Nothing screams chaos like three-letter acronyms fighting for dominance in the chaotic world of live streaming protocols: RTMP, HLS, and WebRTC. If you’ve ever asked yourself why your live video lags, buffers, or looks like it was shot on a potato — chances are, you’re dealing with the wrong protocol.
RTMP wants you to believe it’s still the backbone of live video, even though it was born in the same era as dial-up modems and MySpace profiles.
HLS swaggers in with Apple’s branding, acting like it’s the savior of the modern web, while quietly sneaking in a 30-second delay and pretending it’s “live.”
Then there’s WebRTC, the scrappy new kid, obsessed with shaving milliseconds off its latency, even if it melts your server to do it.
In this Article:
Welcome to the alphabet soup of live streaming protocols, where everyone claims to be the fastest, the smoothest, the most scalable — and no one bothers to tell you what any of it actually means for your viewers. Today, understanding the difference between these video streaming protocols isn’t optional — it’s table stakes for any content creator, brand, or live streaming company worth their bitrate.
In this blog, we’re not just comparing specs like latency and bitrate. We’re unraveling the personalities behind the protocols. We’ll tell you who lies, who delivers, and who just looks good on paper. And we’ll show you how OneStream Live plays protocol diplomat — blending RTMP’s reliability, HLS’s compatibility, and WebRTC’s real-time charm into a streaming setup that actually works.
Live streaming has never been more powerful — or more misunderstood. Let’s fix that.
What Are Live Streaming Protocols, And Why Do They Matter?
Streaming protocols are the set of rules that govern how live video data travels from your camera or encoder to viewers across the internet. In other words, they’re the behind-the-scenes carriers of your live video.
Some common streaming protocols in cloud computing and live video distribution include RTMP, HLS, and WebRTC (our focus here). Others you might hear about are RTSP, DASH, and SRT – each with its own niche. Ultimately, streaming protocols matter because they determine how quickly and smoothly your live video reaches your audience.
Fun fact: Live video is highly engaging – about 80% of people prefer watching live video over reading posts – so using the right protocol to deliver that video can make or break your stream!
What Is RTMP and How Does It Work For Live Video Streaming?
It’s 2025, and RTMP (real time streaming protocol) is still shambling through the internet like a well-dressed dinosaur that refuses to go extinct. This relic of the past is still running the show behind many modern live streaming protocols. It was developed in the late 1990s for Adobe Flash Player and became a standard method to stream live video online from an encoder to a server with low delay
RTMP works by maintaining a persistent TCP connection between the streamer (client) and the media server, enabling data to flow continuously in real time. This persistent connection gives RTMP its hallmark low latency of roughly 2–5 seconds – meaning viewers see the action only a few seconds behind real life.
Here’s how RTMP works in practice
Your encoding software (e.g., OBS Studio or Streamlabs) breaks the video into small packets and sends them over the internet to an RTMP server. The server (often the platform you’re streaming to, like YouTube or Twitch) then either redistributes the stream to viewers or converts it into another format for playback.
RTMP is efficient for ingesting live feeds – it’s still widely used by platforms for accepting live streams. For example, when you stream from OBS to YouTube or Facebook, you’re likely using an RTMPS URL (RTMP over a secure connection) as the ingest point.
Why was RTMP so popular?
Because RTMP is the blue-collar workhorse of modern video streaming protocols. It provided a relatively reliable, low-latency stream when bandwidth was limited.
It uses TCP (which resends lost packets) to ensure smooth delivery, and it interleaves audio/video data to keep them in sync. RTMP can carry video encoded with codecs like H.264 as well as audio (AAC, MP3, etc.), making it versatile.
Even though Flash Player is obsolete (so browsers no longer play RTMP streams directly), RTMP remains important as a transport protocol to send live video to platforms. Modern streaming workflows often use RTMP for getting the stream to the cloud, then another protocol (like HLS) for delivering to viewers.
OneStream Live & RTMP
OneStream Live fully supports RTMP. For instance, you can use OneStream Live’s External RTMP Encoder feature to send a real-time feed from any RTMP-based tool (like OBS, Zoom, or XSplit) into OneStream Live.
From there, OneStream Live will multistream it to all your destinations. In practical terms, OneStream Live acts as an RTMP server that receives your stream and then redistributes it.
It even provides features like unique stream keys, password-protected RTMP streams, and real-time analytics for your RTMP feed. In short, if you have a favorite broadcasting app, you can go live via RTMP to OneStream Live, and we’ll handle the rest – sending your stream to YouTube, Facebook, Twitch, or any custom RTMP server you choose.
Key Takeaway:
RTMP is an older protocol that still excels at low-latency ingest – getting your live video from an encoder to a platform in ~2-5 seconds. It’s not used directly by viewers anymore, but it’s the workhorse for sending live streams to services (including OneStream Live).
What Is HLS And How Does HLS Streaming Work?
HLS (HTTP Live Streaming) is a popular protocol for delivering live and on-demand video to viewers over the internet. Born in Apple’s walled garden in 2009, HTTP Live Streaming (HLS) strutted into the scene with a revolutionary idea: what if we stopped pretending video was a “stream” and just delivered it in tiny, digestible chunks (often 2 to 6 seconds each) and offers those segments over standard HTTP?
The result? A protocol that can scale to the moon and back — or at least to a million simultaneous viewers without having a panic attack!
An index file (called a playlist, usually .m3u8) tells the player the order of segments. This design allows HLS to leverage ordinary web servers and CDNs to distribute video at massive scale – if a million viewers request your stream, a CDN can handle it just like serving millions of small files.
How HLS works
As you stream, your encoder or server creates new segment files continuously (e.g., segment1.ts, segment2.ts, etc.). Viewers’ devices download these segments via HTTP and play them in sequence, reassembling the video stream.
Because it uses HTTP, HLS works seamlessly with browsers and devices – no special plugin needed. In fact, HLS became the “king” of cross-platform compatibility; it’s supported on iPhones, Android, smart TVs, basically any device with an HTML5 browser or streaming app.
One huge advantage of HLS is adaptive bitrate streaming (ABR). The server can provide multiple renditions of the stream (low, medium, high quality), and the HLS player will automatically pick the best one for each viewer’s internet speed.
This means fewer pauses and buffering, since someone on a slow connection can be served a lower-quality stream that won’t stall. RTMP, by contrast, can’t adapt on the fly – HLS’s segment approach makes it easier to switch quality at segment boundaries.
Latency in HLS
But here’s the catch: HLS is slow.
Like, “why is this goal celebration still buffering?” slow. Traditional HLS carries a latency tax — 15 to 30 seconds behind real life — because it waits to collect a few video chunks before playing anything. That makes it great for live concerts, video game events, or keynote addresses. But not so much for interactive livestreaming, where your audience wants their “🔥🔥🔥” comments seen in real time.
Thankfully, the engineers of the world gave us a fix: Low-Latency HLS (LL-HLS). By using shorter segments and other tweaks, LL-HLS can get latency down to ~2-5 seconds, making it competitive with near-real-time protocols. Major streaming platforms are adopting LL-HLS for more interactive live content.
OneStream Live & HLS
You don’t have to understand .m3u8 playlists to love HLS — that’s where OneStream Live steps in. When you create a Hosted Live Page or use our Embed Player, you’re already streaming with HLS — without even knowing it.
OneStream Live handles the transcoding, the segmentation, the adaptive bitrate, and the distribution in the background. You hit “Go Live,” and your viewers get buttery-smooth playback (whether it’s from Studio or an RTMP encoder) into an HLS stream for broad distribution.
We also support 24/7 HLS streaming on YouTube (continuous streams) and offer features like adaptive playback and ABR on our players so your audience gets a smooth experience. In short, OneStream Live marries the reliability of HLS delivery with the convenience of a cloud service, so you don’t have to set up any complex streaming servers.
Read Blog on How to Live Stream 24/7 on Your YouTube Channel with OneStream Live
Key Takeaway:
HLS is the protocol that powers most large-scale live video distribution on the web. It’s highly adaptive and compatible, able to reach many viewers with robust quality, though standard HLS comes with higher latency (15-30s).
New Low-Latency HLS can reduce delay to just a few seconds, bringing HLS closer to real-time. OneStream Live uses HLS to deliver your streams seamlessly to any device.
What is WebRTC, And How Does It Enable Real-Time Streaming?
Let’s talk about WebRTC for live streaming now, the protocol that behaves like it just downed three espressos and decided to rewrite the rules of live streaming protocols.
While RTMP and HLS are busy moving video around the internet like it’s a Sunday picnic — predictable, steady, painfully polite — WebRTC for live streaming shows up like an improv comic with a Red Bull and no filter.
This protocol wasn’t built for broadcasting to the masses. It was built for real-time communication, for one-to-one, one-to-few, and maybe — if you’re feeling brave — one-to-many (with the help of some serious back-end scaffolding).
You’ve already used it. Every time you’ve fired up a Zoom call, joined a Google Meet, or stared awkwardly into your webcam during a virtual job interview, WebRTC was there, streaming your high-res anxiety to the world with sub-second latency.
Learn How to Multistream your Zoom Meetings & Webinars for Wider Outreach
How WebRTC works
WebRTC is actually a collection of standards (APIs and protocols) that work together so that browsers (or mobile apps) can stream directly to each other. When you go live with WebRTC live streaming (say, via a WebRTC-based streaming studio), your browser uses UDP connections to send data, optimizing for real-time delivery.
It also uses clever techniques like adaptive bitrate and peer-to-peer networking. For example, WebRTC can dynamically adjust video quality on the fly to network conditions (similar to HLS ABR, but in real time).
It also has built-in encryption – streams are secured via SRTP (Secure RTP) and other protocols by defaul. One big advantage: no plutefox, Safari, Edge, etc., all support WebRTC natively.
WebRTC shines for ultra-low latency use cases, such as live auctions, interactive webinars, gaming streams with audience participation, or a coach giving real-time feedback to a student over live video. With WebRTC, you can achieve essentially a live conversation level of delay (well under a second) – far lower than standard HLS. This real-time capability is why WebRTC is used for things like video conferencing, online gaming communications, and any live streaming where interaction is key.
Read Blog on How Streaming Can Improve Remote Collaboration for Your Business
OneStream Live & WebRTC
OneStream Live’s Studio is powered by WebRTC technology. This means when you broadcast directly from your browser using OneStream Live Studio, you’re using WebRTC for real-time capture and streaming.
For example, you can invite guests onto your live stream – those guest feeds come in via WebRTC (no plugin needed, they just join via a link). You and your guests are essentially in a live WebRTC session, which allows interactive features like live chat, on-screen comments, and instantaneous Q&A.
Learn How to Create & Multistream your Broadcasts with OneStream Studio
OneStream Live then takes that WebRTC live stream and handles distributing it out to your viewers on platforms or your website. In essence, OneStream Live uses WebRTC live streaming to achieve ultra-low latency in the production stage (so you’re not lagging with your co-hosts or guests), and then it can convert or restream the output to RTMP destinations or HLS players for your audience.
This hybrid approach gives you the best of both worlds – real-time interactivity while broadcasting widely. OneStream Live Studio also has nifty tools like screen sharing and media injection (even allowing you to bring in an RTMP feed into the WebRTC live streaming studio), showcasing how flexible live streaming with WebRTC can be when combined with a cloud platform.
Key Takeaway:
WebRTC is the go-to solution for real-time streaming with latency often below 1 second. It enables interactive live video (video calls or streams with audience participation) directly in browsers without plugins.
While incredibly fast, it’s more complex to scale to huge audiences. OneStream Live uses WebRTC in its Studio to let creators produce live streams with minimal delay and high engagement.
RTMP vs. HLS vs. WebRTC: How Do These Live Streaming Protocols Compare?
Now that we’ve explained each protocol individually, let’s compare RTMP, HLS, and WebRTC side by side. Each has its strengths and ideal use cases. Here’s a quick overview in key areas:
1. Latency (Delay)
Latency is often the deciding factor for live streaming. WebRTC has the lowest latency – usually <1 second, practically real-time.
RTMP is also considered low latency, around 2-5 seconds in typical settings. HLS, in its standard form, has high latency, ~15-30 seconds. (However, with Low-Latency HLS, this can drop to 2-5 seconds, narrowing the gap.) If your stream is interactive, latency matters a lot. For general broadcasts, a bit of delay is acceptable.
2. Scalability
HLS is the most scalable – it was designed to serve massive audiences by using CDNs and standard web servers. Millions of viewers can watch an HLS stream if the infrastructure (CDN) is in place.
RTMP can be scaled via media servers, but it usually serves as an ingest; you wouldn’t send RTMP directly to millions of viewers (it would overwhelm a single server). Platforms convert RTMP to HLS/DASH to scale out.
WebRTC can be scaled using server networks (SFUs), but it’s not as plug-and-play scalable as HLS. Many WebRTC solutions cap out or require complex clustering beyond a certain number of participants (e.g., thousands). So for a one-to-many broadcast to a huge audience, HLS or a similar HTTP-based protocol is generally preferred.
3. Compatibility
HLS wins for device/browser compatibility among viewers – any modern device can play HLS streams (especially with HTML5 video players), and it’s the default for iOS devices.
WebRTC is supported in modern browsers (for both broadcasting and viewing), which means most desktop and mobile browsers can join a WebRTC stream without extra software – that’s a big plus for interactivity.
RTMP, on the other hand, is not supported for playback in browsers anymore (since Flash was retired). So, compatibility-wise: RTMP is for behind-the-scenes transport; HLS and WebRTC are for reaching viewers (each in different scenarios).
Also, HLS and WebRTC can more easily traverse firewalls and work over HTTPS. RTMP may be blocked in some strict corporate networks unless using RTMPS (SSL).
4. Adaptive Bitrate (Quality)
HLS supports adaptive bitrate streaming (ABR) out of the box – multiple quality levels and dynamic switching.
WebRTC isn’t segment-based, but it does adapt bitrate dynamically using codec features (it can lower resolution or frame rate on poor networks to avoid stalling).
RTMP does not have built-in ABR; it’s typically a single stream at one bitrate. If a viewer can’t handle that bitrate, they’ll experience buffering. This is why modern platforms ingest one RTMP stream and then transcode it to multiple bitrates for HLS delivery.
5. Video/Audio Quality
All three can deliver high-quality video, but there are differences in codec support. RTMP (because it’s older) works with H.264 video and AAC/MP3 audio primarily – it doesn’t support newer codecs like HEVC or VP9 in their typical usage.
HLS will support any codec your player supports (H.264, HEVC, VP9, AV1, etc. depending on device), which means it’s ready for 4K, HDR streaming with the right setup.
WebRTC’s common codecs are VP8, VP9, H.264 for video and Opus for audio – these provide great quality and efficiency for real-time, but broadcasting 4K via WebRTC is less common (it can be done, but requires powerful hardware and network).
In most real-world cases, if you need the highest resolution and bitrate (e.g., a 4K event stream), HLS (or DASH) is used; RTMP would choke on 4K (and RTMP’s old format doesn’t support the HEVC codec).
6. Security
WebRTC and HLS are secure by default. WebRTC mandates encryption (DTLS/SRTP), so all WebRTC streams are encrypted end-to-end.
HLS can use HTTPS and also offers AES-128 encryption for the segments, plus DRM for content protection.
RTMP was not encrypted originally; to secure it you use RTMPS, which wraps the RTMP stream in TLS (SSL) encryption. All major platforms now require RTMPS (YouTube, Facebook etc. enforce a secure connection for ingest).
In short, all three can be used securely, but WebRTC and HLS have the edge in being designed with modern security in mind.
7. Use Cases
- RTMP:
Best used for live stream ingestion from a broadcaster to a server or cloud. If you’re a creator using a tool like OBS, you’ll stream via RTMP to platforms.
It’s also good for low-latency contribution (e.g., sending a live feed from an event site to a studio production system). Some niche uses: internal broadcasts or linking systems that all understand RTMP. - HLS:
Ideal for broad public broadcasts and multi-platform streaming. If you need to deliver a stream to a wide audience with varied devices – like streaming a conference, church service, or gaming tournament to thousands of viewers – HLS is a solid choice.
It’s also the default for most live streaming CDN services and for video on demand streaming (e.g., most big streaming apps use HLS or DASH under the hood). - WebRTC:
Perfect for real-time interactive streaming. Use WebRTC for live scenarios where timing is crucial: webinars with Q&A, interactive classes, video consultations, live shopping events with audience interaction, or bringing remote guests into a live show.
It’s also behind video conferencing and peer-to-peer chat. WebRTC is great when you have smaller audiences or when you’re embedding an interactive player on a site for a specific event where everyone’s in near real-time sync.
Which Streaming Protocol Should You Use? A Brutally Honest Breakdown
The choice between RTMP vs WebRTC vs HLS comes down to your specific use case and priorities. Here are some questions and scenarios to guide your decision:
1. Is ultra-low latency (real-time interaction) crucial?
If you need people to virtually participate in real time – for instance, a live quiz show, an auction, or a video conference – go with WebRTC or similar real-time solutions. WebRTC was built for near-instant communication, whereas HLS’s normal latency would be too slow for back-and-forth interaction.
RTMP can handle low latency fairly well (a few seconds delay), so it’s sometimes used for semi-interactive streams, but true real-time apps favor WebRTC.
2. How large is your expected audience?
For a huge audience (thousands or millions), HLS is usually the safest bet because it scales using web servers and CDNs. If you expect viewers on all sorts of devices globally, HLS (or its cousin MPEG-DASH) will ensure everyone can tune in without custom software.
WebRTC can reach large audiences too, but it will require a robust infrastructure (and potentially more cost). RTMP alone should not be used to directly serve a large audience—it’s more for getting the stream into a platform, which then uses HLS/DASH to fan it out.
3. What are your viewers’ devices and connections?
If you need to cover a wide range of devices and network conditions, HLS is very adaptable (thanks to ABR) and compatible with virtually everything. Mobile viewers, smart TV apps, desktop browsers – all can handle an HLS stream.
WebRTC requires a compatible browser or app, but most users on updated browsers will be fine. If some viewers are on old hardware or behind strict networks, HLS over HTTP might succeed where peer-to-peer WebRTC fails.
Also, HLS can send multiple quality levels, accommodating both high-bandwidth and low-bandwidth users in one stream.
4. Are you using interactive elements (chat, Q&A, multiple presenters)?
If your live stream isn’t just one-way but involves guests, audience questions, or other live interactions, consider using WebRTC for the production side. For example, a live panel discussion with remote guests would benefit from a WebRTC-based studio (low delay between host and guests).
You could then output that stream via HLS to a large audience. If interaction is limited to text chat or comments, using HLS for video is fine, since the slight delay in video won’t ruin the experience – just keep in mind your responses will reach them half a minute later.
5. Do you need top-notch video quality (4K, HDR)?
If you’re streaming something like a major sports event or a concert in 4K, the HLS (or DASH) route with modern codecs will serve you best. WebRTC at 4K is possible but not common – it’s more taxing and many WebRTC implementations focus on 1080p or so.
RTMP can’t handle HEVC 4K since RTMP’s legacy format doesn’t support HEVC codec (YouTube, for example, requires HLS or DASH ingest to use HEVC for 4K streaming). So for the absolute best video fidelity to viewers, HLS is usually part of the pipeline.
6. What’s your streaming setup or platform?
Many streamers won’t explicitly “choose” a protocol – the platform does it for you. For example, when you use OBS with YouTube, you’re inherently choosing RTMP (because that’s what YouTube ingest uses). When you go live on Instagram via their app, you’re using whatever Instagram’s backend uses (actually RTMPS).
If you use a tool like OneStream Live, you have flexibility: you might broadcast via the OneStream Live Studio (WebRTC) or via an RTMP encoder, and OneStream Live will ensure the content is delivered appropriately (often via HLS or platform-native methods to viewers). So, use the protocol that fits into your workflow and let the platform optimize delivery.
Quick Tips
- If unsure, stream via RTMP to your platform – it’s the most universally accepted by streaming platforms for ingest. The platform will handle converting it for viewers (usually to HLS/DASH).
- Use HLS/DASH for building your own streaming service or website player where you anticipate a broad audience and need resilience and ABR.
- Use WebRTC for interactive experiences or private low-latency streams (like a small group seminar, live support chat with video, etc.).
Often, the best solution is hybrid. Modern live streaming solutions, including OneStream Live, use a mix of protocols: RTMP to ingest from various sources, WebRTC for real-time production and guest onboarding, and HLS to distribute to wide audiences. By combining them, you don’t have to compromise.
FAQs—Live Streaming Protocols
- RTMP (Real-Time Messaging Protocol) sends live video from your encoder to a streaming server.
- WebRTC is browser-based and offers sub-second latency, making it ideal for live interviews, gaming, or anything interactive.
- HLS (HTTP Live Streaming) is best for delivering streams to viewers across devices. It uses adaptive bitrate but has higher latency (10–30 seconds).
It depends on your goal:
- RTMP is better for live ingesting, such as sending video from OBS or encoders to your server.
- HLS is better for delivery, playing the stream smoothly across mobile, desktop, or smart TVs.
- WebRTC allows ultra-low latency (under 1 second) and runs directly in browsers, so no plugins or players are needed.
- RTMPs are just the secure version of RTMP (encrypted over the Internet used to transmit streams to servers safely.
No one-size-fits-all answer.
- WebRTC is best for real-time interaction.
- RTMP/RTMPs are best for sending streams to platforms or servers.
- HLS is best for scalable, device-friendly playback.
Use them based on your needs — most modern platforms combine all three for optimal performance.
The most common setup uses:
- RTMP is used to send the stream from the encoder to the server,
- Then, HLS or WebRTC is used to deliver the stream to viewers.
This hybrid approach balances latency, scalability, and compatibility — exactly what platforms like OneStream Live do behind the scenes.
Final Frame: Making Peace with Live Streaming Protocols
Here’s the bottom line: there’s no universal best when it comes to live streaming protocols. Each one — RTMP, HLS, and WebRTC — plays its part depending on what you need.
The smart move? Don’t pick one. Use them all — where they make sense.
OneStream Live does exactly that. It doesn’t make you choose between interactivity, quality, or reliability. This kind of setup isn’t theoretical. It’s proven.
We’ve built for creators, churches, businesses, and broadcasters. We know what breaks, what scales, and what your audience actually sees.
So before you obsess over which protocol is “right,” zoom out. What kind of experience are you trying to create? Who’s watching — and from where? What matters more: instant interaction or pristine playback?
Then build around that. Or let us do it for you.
So, what should guide your choice?
- Need real-time interaction? WebRTC has you covered.
- Expecting global viewership on diverse devices? Let HLS take the wheel.
- Broadcasting from OBS or any encoder? RTMP is your launchpad.
- Want it all? OneStream Live delivers it — without compromise.
Ready to Stream Smarter?
You’ve got the map. Now it’s time to hit “Go Live.” Sign up for free on OneStream Live, and let your content reach farther, faster, and better. No protocol confusion. Just performance.
OneStream Live is a cloud-based live streaming solution to create, schedule, and multistream professional-looking live streams across 45+ social media platforms and the web simultaneously. For content-related queries and feedback, write to us at [email protected]. You’re also welcome to Write for Us!
