React Livestreaming Tutorial

In this tutorial we'll quickly build a low-latency in-app livestreaming experience. The livestream is broadcasted using Stream's edge network of servers around the world.

We'll cover the following topics:

• Ultra low latency streaming
• Multiple streams & co-hosts
• RTMP in and WebRTC input
• Exporting to HLS
• Reactions, custom events and chat
• Recording & Transcriptions

Let's get started, if you have any questions or feedback be sure to let us know via the feedback button.

For this tutorial, you'll need a few tools to be installed on your device. You can skip this step in case you already have them installed.

• Node.js (version 18 or higher)
• Yarn (version 1.22 or higher)

In this step, we will create a new React application using the Vite CLI, and install Stream's React Video SDK. We recommend using Vite because its fast and easy to use.

The following code shows how to publish from your device's camera. Let's open App.tsx and replace its contents with the following code:

If you run the app now, you'll see that the client fails to connect. To fix that, we need to provide valid credentials. They shall be set in the following variables:

Replace them now with the appropriate values from the table below:

When you run the app now you'll see a text message saying: TODO: render video.

Before we get around to rendering the video let's review the code above.

User setup

First we create a user object. You typically sync these users via a server side integration from your own backend. Alternatively, you can also use guest or anonymous users.

Client setup

Next, we initialize the client by passing the API Key, user and user token.

Create and join call

The most important step to review is how we create the call. Stream uses the same call object for livestreaming, audio rooms and video calling. Have a look at the code snippet below:

To create the first call object, specify the call type as livestream and provide a unique callId.

The livestream call type comes with default settings that are usually suitable for livestreams, but you can customize features, permissions, and settings in the dashboard. Additionally, the dashboard allows you to create new call types as required. For more information, check the Call Types docs.

Finally, using call.join({ create: true }) will not only create the call object on our servers but also initiate the real-time transport for audio and video. This allows for seamless and immediate engagement in the livestream.

Background noise in a livestream is never a pleasant experience for the listeners and the hosts.

Our SDK provides a plugin that helps to greatly reduce the unwanted noise caught by your microphone. Read more on how to enable it here.

In this step we're going to build a UI for showing your local video with a button to start the livestream.

In App.tsx replace the LivestreamUI component with the following code:

Upon running your app, you will be greeted with an interface that looks like this:

Stream uses a technology called SFU cascading to replicate your livestream over different SFUs around the world. This makes it possible to reach a large audience in realtime.

Now let's press Start Livestream in your new app and click the Join Call button below to watch your livestream in another tab in your browser. Please, make sure the call is created using the credentials from section Step 2 before you click Join Call:

Let's take a moment to review the code above.

You can see we use a few hooks to get the call state. The most important ones are:

The Call & Participant state lists all available hooks you can use for accessing call and participant state.

The layout is built using standard React DOM elements. The ParticipantView component is provided by Stream's React Video SDK. ParticipantView renders the video and a fallback. You can use it for rendering the local and remote video.

Backstage mode

In the example above you might have noticed the call.goLive() method and the isCallLive value. The backstage functionality is enabled by default on the livestream call type. It makes it easy to build a flow where you and your co-hosts can setup your camera and equipment before going live. Only after you call call.goLive() will regular users be allowed to join the livestream.

This is convenient for many livestreaming and audio-room use cases. If you want calls to start immediately when you join them that's also possible. Simply go the Stream dashboard, click the livestream call type and disable the backstage mode.

The call.goLive method can also automatically start HLS livestreaming, recording or transcribing. In order to do that, you need to pass the corresponding optional parameter:

The example above showed how to publish your device camera to the livestream. Almost all livestream software and hardware supports RTMPS. OBS is one of the most popular livestreaming software packages and we'll use it to explain how to import RTMPS. So let's see how to publish using RTMPs. Feel free to skip this step if you don't need to use RTMPs.

Log the URL & Stream Key

Open OBS and go to settings -> stream

• Select "custom" service
• Server: equal to the rtmpURL from the log
• Stream key: equal to the streamKey from the log

Press start streaming in OBS. The RTMP stream will now show up in your call just like a regular video participant.

Now that we've learned to publish using WebRTC or RTMP let's talk about watching the livestream.

Watching a livestream is even easier than broadcasting. Our SDK provides a built-in LivestreamLayout that you can use out of the box. Read more about it here: Watching a livestream (WebRTC).

Keep in mind that this component respects the Call Type settings, and it may happen that audio and video are published automatically, unless explicitly disabled.

You can check the full example in this CodeSandbox.

There are two ways on how to end a livestream:

call.stopLive()

This operation will turn on backstage again, and the viewers will be kicked out of the call, but the hosts will remain in the call. Similar like in step 3, call.goLive() will mark the call open for viewers to join.

call.endCall()

This operation will end the call for all participants (including hosts). Unless the call type permissions are changed, neither the hosts nor the viewers will be able to join the same call instance again.

Another way to watch a livestream is using HLS. HLS tends to have a 10 to 20 seconds delay, while the above WebRTC approach is realtime. The benefit that HLS offers is better buffering under poor network conditions. So HLS can be a good option when:

• A 10-20 second delay is acceptable
• Your users want to watch the Stream in poor network conditions

One option to start HLS is to set the start_hls parameter to true in the call.goLive() method, as described above. If you want to explicitly start it, when you are live, you can use the following method:

You can view the HLS video feed using any HLS capable video player.

This tutorial covered broadcasting and watching a livestream. It also went into more details about HLS & RTMP-in.

There are several advanced features that can improve the livestreaming experience:

• Co-hosts You can add members to your livestream with elevated permissions. So you can have co-hosts, moderators etc.
• Custom events You can use custom events on the call to share any additional data. Think about showing the score for a game, or any other realtime use case.
• Reactions & Chat Users can react to the livestream, and you can add chat. This makes for a more engaging experience.
• Recording The call recording functionality allows you to record the call with various options and layouts

It was fun to see just how quickly you can build in-app low latency livestreaming. Please do let us know if you ran into any issues. Our team is also happy to review your UI designs and offer recommendations on how to achieve it with Stream.

To recap what we've learned:

• WebRTC is optimal for latency, HLS is slower but buffers better for users with poor connections
• You setup a call: const call = client.call("livestream", callId)
• The call type "livestream" controls which features are enabled and how permissions are setup
• The livestream by default enables "backstage" mode. This allows you and your co-hosts to setup your mic and camera before allowing people in
• When you join a call, realtime communication is setup for audio & video: await call.join()
• Call state call.state and helper state access hooks exposed through useCallStateHooks make it easy to build your own UI

We've used Stream's Livestream API, which means calls run on a global edge network of video servers. By being closer to your users the latency and reliability of calls are better. The React SDK enables you to build in-app video calling, audio rooms and livestreaming in days.

We hope you've enjoyed this tutorial and please do feel free to reach out if you have any suggestions or questions. You can find the code for this tutorial in this CodeSandbox.

The source code for the companion Livestreaming App, together with all of its features, is available on Github.

Final Thoughts

In this video app tutorial we built a fully functioning React messaging app with our React SDK component library. We also showed how easy it is to customize the behavior and the style of the React video app components with minimal code changes.

Both the video SDK for React and the API have plenty more features available to support more advanced use-cases.

Give us Feedback!

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