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WebRTC and Low-Latency Streaming

Learn more about streaming latency at Streaming Media East 2022.

Read the complete transcript of this clip:

Ryan Jespersen: When we talk about low latency, what are we talking about? Are we talking about contribution? Are we talking about all of us working together on content in real time? Or are we talking about distribution? I think there's three different pieces where that's key. Some of the questions we're getting, as far as the contribution side, are how can we use WebRTC for encoder workflows and high-quality workflows? And how can we add the interactivity piece like we can in Zoom while still retaining the broadcast quality that we want in the contribution side? I think what you're seeing a lot of is, WebRTC is doing a lot on that side from a client SDK side and trying to build those workflows into encoders.

The other part that I think very compelling is using the cloud as the mixer for now doing these new advanced workflows. We had BirdDog as a sponsor, right, and congratulations to them. They went public on the Australian stock exchange about a month ago. And with BirdDog, obviously they're doing NDI, but we worked with them actually doing NDI to WebRTC transmuxer so that people could now mix NDI BirdDog cameras in the cloud using WebRTC, and then distribute over a WebRTC CDN.

And then there's a distribution part. And I think obviously I'm pitching WebRTC here. I think WebRTC can be used end-to-end for contribution mixing and distribution, keeping the native protocol end-to-end. I think the beautiful thing about WebRTC is that by default over UDP it prioritizes latency over anything else. It would rather drop packets than take any more time. And I think as people talk about SRT or they talk about other HTTP protocols or even RTMP, these are TCP protocols that actually try to mimic--especially with SRT or Zixi--the resiliency of TCP, but while still trying to act like UDP. The reality is if you want speed, you've gotta drop packets on bad connections or find a way to adaptively adapt the encoding at source. And that's what WebRTC does natively.

So there's something beautiful about that workflow. As WebRTC matures, people are building broadcast-quality tools around that to actually do the contribution side as a way to replace it.

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