Video: Is WebRTC Today's Best Real-World Option for Low-Latency Streaming Playback?
Learn more about streaming playback and meeting the challenges of low-latency delivery at Streaming Media's next event.
Read the complete transcript of this clip:
Robert Reinhardt: WebRTC is the most interesting flavor or real-world options for live streaming playback because it's really the replacement for most of the tech I'm talking about when you need low latency live streaming. And again, for a lot of the government broadcast work I do, for any kind of webinar formats that I'm doing, even if you have a real-time chat that's a text chat with participants in the live stream, you don't necessarily need low latency. It's okay, if someone's asking questions to a presenter or moderator, it can be delayed a little bit. It's not the end of the world. But if you're doing a wine auction, a cattle auction, a game show, you can't have 10-second delays between a question being asked and someone responding to it.
And so WebRTC has an option for reducing that latency and that's mainly because we've got access to things like UDP now. At a browser level or not a native application stack, we can have libraries that consume WebRTC, streams or data channels, whatever you want to call it, coming out of the server. And we can even do P2P, if that's what you want to do, where you just have a server get you to another person and then you just start transmitting bites between each other and you don't even need the server to be there anymore.
So there's lots of options with WebRTC. The problem I find with WebRTC is really just that adoption across browsers. It's getting better. Apple finally added it to Safari, and more importantly iOS-Safari, or mobile Safari last year. So that's adding to the mix now. We can finally start to do more in the browser space. And it supports AES encryption which is nice. I have a lot of requests like I'm working on a surveillance type of application for one of my clients where encryption is important. And if you're using RTMP or even RTSP, encrypting those bits with any kind of off-the-shelf SDK can be of a concern where you get it right away with WebRTC.
So that's WebRTC in a nutshell. I believe in it as the future. I believe that you can develop in it now. But again, for anyone that had anything legacy built, this is being built like ... when I say legacy Flash, they're building WebRTC on top of that, right? So they're still probably likely maintaining some kind of Flash application that's doing the live streaming for people who have those bits and people who have WebRTC bits, great, we'll do that.
There's a cost to being cutting-edge, and for low-latency live video streaming that involves learning WebRTC and accepting limited browser support.
A Stanford University research team has created an architecture called Salsify that might offer a better way to deliver video for real-time applications
Streaming Video Alliance's Jason Thibeault and Limelight's Charley Thomas address the question of whether WebRTC provides a viable solution for network latency issues in this panel from Live Streaming Summit.