First released in 2011 WebRTC has been a standardized and open source video streaming protocol. It is extremely important and supported by all major browsers. It can help with real time communication across the world.
Nowadays, developers can easily make apps that play video and audio really well without using extra software or plugins. This article teaches you all the basics about WebRTC.
What does WebRTC mean?
It is short for Web Real-Time Communication. This is a tool that lets developers add live communication features to their apps without needing extra software. It works through browser-based audio and video streaming directly between users.
So no extra downloads are needed. It is extensively accessible because of the backing of major corporations.
The start of WebRTC:
Google created WebRTC first. It came from buying a company called GIPS. They had already had made many of the major features of RTC. WebRTC is being used for multiple purposes.
How Does WebRTC Work?
WebRTC lets developers add real-time features like video calls to their websites. It does this with three things:
Special code
Websites use a special coding language like JavaScript and tools like APIs to talk to WebRTC protocol.
Direct connection
The direct peer to peer connection is made through WebRTC without needing to set a server between. This is like two friends calling each other directly instead of going through an operator.
Making data smaller
WebRTC shrinks videos and audio before sending them (compression) and inflates them back to normal size when they arrive (decompression).
In short, WebRTC protocol acts like a middleman, letting websites use your browser for direct connections and smoother communication.
Why would you use WebRTC?
Advantages of WebRTC
WebRTC offers several advantages that make it attractive for both users and developers:
- No downloads needed
Bye bye to extra apps/plugins! With WebRTC you can directly video chat or stream live in the browser.
- Speedy Streams
WebRTC transmits video and audio in a low latency manner. This means real-time interaction hence a smooth experience for you.
- Bends with the Web
No matter what type of internet connection you use, WebRTC changes the stream quality to keep down the interruptions.
- Free and open for all
Developers and coders like WebRTC because it is completely free and open for use. It also has a big community to back it up all the time and for upgrades.
- Compatible With Most Browsers
Regardless of whether you are using Chrome, Firefox, or Edge WebRTC is a widely-compatible technology to facilitate easy access.
Disadvantages of WebRTC
- WebRTC’s Bandwidth Hiccup:
WebRTC is great for small groups, but for big audiences, it can struggle. This is because it relies on a direct connection between viewers, which can be demanding on internet speed (bandwidth). Think of it like a narrow pipe – if too much water (data) tries to flow through, things get backed up. For larger streams, a special server helps manage the flow, ensuring a smoother experience.
- Connection Quality Matters:
Just like any online video, WebRTC streams are affected by your internet connection. Slow or unstable connections can lead to choppy or pixelated video.
How Does WebRTC Compare to Other Protocols?
As mentioned earlier, WebRTC is more than simply a protocol; it’s a substitute for a number of protocol-only video delivery formats, such as SRT, RTMP, RTSP, and HLS.
- WebRTC vs. RTMP and SRT
Historically, RTMP was the most popular protocol since it could be found in Adobe Flash. Nevertheless, RTMP has increasingly lost its popularity as a streaming workflow for some people after Flash’s demise. Yet still, it remains as one of the best choices in terms of compatibility with encoders when it comes to initial contribution. Many workflows have RTMP encode before video assets are transcribed as HLS for final delivery.
RTMP usually has a latency of five seconds which is not as fast as WebRTC’s incredible speed. However, it does better than HLS and DASH in respect to latency. A lot of times WebRTC is equal or more than RTMP in matters like security and compatibility. But this is due to slackening browser support for RTMP whereas functionality is an area where RTMP leads with features such as captions/timed metadata and ad markers.
WebRTC proves to be an excellent choice for end-to-end streaming that doesn’t necessitate transcoding. Particularly when utilizing web browsers at both ends. However, many content distributors still opt for RTMP when they require greater control over encoding settings.
Additionally, it’s crucial to compare WebRTC with Secure Reliable Transport. SRT emerged as an alternative to RTMP. It addresses issues related to poor network conditions to ensure low-latency and dependable streams. As a result, the disadvantages and advantages of SRT and WebRTC are similar. SRT can be employed for initial contribution before transcoding.
- WebRTC vs. RTSP
The Real-Time Transport Protocol (RTP) is responsible for moving data from clients to servers; the Real-Time Streaming Protocol (RTSP) does not handle this task. However, RTSP facilitates multimedia playback and is frequently compared to RTMP.
RTSP and WebRTC often complement each other, particularly because RTSP is a common protocol for IP cameras. These cameras utilize RTSP for initial data contribution, then transcode the footage into WebRTC for final delivery.
This approach significantly reduces latency, a crucial factor in surveillance applications. WebRTC is not employed for both ends of the process due to its primarily browser-based nature, necessitating the use of RTSP for encoding by IP cameras.
Is it possible to stream live video using WebRTC?
Absolutely! In fact, WebRTC is a perfect streaming video server tool.
Here’s why:
- Real-time focus
WebRTC was constructed for real-time communication; thus, this feature makes it ideal for live streams where low latency (little delay) is indispensable.
- Browser friendly
Viewers can now live-stream directly from their browsers on the webpage without the extra software requirements of WebRTC.
- Direct connections
WebRTC can be used to set-up peer-to-peer connections between viewers in the case of some stream types, and, if the servers’ load is reduced, the quality of the stream may improve.
- The incorporation with Other Technologies
WebRTC can be combined with other technologies, such as adaptive bitrate streaming. This to improve the quality and consistency of live streaming.
To sign off
We hope you now understand what is WebRTC. It is a powerful protocol that enables online real-time communication. Uses for this technology in live streaming applications are widespread.
With its low latency and cost-effectiveness, it is an excellent choice for developers looking to create real-time communication apps.