What is WebRTC (Web Real-Time Communications)?

Your search for real-time communication solutions with ultra-low latency may have resulted in a repetition of a common theme among all options. That theme is the wide use of WebRTC. Thus, if you are wondering what is WebRTC, this article will answer it for you. So, grab a cup of coffee and join us on this journey into the exciting world of WebRTC.

What is WebRTC?

WebRTC, or Web Real-Time Communication, is Google’s free and open-source technology to facilitate real-time communications, introduced in 2011. What is it used for? It enables real-time communication between web browsers and mobile applications. To explain WebRTC’s meaning in simple words, it enables web applications to have real-time communication capabilities without needing any plugins or additional software.

Moreover, it is essential to have real-time communication apps like video conferencing, voice calling, and file sharing. It is primarily based on standardized APIs, making it a popular choice for developers to build web applications. WebRTC’s native support in all major browsers like Chrome, Firefox, and Edge also helps make it popular.

History of WebRTC

WebRTC is a groundbreaking technology that has significantly transformed the landscape of digital communication. The journey of WebRTC began in 2011 when Google released an open-source project aimed at enabling real-time communication in web browsers without the need for plugins or third-party software. This project laid the foundational framework for WebRTC, which was designed to support browser-to-browser applications for voice calling, video chat, and peer-to-peer file sharing.

The technology quickly gained momentum due to its potential to facilitate direct and efficient communication. By May 2011, Google acquired Global IP Solutions (GIPS), a company specializing in web-based voice and video processing, which further accelerated the development of WebRTC. This acquisition helped integrate superior voice and video capabilities directly into web browsers.

In 2012, the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF) began working on standardizing WebRTC, ensuring compatibility and reliability across different platforms and browsers. WebRTC 1.0 became an official W3C Recommendation by January 2021, marking a significant milestone in its development.

Today, WebRTC is supported by all major browsers, including Chrome, Firefox, Safari, and Edge. Its ability to perform high-quality, real-time communication has made it a key player in various applications, from video conferencing and live streaming to online gaming and telehealth, continuing to push the boundaries of what can be achieved in web communication.

Key Features of WebRTC

WebRTC (Web Real-Time Communication) is a transformative technology that enables real-time communication directly in web browsers. Here are the key features that make WebRTC a powerful tool:

Peer-to-Peer Connections: WebRTC facilitates direct peer-to-peer data sharing and video/audio streaming, enhancing the quality and efficiency of communications by eliminating intermediaries.
Real-Time Communication: It supports instant video, audio, and data transfers, crucial for applications like telehealth, online gaming, and virtual classrooms, ensuring smooth and immediate interactions.
High-Quality Video and Audio: WebRTC is built to maintain high-quality communication with features like adaptive video resolution, echo cancellation, and noise suppression, even in varying network conditions.
Encryption and Security: Security is integral to WebRTC, which mandates encryption for all data transfers using protocols like DTLS and SRTP, protecting the privacy and integrity of the communication.
Device and Network Independence: WebRTC works across different devices and networks, making it highly accessible without needing plugins or third-party software, enhancing user experience and accessibility.
Scalability and Flexibility: While optimized for peer-to-peer communications, WebRTC can scale up for larger broadcasts using mechanisms like SFUs or MCUs, adapting to a range of communication needs.
Open-Source and Community-Driven: As an open-source project, WebRTC benefits from a robust community contributing to its development, ensuring it remains cutting-edge.

These features make WebRTC a robust choice for developing applications that require reliable, high-quality, and secure real-time communication.

The Advantages and Disadvantages of WebRTC

While WebRTC offers powerful capabilities for real-time communication within web browsers, developers must navigate its disadvantages, such as scalability challenges and browser compatibility issues. By understanding these trade-offs, developers can more effectively use WebRTC to build robust and user-friendly communication applications.

Advantages of WebRTC:

Real-Time Communication: WebRTC allows for real-time audio and video communication directly in the browser without the need for plugins or third-party software. This makes it incredibly convenient for users to engage in video chats, voice calls, and data sharing with minimal setup.
High-Quality Media Streaming: WebRTC supports high-quality video and audio streaming with dynamic adjustment to bandwidth availability. This means it can provide clear and uninterrupted communication even in fluctuating network conditions, ensuring a superior user experience.
Open Standards and Free: WebRTC is an open-source project that uses free and open standards. This accessibility encourages developers worldwide to use and contribute to the technology, fostering a large community and evolving the platform rapidly.
Peer-to-Peer Connection: By establishing direct peer-to-peer connections, WebRTC minimizes latency and preserves bandwidth. This direct connection ensures that media data doesn’t need to travel through intermediate servers, leading to faster and more secure communications.
Cross-Platform Compatibility: WebRTC is supported by all major browsers, including Chrome, Firefox, Safari, and Edge. This broad compatibility helps developers reach a wider audience without worrying about platform-specific constraints.
Data Security: WebRTC implements robust security standards like DTLS (Datagram Transport Layer Security) and SRTP (Secure Real-Time Transport Protocol) for encryption and secure data transfer. This commitment to security is crucial for sensitive communications.

Disadvantages of WebRTC:

Browser Compatibility Issues: While most modern browsers support WebRTC, discrepancies in implementation may cause issues. Some older browsers or certain versions may not fully support all WebRTC features, leading to inconsistencies in user experience.
Complexity in Scaling: Managing peer-to-peer connections with WebRTC becomes complex as the number of participants increases, especially in large group calls or broadcasts. This can require additional server infrastructure, like TURN or STUN servers, to coordinate connections and traverse NAT/firewalls.
Lack of Universal Standards: Despite being based on open standards, variations in how different browsers and devices implement WebRTC can lead to compatibility challenges. Developers often need to write additional code to handle these variations, increasing the complexity of their applications.
Privacy Concerns: The requirement for WebRTC applications to request access to a user’s microphone and camera can raise privacy issues. Users may be wary of granting these permissions, especially if the reasons aren’t clearly communicated.
Battery Drain on Mobile Devices: WebRTC can be resource-intensive, particularly for mobile devices. The use of real-time audio and video processing can lead to significant battery drain, affecting the usability of mobile applications.
Dependency on Internet Quality: Since WebRTC is designed for real-time communication, its performance heavily depends on the quality and stability of the user’s Internet connection. Poor bandwidth or unstable connections can degrade the quality of communication, affecting audio and video clarity.

WebRTC Use Cases

WebRTC is a powerful technology enabling real-time communication in web browsers. Here are four key use cases that illustrate its versatility:

Video Conferencing and Virtual Meetings

WebRTC powers video conferencing tools that are crucial for remote work. It allows high-quality video and audio calls directly in the browser, with features like screen sharing and file transfer, enhancing collaboration. Platforms like Google Meet use WebRTC to connect people globally, making it indispensable for virtual meetings.

Telehealth and Remote Medical Consultations

In healthcare, WebRTC enables telehealth services, allowing patients to have video consultations with doctors. This is vital for providing accessible care in remote areas. It supports secure and private interactions, essential for handling sensitive medical information, and improves healthcare outcomes by facilitating remote diagnostics and treatment planning.

Customer Support and Live Chat Services

WebRTC integrates into customer support systems to offer live video, voice, and chat directly on websites. This improves the support experience by enabling co-browsing, where representatives interact with the customer’s screen, resolving issues more efficiently and enhancing customer satisfaction.

Interactive Broadcasting and Live Streaming

WebRTC transforms broadcasting by enabling real-time media streaming. Platforms like Facebook Live use it for low-latency broadcasts to large audiences. It supports interactive elements like live polls and chats, making it perfect for events, concerts, and webinars where viewer engagement is key.

How Does WebRTC Work?

To fully understand how WebRTC works, you first need to know the technologies it depends upon for efficient working. It generally uses a combination of RTP, SDP, ICE, and UDP. When two users want to communicate using WebRTC, their browsers establish a connection through a signaling server. The signaling server exchanges information between the browsers, including network addresses and media capabilities.

For initiating a communication session between two users, the first sends an offer to the other user’s browser. Furthermore, it contains information about the sender’s browser’s media capabilities and the session’s details. In response, the receiver browser sends its media capabilities and network information. After the exchange, the two browsers use ICE to determine the best path for data transmission.

ICE tries different methods to establish a direct connection between the two browsers. If a direct connection isn’t possible, the data will be relayed through a TURN (Traversal Using Relays around NAT) server. After a peer-to-peer connection is established, users can now communicate in real-time. The media is usually sent using RTP and UDP, which provide low-latency real-time communication.

What are the Benefits of Using WebRTC?

Using WebRTC tutorial during web real-time communication app development offers many advantages. These benefits range from low-latency real-time communication to wide platform support. Summarized below are some of these benefits:

Low Latency: WebRTC has a primary advantage in enabling real-time communication with minimal latency, resulting in negligible delay between sending and receiving a message. Low latency is possible because it doesn’t require any intermediary to establish a connection.
Security: WebRTC is also highly secure as it uses end-to-end encryption to secure user communication. Moreover, its end-to-end encryption makes it a perfect option for communication applications prioritizing user privacy. Additionally, it also creates obstacles for any external parties attempting to intercept or manipulate the communication.
Ease of Use: Due to standardized APIs, WebRTC is easy to use and can be effortlessly integrated into web applications with minimal effort. Moreover, it eliminates the need for additional software or plugins, saving time and resources.
Cross-Platform Compatibility: Having native support by all major browsers is also among the key benefits of WebRTC. The list of browsers supporting it includes giants like Google Chrome, Mozilla Firefox, and Microsoft Edge. This wide support means that using WebRTC ensures there are no compatibility issues across different platforms and devices.
Cost-Effective: It is free and open-source, allowing you to develop real-time communication applications without paying licensing fees. Additionally, with its open-source nature, a robust community exists to address any challenges you encounter while incorporating its API.

Commercial RTC Solutions VS. WebRTC (P2P) Solutions

It would be fair to say that commercial RTC solutions have a clear competitive edge over pure WebRTC (P2P) solutions. To be more specific, the following are some significant aspects where commercial systems outperform pure WebRTC solutions:

1. Cost-efficiency

Building a commercial-grade RTC system is an arduous task that requires a significant amount of R&D time and effort.

For instance, companies will need to assemble a team of veterans with at least five years of multimedia software development experience. It would take about six months to a year to build a functional RTC system and make it generally available. The multimedia team will have at least need 6 developers to work on the essential modules of WebRTC, such as transmission (RTP/RTP), voice engine (NetQ, ANS, AGC, and AEC), and video engine (jitter buttering, etc.).

A commercial RTC platform allows you to add RTC capabilities to your product with a few lines of code and get your project off the ground in a couple of weeks. In addition, it allows your platform to use RTC services in a pay-for-usage model. So, by using a commercial RTC system, you can save development costs and accelerate your time to market. Furthermore, you can enjoy commercial-grade RTC performance without maintaining all the underlying infrastructures.

2. Scalability

It is tough to scale up a pure WebRTC (P2P) solution because:

There is no server for a pure WebRTC (P2P) solution (ICE, STUN, and TURN are only for NAT traversal) to support system expansion;
Secondly, the nature of the P2P communication model consumes an excessive amount of bandwidth at the user end and makes it very hard to scale up;
Lastly, a sophisticated data transmission network is necessary to support a high volume of concurrency and cross-border communications with global coverage.

A commercial RTC platform can save you from the constraints above, given its server-centric architecture and robust data transmission network. With clusters of servers taking care of heavy computations and transmissions such as nearby network access, intelligent routing, transcoding, transmuting, and stream mixing, a commercial RTC system doesn’t have a limitation on scaling up. Meanwhile, to guarantee the user experience in cross-border communications, a commercial RTC platform typically has many servers deployed worldwide.

ZEGOCLOUD has deployed more than 500 BGP servers around the globe. It has joined all these infrastructure components with its sophisticated transmission and routing algorithms to form a global data network called Massive Serial Data Network (MSDN), which allows you to scale up your user base without constraints and accelerate voice & video data transmission to reach an ultra-low latency.

3. User experience

The most critical factors in evaluating an RTC platform’s user experience include audio/video quality, smoothness, and latency, which significantly impact user experience. A P2P WebRTC-based system cannot support multiple streams of high definition because of its bandwidth constraint at the user end. Without the support of a data transmission network, a pure WebRTC (P2P ) solution cannot guarantee a smooth and low-latency communications experience.

On the other hand, a commercial RTC platform based on the client-server model has a data transmission network. A communication peer just needs to publish an uplink stream and can either subscribe to a single downlink stream or multiple streams. This gives a commercial RTC platform a much higher capability for streaming high-definition videos.

Furthermore, mature commercial RTC platforms like ZEGOCLOUD have been in the market for 7 years. As driven by clients’ demands, ZEGOCLOUD has optimized its user experience to stay ahead of the curve. After all, if a commercial system cannot deliver excellent performance, its clients won’t pay to use its service.

4. Use case support

WebRTC is an open-source technology with authentic voice and video call abilities but was not designed for any specific business purposes or use cases in the first place.

5. System compatibility

RTC technology has evolved for decades, and there have been various protocols to cater to different needs, to name a few: SIP, H323, and WebRTC. Moreover, in terms of transmission protocols, there are RTMP and RTP/RTP.

Regarding encoding and decoding, there are H264 and VP8 for video and AAC and OPUS for audio. For a platform, its users may come from various kinds of terminals. For example, a user who makes a video call from an RTMP-based RTC app won’t be able to communicate with the other user using a Google Chrome browser. There must be a gateway server in between to do protocol translating and media data payload transcoding. This scenario is far beyond what a typical pure WebRTC(P2P) solution can do.

A commercial RTC platform is compatible with terminals of different kinds. For example, ZEGOCLOUD allows access from various kinds of terminals besides its own native SDK, such as WebRTC and SIP. ZEGOCOUD’s MSDN is compatible with a WebRTC terminal through a WebRTC gateway. The WebRTC gateway translates the WebRTC protocol into ZEGOCLOD.

Take Advantage of WebRTC with ZEGOCLOUD SDK

ZEGOCLOUD, a global cloud communication service provider, uses a proprietary WebRTC gateway server to bridge its data network and the web applications to support web client access.

ZEGOCLOUD’s RTC platform mainly consists of an acceleration data network called Massive Serial Data Network (MSDN) and a client-side RTC engine. Both components were built in-house by ZEGOCLOUD. The platform supports access from native mobile applications and web applications. It also enables communications between native mobile applications and applications.

When it comes to finding solutions that employ WebRTC meaning to its fullest, ZEGOCLOUD SDKs and APIs stand out the most. ZEGOCLOUD offers a wide range of solutions encompassing text messaging, voice, video, and live streaming. By utilizing ZEGOCLOUD SDKs and APIs, you can create a high-quality real-time communication mobile and web application at a reasonable cost.

By integrating ZEGOCLOUD video calls, voice chat SDKs, and APIs into your application, you can take advantage of the benefits of WebRTC without having to manage the infrastructure yourself. Moreover, you can also easily integrate them into your apps as it provide detailed documentation along with video tutorials and comprehensive FAQ sections.

Prominent Features and Benefits of ZEGOCLOUD SDKs and APIs

The wide range of features offered by ZEGOCLOUD SDKs and APIs make them popular among developers. Summarized below are some of these key features and benefits:

High Scalability: ZEGOCLOUD SDKs and APIs are highly scalable, which means they can handle large volumes of traffic without any degradation in performance. It is carefully made possible through a distributed architecture allowing automatic load balancing and failover.
Ultra-Low Latency: The SDKs and APIs offered by ZEGOCLOUD to build real-time communication apps ensure ultra-low latency and provide a realistic experience to users. It also provides high-quality video and audio transmission and features like recording and playback.
Security: Any real-time communication app nowadays is incomplete without advanced security measures. ZEGOCLOUD APIs and SDKs support highly advanced end-to-end encryption algorithms. As for data security and protection, these fully comply with GDPR and HIPAA.
Cross-Platform Compatibility: All ZEGOCLOUD SDKs and APIs are fully compatible with all major web browsers and operating systems. It means you can deliver real-time communication experiences to users on any device. Moreover, all the APIs and SDKs fully complement each other, turning your app into the best one.

Conclusion

All things considered, knowing what is WebRTC is essential to building a low-latency real-time communication app. It is the tool most WebRTC APIs and SDKs use to ensure low latency in real-time communication applications. To take full advantage of WebRTC’s capabilities, you should use ZEGOCLOUD video call SDK or voice call SDK. You can build secure and cost-effective real-time communication solutions with these amazing tools.

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