webRTC, Part 1 – working w/MediaStreams. Getting and interacting with the camera, mic and screen
This course covers the first part of webRTC: getting and working with the screen, camera and mic feeds. Part 2 covers how to send that feed to another computer with RTCPeerConnection.
What you’ll learn
- Build a webpage that can pull an audio or video stream.
- What webRTC is and why you’d use it.
- Change video streams dynamically, such as framerate or resolution.
- Learn how to manipulate a MediaStream.
Course Content
- Introduction to webRTC –> 6 lectures • 21min.
- Project – getUserMedia playground – (lame) project that teaches you the basics –> 12 lectures • 1hr 37min.
Requirements
This course covers the first part of webRTC: getting and working with the screen, camera and mic feeds. Part 2 covers how to send that feed to another computer with RTCPeerConnection.
It’s 2023. Whether because of the pandemic, chat bots, or cost savings, working remote is a thing. Telemedicine is a thing. Talking to people 6 time zones away is a thing. You can abstract your users from your app to Zoom because it always works, but you lose all control and tracking of the interaction.
Enter webRTC.
WebRTC is one of the browsers must mind blowing apis. It allows access to the mic, camera, even the screen AND to share them across a network socket DIRECTLY to another browser. No server (mostly) or other middleman to add bandwidth, bugs, and chaos.
Along with websockets, webRTC presents the video-side of browser real-time communication, bridging one of the last gaps in both human and web-based communication. There’s a good chance if you’re reading this, you’ve heard about webRTC. Maybe even done a tutorial on it. But how far did you get? In my experience, the vast majority of the material on the web goes no farther than a quick-start, zoom clone. You don’t learn how anything works, never look at the docs, and are stuck at the end wondering what to do now. Is that all webRTC can do? The remaining shred of material is waaaaay over everyone’s head. The fact that the webRTC was released about the same time as the websocket API and most developers still don’t know how to use it is evidence of the gap.
This course is the first step to alleviate that! It is not a quick start guide to webRTC. There are loads of those all over the Internet. You should definitely look elsewhere if you are wanting a 10 minute intro to the 3-4 things you need to know to make a basic zoom clone. On the other hand, if you are looking to really learn one of the most awesome JavaScript APIs that no one seems to know, you should stick around. Like Express and other JavaScript/Node pieces, it’s getting passed over in the wave to learn just enough to get to the term “full-stack developer.” My main goal is to help you figure out how to go from being a good developer to a great developer. Understanding… not just knowing a few methods… WebRTC is part of that!
I first used webRTC in 2015 for a startup similar to telemedicine. I’ve been following since and have been frustrated that it hasn’t gotten more mainstream notice due to Apple’s reluctant support, but mostly because devs don’t know it. It opens the way for so many improvements to existing applications and obvious groundwork for new ones. Let’s change that 🙂 Prepare to for a detailed look at webRTC and start going in-app real-time video/screen chat.
What we cover:
- getUserMedia() – getting access to the mic and camera in the browser
- playing the feed in a <video />
- MediaStream and MediaStreamTrack – what makes up a video feed
- Constraints – getSupportedConstraints() and getCapabilities() – seeing what this browser can do
- applyConstraints – changing the feed on the fly
- Recording video/audio and playing it back
- Capturing the Screen for screen share and recording it
- Changing input/output devices in your feed
If you are familiar with MediaStreams and getUserMedia() and would like to learn how to make a Peer-to-Peer connection, that is available in part 2.