webtc和低延迟流

了解更多关于流延迟的信息流媒体东部2022.

阅读这段录音的完整文本:

瑞安Jespersen: 当我们谈论低延迟时，我们谈论的是什么? 我们讨论的是贡献吗? Are we talking about all of us working together on content in real time? 或者我们讨论的是分布? 我认为有三个不同的部分是关键. 我们得到的一些问题, 至于贡献方, 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? 我认为你们经常看到的是, 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. 还有鸟狗, 很明显他们在做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, 然后通过WebRTC CDN进行分发.

然后是分布部分. 显然，我在这里推销的是WebRTC. I think WebRTC can be used end-to-end for contribution mixing and distribution, 保持本地协议端到端. I think the beautiful thing about WebRTC is that by default over UDP it prioritizes latency over anything else. 它宁愿丢包也不愿花更多的时间. 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, 但同时仍然试图像UDP一样行事. 现实是，如果你想要速度, you've gotta drop packets on bad connections or find a way to adaptively adapt the encoding at source. 这就是WebRTC本身所做的.

所以这个工作流程很漂亮. 随着WebRTC的成熟, people are building broadcast-quality tools around that to actually do the contribution side as a way to replace it.

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现在就订阅最新一期过去的问题

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