Is it possible to achieve less than a second latency in a video broadcast? What if the stream goes to a thousand people? Yes. How? Let’s answer using the project WorldCast Live as an example. We did it using WebRTC. WCL streams HD concerts to an audience of hundreds and thousands of people.

Why would I reduce the broadcast latency?

Latency less than a second in a videoconference is normal, otherwise speaking is nearly impossible. For one-sided streaming, the latency of 2 or even 20 seconds is fine. For example, TV delay is about 6 sec. However, there are cases where you’d try to reduce it to 1 second.

• Sport events

It’s highly unlikely that the user will be happy when their neighbors shout, GOAL!, and he still sees the ball somewhere in the middle of the field. What if the user is also live betting?

• Interviews

Thanks to the pandemic, not only talks with friends have moved online, but also interviews with celebrities. Take an interviewer’s latency, add to the interviewee’s latency, and add the time spent while it all goes to the user. The more you get, the worse the experience is for all sides.

• Concerts

The WCL player is embedded in different sites. Concerts broadcast online to all of them. If you and your friend use different sites to watch the same concert, it will negatively affect your viewing experience. Thus, the concert organizers are trying to minimize the latency. Although the user wouldn’t really care whether he hears the guitar riff now or in 2 seconds, there is a general trend on minimizing the latency. The faster the better!

And also…

The examples above combine. In WCL viewers and performers talk via a video chat. Therefore, the latency has to be like that of a video chat, as we need the minimum difference in time between the question and the answer.

How to reach low latency in live streaming?

Use WebRTC

The standard WebRTC package offers average latency of 500 ms (half a second). We could’ve finished the article here: creating a video chat where people can connect with each other isn’t difficult. What’s difficult is making it all work stable for thousands of people and customizing streams to improve their quality.

Base WebRTC isn’t about HD streaming. Video and audio quality is enough for speaking but isn’t enough for streaming music. To decrease the latency, WebRTC can reduce the quality and skip parts of the content. To avoid it, you have to go under the hood of WebRTC, which we did.

Set up WebRTC

To ensure that the low latency doesn’t go against quality and the final user’s experience, you need to go for extra development. Here’s what we did for WCL so it can broadcasts concerts to thousands of people:

• Enabled multi-channel audio

The standard audio in WebRTC is mono, it’s 1 channel. The stereo is 2 channels. There are 5 channels on WCL.

• Upped bitrate

WebRTC settings limit the video transmission speed to 500 kb/s. That’s not much for action on the screen: if light colors change quickly with this limit, the quality will be lower because of trying to go inside that channel, pixels might appear. Not a great watching experience. Therefore we’ve increased the bitrate to 1,5 Gb to transmit HD video.

• Increased discretization frequency

By this, we’ve improved the quality of audio and video, low and high frequencies. They are not lost anymore. We can’t disclose what exactly we did, but if you’re interested in that, let us know in the comments!

Scale Kurento

Kurento Media Server is an open-source WebRTC server. Set up Master Kurento to stream video. 500 hundred people will connect directly to Master Kurento, from where they’ll get the stream. If there are more viewers, Edge Kurento is in play, to which other viewers connect. The more viewers there are on the stream, the more Edge Kurento you need to use. All together it creates a tree-like scheme.

When do you leave the latency of more than a second?

When the budget is limited. WebRTC is more expensive than HLS if you need scaling. The WebRTC server on WCL costs $0,17/h which is $122,4 monthly if we take 30 days.

HLS, however, costs $0,023/h and can be turned on and off, unlike WebRTC. If we take 3 hour-long concerts a week, then we’ll spend a bit less than $0,28 for 12 concerts. Note that there are many server providers, the prices are always different, but you can see what it looks like using our project as an example.

TL;DR

Latency less than a second in a broadcast is necessary when there’s communication between participants or a stream where things change all the time. WebRTC makes it possible for even a thousand people if you work with the technology.

If your app is about something from here, make sure to take a look at WebRTC. Contact us, too, we’ll help! Get in touch on our website or DM on Instagram, which we created not so long ago 🙂

The first stable version of subsecond latency on WCL took us 3,5 weeks. If there’s no necessity in very low latency, why spend?