Will WebRTC Replace RTMP End-to-End?
Learn more about WebRTC at Streaming Media East 2022.
Read the complete transcript of this clip:
Tim Siglin: Does RTMP still have legs, and where do RTMP and WebRTC coexist?
Ryan Jespersen: I think RTMP defines streaming media. And I think for a technology to live for almost 20 years in this day and age is unbelievable. RTMP is still the most widely used contribution protocol in the world and will be for at least another couple of years. The problem is, that's where it stops. It's only a contribution piece. And when we compare that to what we're trying to build now, we're trying to build a point-to-point contribution mixing and distribution protocol that never has to leave that protocol. It never has to be transmuxed. You can retain all the value point to point. A lot of the stuff that we want to build around these workflows is actually not even in the video or the audio--they're in the data channel.
The minute we have to go from one protocol to another, we lose all of that control. Just think of what we're trying to do with all these RTMP features. And then we have to pull them into HLS. So we have servers like Wowza and others that do all that translation to add that value and translate that value through those little translators. I think when it goes back to talking about SRT and NDI--SRT is a great protocol, but it's a contribution protocol. It never will be a distribution protocol. What we have to learn with RTMP point to point is there's one vendor at the center of it. And there was another vendo, Apple, that put an end into it by the iPhone from a distribution side. We need to recreate that, but we need to recreate that using internet web standards that are here for the foreseeable future.
And I don't think there's anything else that does that outside of WebRTC from contribution mixing and distribution. And I'd love to see there's a lot of questions around encode or maturity with web RTC. And it's, it's little by little, you know, we are getting a few software encoders now, now adopting that the minute we start getting a lot of hardware encoders built on WebRTC and standardizing the signaling protocol, the more you're going to see those point-to-point use cases that allow us to use SVC end-to-end, allow us to control the data channel and standardize the data channel end-to-end. That's the piece where I think it starts becoming a very compelling use case for WebRTC specifically., and truly replaces RTMP end-to-end.
We're trying to compare WebRTC to RTMP or HLS--RTMP has been around almost 20 years. So it has a maturity level that it's not quite fair to compare. The other side too, is RTMP was designed for broadcast and for streaming. WebRTC was originally designed for web use. Now, as it's maturing, it is becoming a broadcast contribution protocol. I have a ton of broadcasters using it for live contribution with WebRTC. So now, as more encoders at the client-side level start building in the maturity level, that's where we need the broadcast industry to start adopting it, because then we're gonna start being able to compare apples to apples between one protocol and the other.
Andy Howard and Tim Siglin discuss the importance of low latency in video conferencing and how popular video conferencing tools like Zoom, Teams, and WebEx use scalable video encoding to guarantee access and lower latency in this clip from their panel at Streaming Media Connect 2022.
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