What is WebRTC, and Why Is It Important?

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.

The Advantages and Disadvantages of WebRTC

WebRTC allows communication in a peer-to-peer (P2P) model, where the participating peers can connect directly without a server involved. Unlike P2P, another kind of communication comes in a client-server model. The server handles the media relay and signaling. A typical commercial RTC system uses a client-server model.

WebRTC P2P model

In an essential WebRTC P2P connection, the two communication peers connect directly without a server. If there is a NAT between them, WebRTC uses a STUN server for firewall traversal. If a STUN server doesn’t work well, a TURN server is used to relay media data for both peers. Whether to use STUN or TURN depends on an ICE server.

Usually, these three servers run on one single physical server. Since there is no server to guarantee bandwidth and computational power, the transmission quality is vulnerable to congestion that often occurs on the Internet.

The advantage of the P2P model is its low cost of communication. As no server is involved, bandwidth and computation costs are eliminated. Without the server relay, traveling time between the two ends will be reduced. However, the disadvantage of the P2P model is its low quality of communication. Without a back-end media server, the Internet’s best-effort delivery approach often results in stutter and delay in P2P real-time communication.

Client-server model

In a typical case of RTC connections in a client-server model, the two communication peers connect indirectly via a cluster of servers, which take care of many things, including network access scheduling, routing, load balancing, media relay and mixing, and other tasks. There are many servers in the cluster, organized in a sophisticated way to form a data transmission network. In ZEGOCLOUD’s case, MSDN plays the role of this data transmission network. In a route between the two communication peers, the quality of transmission is safeguarded by sufficient bandwidth and computational power of the servers.

Scenarios

Let’s take a closer look at several communication scenarios and see the disadvantages of the P2P model with WebRTC as an example.

One-on-one video call is a typical example. Each user publishes an uplink stream and subscribes to a downlink stream, and the bandwidth required under the P2P model is similar to that required under the client-server model.
Many-to-many. In a video conference with multiple parties, users must publish (X-1) uplink streams and subscribe to (X-1) downlink streams. In the client-server model, each user has to publish one uplink stream and subscribe to (X-1) downlink streams. The bandwidth consumption for uplink streaming in the P2P model will be much heavier than that in the client-server model. Hence, it is tough to scale up the number of users in the P2P model.
One-on-many. A typical example of this kind of scenario is one-way live streaming. The P2P model is unsuitable for such a scenario because it cannot support the scale of concurrent stream subscribers.
Communication with other protocols. Live streaming can be used as an example for this scenario too. For live streaming, a content distributing network (CDN) must support a large scale of concurrent users. To use a CDN, streams need to be delivered using the RTMP protocol, which uses AAC and H.264 to encode and decode media data. As WebRTC uses RTP/RTCP protocol for data transmission, a WebRTC stream needs to be converted to RTMP protocol, and its media payload needs to be transcoded from OPUS&VP8 to AAC&H.264 or the other way around. An MCU is needed here to perform transcoding and transmuting. Apparently, WebRTC in the P2P model doesn’t satisfy this requirement.

Overall, WebRTC in the P2P model has many constraints. Among them are bandwidth, scalability, stability, smoothness, real-timeness, and compatibility. As discussed above, its l

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 is the Benefit 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 build 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.