Today we’re diving into two super important protocols for live streaming—RTMP and SRT.
A lot of folks ask me, “Hey, does SRT actually use less bandwidth than RTMP?”
And honestly, it’s not a simple yes or no. So before we answer that, let’s take a step back and break down how these two protocols really work.
RTMP & TCP
RTMP is a protocol that sits on top of TCP. So, what’s TCP? Think of it like downloading a file in your browser: your browser says, “I want this file,” and the server keeps sending data packets. Every single packet has to be confirmed as received, and if one gets lost, it just keeps sending it again until the whole file is fully downloaded.
This works great for files, super reliable—but for live streaming, it gets a bit tricky. Why?
Because every frame in a live stream is new, happening in real time. If the network drops a packet or there’s a delay, RTMP tries to resend it while new frames are still coming in. The result? Your video can lag, stutter, or even drop frames.
Picture it like a delivery scenario: if the first few packages can’t get through, all the new ones have to wait in line, and everything slows down.
SRT & UDP
UDP is super simple: once you send a packet, that’s it—it doesn’t check if it arrived or in what order. That makes it really low-latency, perfect for real-time stuff, but yeah, sometimes frames get lost along the way.
So, where does SRT come in? SRT takes the best of both worlds—UDP’s low latency plus TCP’s reliability.
It has its own way of tracking which packets made it, which didn’t, and which need to be resent. But here’s the smart part: it won’t bother resending frames that are already outdated—for example, a frame from a second ago isn’t worth sending. This keeps your live stream truly real-time.
Think of it like this: TCP is like a live stream that insists on showing every single chat message on the screen, no matter if it’s old or new. All the messages pile up, taking up space, and the new ones have to wait in line—so the screen can end up lagging.
SRT, on the other hand, is like a smart chat system that only shows the latest messages and automatically drops the old ones. That way, what you see is real-time interaction, the stream stays smooth, and there’s no lag.
SRT works like this—it “knows what it’s sending” and handles the video smartly, keeping things real-time while cutting down on unnecessary data.
H.264 & H.265
When it comes to video quality, RTMP usually sticks with H.264, while SRT lets you use more efficient codecs like H.264 and H.265.
H.265 compresses way better, so you can get the same quality using less bandwidth. For instance, an RTMP stream might need around 6 Mbps to look good, but with SRT + H.265, you could get almost the same quality at just 2 Mbps. This really shows when you’re streaming remotely or the network isn’t perfect.
Bottom line: what makes SRT exciting is that it balances low latency with reliability, and it lets you use more efficient codecs.
That means better-looking streams at the same—or even lower—bandwidth. That’s why more and more streaming encoders and devices are now built to support SRT and H.265 right out of the box.
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As a product operation specialist at YoloLiv, Jennie is primarily responsible for managing the YoloLiv YouTube channel. This includes planning topics, writing scripts, filming, post-editing, and responding to comments, focusing on enhancing the live streaming experience and providing guidelines for beginners.
