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? 我们讨论的是我们所有人一起实时制作内容吗? 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, 我们如何将WebRTC用于编码器工作流和高质量工作流? 我们如何才能像在Zoom中那样添加交互性，同时保持我们在贡献方面想要的广播质量? I think what you're seeing a lot of is, WebRTC在客户端SDK方面做了很多工作，并试图将这些工作流构建到编码器中.

我认为另一个非常引人注目的部分是使用云作为混合器来完成这些新的高级工作流程. 我们有BirdDog作为赞助商，恭喜他们. 他们大约一个月前在澳大利亚证券交易所上市. And with BirdDog, obviously they're doing NDI, 但是我们和他们一起做了NDI到WebRTC的转换器，这样人们就可以使用WebRTC在云中混合NDI BirdDog相机, and then distribute over a WebRTC CDN.

And then there's a distribution part. And I think obviously I'm pitching WebRTC here. 我认为WebRTC可以端到端用于贡献混合和分发, keeping the native protocol end-to-end. 我认为WebRTC的美妙之处在于，默认情况下，它优先考虑延迟，而不是其他任何事情. It would rather drop packets than take any more time. 我认为当人们谈论SRT或者其他HTTP协议甚至RTMP时, 这些TCP协议实际上试图模仿TCP的弹性，尤其是SRT或Zixi, but while still trying to act like UDP. The reality is if you want speed, 您必须在不良连接上丢弃数据包，或者找到一种自适应地调整源代码编码的方法. And that's what WebRTC does natively.

So there's something beautiful about that workflow. As WebRTC matures, 人们正在围绕它构建广播质量的工具，以真正完成贡献方面的工作，作为替代它的一种方式.