. Synchronization ensures changes to project status, whiteboards, and documents appear instantly for more productive, collaborative experiences.

Live Notifications and Updates: Many app features that require live updates or notifications use WebSockets, like social media activity feeds, sports wagering, and trading platforms. Since WebSockets can deliver updates with low latency, users receive information promptly.

When To Combine WebRTC and WebSockets

WebRTC and WebSockets complement each other and can be combined in applications requiring signaling and coordination. You can configure WebRTC for peer-to-peer transmission and WebSockets for real-time message exchange.

Supporting real-time online communication between users requires a server for communication coordination. Signaling is the process of transferring information between peers. This includes network data, session control messages, and media metadata.

WebRTC lacks signaling features, which is where the WebSocket protocol comes in as an efficient signaling mechanism. In this scenario, WebRTC manages media and data streams between users while WebSockets handles signaling and data synchronization.

Common Use Cases and Applications

Many modern applications use real-time communication protocols to enhance connectivity and user experiences. 

WebRTC Use Cases

Gaming and Esports: WebRTC supports sports broadcasting and live streaming with low latency, enabling players to collaborate on live gaming events.

Telehealth: WebRTC provides end-to-end encrypted communication in telemedicine apps, ensuring data is securely transmitted between medical practitioners and patients. Telehealth apps can use WebRTC for virtual consultations, screen sharing, audio communication, and session recording.

It's also used with remote monitoring devices to transmit real-time patient data to healthcare practitioners.

Online Education: WebRTC has many practical applications in remote learning. It's used to build virtual classrooms with built-in video and audio streaming, chat, and whiteboards. These features facilitate immersive online learning environments, one-on-one sessions, collaborative learning activities, and remote assessments.

Live Events: WebRTC is helpful on platforms that combine shopping with video power. Brick-and-mortar businesses can host live streams to showcase products and attract potential customers. Viewers can share opinions about products, connect with brands and retailers, and purchase products directly from the video streams.

Corporate Communications: Most video conferencing apps, webinars, and virtual meetings use WebRTC. This lets remote teams share information and collaborate in real time.

WebSockets Use Cases

Live Chat: The WebSocket protocol is at the core of many real-time chat features, customer support tools, social media and messaging apps, and livestream platforms. It delivers instant messages to give users an interactive chat experience.

Data Broadcast: Apps that require live data broadcast leverage WebSockets to provide real-time updates. These include live scores, stock market updates, and news alerts. This is particularly essential on trading platforms, where even minimal delays can have significant financial impacts.

Data Synchronization: WebSockets provides live updates in online polls, quizzes, and collaborative databases. It facilitates real-time data exchange to improve the efficiency of IoT devices and deliver consistent, up-to-date information.

Collaborative Environments: WebSockets supports team collaboration on shared documents, whiteboards, and presentations by ensuring everyone can see updates and revisions as they happen.

In-App Alerts and Notifications: WebSockets provides a wide range of event-driven app notifications and alerts through system messages and activity updates.

Recent Trends and Developments

WebSockets and WebRTC are still evolving as AI and other emerging technologies create new applications. Now, WebSockets are being integrated with technologies such as WebAssembly, Progressive Web Apps (PWAs), and Service Workers to improve the efficiency of web apps.

Video compression techniques (codecs) like Lyra and AV1 continue to advance video and audio quality. Developed by Google, Lyra minimizes data consumption, ensuring smooth communication even on slow networks. And AV1's lower bitrates allow for more efficient compression than older codecs.

WebTransport is a new web API that is gaining attention as a WebRTC alternative for streaming and broadcasting. It uses the QUIC protocol and delivers low-latency, bidirectional data transmission and efficient server-client communication with more flexible transport capabilities than HTTP/3.

Conclusion

Now you understand what makes WebRTC and WebSocket such potent tools for building real-time web applications. While WebRTC is ideal for apps that need peer-to-peer communication and multimedia streaming, WebSockets works better for bidirectional, low-latency data exchange between clients and servers.

To learn more, check out some of these related tutorials and other resources for building reliable real-time chat, video, and audio features into your app:

• 36 Essential WebRTC Learning Resources
• WebRTC Tutorials
• How to Avoid Multiple WebSocket Connections in a React Chat App
• Debugging Multiple WebSocket Connections
• Chat App Tutorials
• Activity Feed Tutorials
• Video App Tutorials