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A Comprehensive Guide to WebRTC Samples

WebRTC Samples

The WebRTC brand has been gaining a lot of traction lately as more people are understanding its potential. The W3C and IETF have proposed various standards to enable real-time, peer-to-peer communication on the web, and Google has been spearheading the implementation with its Chrome browser.

With all this excitement, you may be wondering where you can get started with WebRTC. If so, you've come to the right place.

In this blog post, we'll take a comprehensive look at the WebRTC samples repository on GitHub. We'll walk you through the different types of samples that are available in the repo. By the end of this post, you should have a good idea to get started with WebRTC.

However, if you don't want to go through all the hassle of WebRTC and want a faster approach, try Dyte and go live in mins not months.

Different WebRTC Samples on GitHub

These samples are intended to illustrate the various features of WebRTC in the simplest way possible. Various viewport sizes are supported by all live demos.

Data channels, peer connections, and getUserMedia are canonical demos here. Let’s have a look at each of them in detail.


The first and most important WebRTC API is getUserMedia(). This API allows web applications to access the user's microphone and camera. It is available in all modern browsers.

Access media devices
Basic getUserMedia demo
Use getUserMedia with canvas
Use getUserMedia with canvas and CSS filters
Choose camera resolution
Audio-only getUserMedia() output to local audio element
Audio-only getUserMedia() displaying volume
Record stream
Screensharing with getDisplayMedia
Control camera pan, tilt, and zoom
Control exposure


WebRTC allows you to choose which devices to use for communication. You can select the audio input and output devices such as microphones, speakers, and headphones. You can also select video input devices such as webcams and digital camcorders.

Query media devices
Choose camera, microphone, and speaker
Choose media source and audio output

Stream Capture

WebRTC provides APIs for capturing media streams from WebRTC-compatible devices. The MediaStream API represents synchronized streams of media content. The getUserMedia() method prompts the user for permission to access the specified media types (audio and/or video), and if the user grants permission, returns a MediaStream object containing the requested types of media tracks (specified by kind).

Stream from canvas or video elements
Stream from a video element to a video element
Stream from a video element to a peer connection
Stream from a canvas element to a video element
Stream from a canvas element to a peer connection
Record a stream from a canvas element
Guiding video encoding with content hints

RTCPeer Connection

WebRTC uses the RTCPeerConnection API to set up real-time media communications between web browsers, typically involving audio and video streaming.

Controlling peer connectivity
Basic peer connection demo in a single tab
Basic peer connection demo between two tabs
Peer connection using Perfect Negotiation
Audio-only peer connection demo
Change bandwidth on the fly
Change codecs before the call
Upgrade a call and turn video on
Multiple peer connections at once
Forward the output of one PC into another
Munge SDP parameters
Use pranswer when setting up a peer connection
Constraints and stats
More constraints and stats
RTCPeerConnection and requestVideoFrameCallback()
Display createOffer output for various scenarios
Display peer connection states
ICE candidate gathering from STUN/TURN servers
Do an ICE restart
Web Audio output as input to peer connection
Peer connection as input to Web Audio
Measure how long renegotiation takes
Choose scalablilityMode before call - Scalable Video Coding (SVC) Extension


The RTCDataChannel API allows two connected peers to share arbitrary data. Data channels are available in Chrome and Firefox.

Send arbitrary data over peer connections
Transmit text
Transfer a file
Transfer data

Video Chat

A typical WebRTC application has two components: a client application running in the browser and a server application running on a web server. The client application needs to know how to connect to the server so that it can exchange signaling information with other clients (peers). Once two peers are connected, they can use the RTCPeerConnection API to communicate directly with each other.

Full-featured WebRTC application
AppRTC video chat client

Insertable Streams

Insertable streams allow you to add your own audio or video tracks to an existing media stream. This can be used, for example, to add subtitles or watermarks to a video chat. The insertable streams API is available in Chrome and Firefox.

API for processing media
End-to-end encryption using WebRTC Insertable Streams
(Experimental)Video analyzer using WebRTC Insertable Streams
(Experimental)Video processing using MediaStream Insertable Streams
(Experimental)Audio processing using MediaStream Insertable Streams
(Experimental)Video cropping using MediaStream Insertable Streams in a Worker
(Experimental)Integrations with WebGPU for custom video rendering

Final Verdict!

The GitHub repository for WebRTC is a great resource for developers who want to get started with the protocol. The samples provide a comprehensive guide to all aspects of WebRTC development, from basic tutorials to more advanced features. If you're looking to get started with real-time communication applications, the GitHub repo is a great place to start.

Nevertheless, if you want to get started faster and with fewer complications then Dyte is the perfect solution for you. With Dyte, you can easily create and manage video and voice conferences with colleagues, clients, or customers in minutes - without any of the hassles of installing software or dealing with complicated settings.

Try Dyte with our sample apps below:

Android UI Kit Sample App
React UI Kit Sample App
Backend Sample App
Flutter Sample App
Kotlin Sample App
NextJS Sample App
ObjC Sample App
React Native Sample App
React UI Kit Sample App
Streams Middleware Sample App
Swift Sample App
Vanilla JS UI Kit Sample App
Vue Sample App
Web Core JS Sample App

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Don't just take our word for it; try it for yourself!  Dyte offers free 10,000 minutes every month to get you started quickly.
If you have any questions or simply want to chat with us, please contact us through support or visit our developer community forum. Looking forward to it!

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