HLS vs WebRTC: Which Streaming Protocol Should You Choose

HLS and WebRTC solve different problems, which is why asking "which one is better" gets you a less useful answer than asking "which one fits what I'm actually building." HLS trades latency for reach: it plays on almost anything and scales through ordinary CDN infrastructure. WebRTC trades reach for speed: it gets video to a viewer in a fraction of a second, but scaling it past a few hundred concurrent viewers takes real infrastructure investment. AIOZ Stream's on-demand delivery runs on HLS today, so this is also a look at why that choice makes sense for video on demand specifically, and where the tradeoff would flip if you needed something else.

TL;DR:

• Standard HLS runs 20-45 seconds of glass-to-glass latency but scales through any CDN and plays almost everywhere.
• WebRTC gets latency down to 100-500ms but needs specialized infrastructure (SFUs, TURN/STUN servers) past a few hundred viewers.
• Low-Latency HLS splits the difference at roughly 2-4 seconds while keeping HLS's CDN scalability.
• For on-demand video, where nobody is watching the same second as anyone else, the latency argument barely applies, which is why AIOZ Stream's VOD delivery uses HLS.

The core tradeoff: reach and cacheability vs latency

HLS breaks a stream into short segments and serves them over ordinary HTTP, which means any CDN can cache them and any device with a modern browser or media player can request them. That HTTP-and-cache model is also exactly why HLS is slow: a player has to buffer several segments deep before it can start smoothly, and every layer between origin and viewer adds its own delay. WebRTC skips the segment-and-cache model entirely and streams data over a direct, real-time transport connection, which is what gets latency down into the hundreds of milliseconds. The cost is that a direct connection doesn't cache the way an HTTP segment does, so serving a large audience means routing every viewer through a Selective Forwarding Unit or media server instead of a CDN edge node. Concretely, a Selective Forwarding Unit sits in the middle of a call, receiving each participant's stream once and forwarding copies out to everyone else, so the server's bandwidth cost scales with viewer count in a way a CDN edge caching one file for millions of requests never does. TURN and STUN servers solve a separate problem, helping two devices behind different home routers and firewalls actually find a path to each other in the first place, infrastructure a segment-and-cache model like HLS simply doesn't need because it's not trying to connect two peers directly.

How the actual numbers compare

Mux, a video infrastructure provider that publishes its own low-latency streaming guidance, puts standard HLS at 20-45 seconds of glass-to-glass latency before accounting for transcoding or CDN propagation, Low-Latency HLS at 2-4 seconds, and WebRTC at 100-500ms. Wowza's own comparison of the two protocols adds a scalability figure to that picture: WebRTC is workable for under 50 viewers without a custom CDN or dedicated video platform, and needs additional infrastructure to scale past a few hundred even with Wowza's own streaming engine. HLS, by contrast, plays back on almost any device or browser without a plugin, which is the reach side of the tradeoff.

Low-Latency HLS: keeping the CDN, cutting the delay

LL-HLS shortens HLS's segments and lets a player start requesting partial segments before they finish encoding, which is what pulls typical latency down from tens of seconds to roughly 2-4 seconds without giving up HTTP delivery or CDN caching. That's Mux's framing for it: low enough latency for the large majority of live use cases (sports, live commerce, news, events) without taking on WebRTC's operational burden of running signaling servers and TURN/STUN infrastructure just to get a stream to viewers. For the actual mechanics behind that latency drop, partial segments, preload hints, and blocking playlist reloads, see our dedicated LL-HLS explainer. And if you want the precise definition of "glass-to-glass latency" itself, camera to screen, and how to measure it yourself, that's covered in this companion piece.

Where AIOZ Stream fits today

AIOZ Stream's on-demand delivery runs on HLS, and that's a deliberate fit rather than a default: nobody watching a VOD title is watching the same second as anyone else, so the tens-of-seconds latency HLS trades for reach and CDN scalability costs a viewer nothing they'd notice. Low-latency, WebRTC-style delivery matters most for live, interactive use cases, and AIOZ Stream's live streaming feature has not shipped as of this article. Once it does, the HLS-versus-low-latency tradeoff described above becomes directly relevant to how that feature gets built, not just background protocol theory. It's also worth watching whether that feature lands closer to LL-HLS or closer to WebRTC once it ships: a live product built for large broadcast audiences (product launches, conference sessions, most sports) points toward LL-HLS's CDN-friendly economics, while one built around tight interaction, live auctions, competitive gaming overlays, real-time Q&A, points toward WebRTC's sub-second floor and the SFU infrastructure that comes with it.

How to actually choose

If you're serving on-demand video, this isn't really a live decision: HLS's latency cost doesn't apply, and its reach and CDN economics win by default. If you're building live streaming, the real question is how much latency your use case can tolerate. A large-scale broadcast where a few seconds of delay doesn't matter, a product launch stream, a conference talk, most sports and events, fits LL-HLS's 2-4 second range fine. An interactive format where viewers react to something happening in real time, live auctions, competitive gaming, a Q&A where a delayed question looks broken, needs WebRTC's sub-second latency and the infrastructure investment that comes with it. A useful test that cuts through most edge cases: if a delay of a few seconds would just look like normal broadcast lag to a viewer, LL-HLS covers it. If a delay of even one second would make two people's actions look out of sync with each other, that's WebRTC's territory, not LL-HLS's, regardless of how big or small the audience is. Audience size on its own is a weaker signal than it looks: a ten-person live auction still needs WebRTC's latency even though it's a tiny audience, while a million-viewer product launch stream is completely fine on LL-HLS despite the scale, because what determines the right protocol is how tightly viewer reaction has to sync with what's happening on screen, not how many people are watching.

The cost difference behind the scaling limit

The "WebRTC doesn't scale past a few hundred viewers on ordinary infrastructure" point is really a cost curve, not a hard wall. Every viewer connected through an SFU consumes real, dedicated server bandwidth for the duration of the stream, since that server is actively forwarding a copy of the stream to each one individually. Add a viewer to an HLS broadcast and the marginal cost is close to zero once the segment is already cached at a nearby CDN node, the same cached file just gets served again. Add a viewer to a WebRTC broadcast and the SFU takes on a new, ongoing forwarding cost for exactly that viewer, for as long as they stay connected. That's the structural reason WebRTC at scale means provisioning real infrastructure ahead of a large audience rather than just pointing more traffic at a CDN that already has the content cached.

Frequently Asked Questions

Does AIOZ Stream support WebRTC today?
Not for a shipped feature. AIOZ Stream's current on-demand delivery uses HLS. Low-latency delivery is tied to the live streaming feature, which has not shipped as of this article.

Why is HLS so much slower than WebRTC?
HLS breaks video into segments requested over HTTP, which requires buffering several segments deep before smooth playback and adds delay at every caching layer. WebRTC streams over a direct real-time connection instead, skipping the segment-and-buffer model entirely.

Is Low-Latency HLS just as fast as WebRTC?
No. LL-HLS gets down to roughly 2-4 seconds; WebRTC gets down to 100-500ms. LL-HLS is fast enough for most live broadcast use cases but not for latency-sensitive interaction like live auctions or competitive gaming.

Why can't WebRTC just scale through a CDN like HLS does?
WebRTC streams a direct connection rather than cacheable HTTP segments, so serving a large audience means routing viewers through Selective Forwarding Units or dedicated media servers instead of ordinary CDN edge caching.

Does the choice between HLS and WebRTC matter for video on demand?
Not much. The latency argument for WebRTC only matters when viewers are watching the same moment together in real time. VOD viewers aren't, so HLS's reach and CDN economics make it the practical choice regardless of the latency gap.

What does a Selective Forwarding Unit actually do?
It sits between WebRTC participants, receiving each person's stream once and forwarding it out to everyone else who needs it, since WebRTC connections don't cache the way HTTP segments do. That forwarding cost is why WebRTC needs dedicated infrastructure to scale past a few hundred concurrent viewers, unlike HLS, which any CDN edge node can serve from cache.

Related Reading

• OBS Studio Setup Guide for AIOZ Stream (the RTMP ingest path in practice)
• Live Streaming on AIOZ Stream: The Complete Guide: how AIOZ Stream's RTMP ingest and HLS delivery work end to end for live broadcasts
• WebRTC Mesh vs SFU vs MCU: the SFU/MCU topology layer that sits behind WebRTC's low-latency side of this tradeoff
• Widevine vs FairPlay vs PlayReady: the cbcs encryption requirement tied specifically to HLS delivery for FairPlay playback
• AIOZ Stream Explained: what happens between ingest and playback, including why on-demand delivery uses HLS today
• What Is AIOZ DePIN: how the underlying node network delivers that HLS traffic
• What Is Low-Latency HLS: the mechanics behind LL-HLS's 2-5 second latency
• Glass-to-Glass Latency Explained: what the term means and how to measure it yourself
• What Is QoE in Video Streaming, and Why It Matters: how streaming quality actually gets measured

References

• Low-Latency Live Streaming for Developers: LL-HLS, WebRTC, and CMAF Explained, Mux: glass-to-glass latency figures and the LL-HLS vs WebRTC operational tradeoff
• HLS vs. WebRTC: What to Know Before Choosing a Protocol, Wowza: device compatibility and WebRTC scalability limits