One of the most significant advances of the streaming era has been the ability to move video across the internet reliably, at scale, and with increasingly low latency. Two protocols dominate that conversation: HTTP Live Streaming (HLS) and Web Real-Time Communication (WebRTC). They aren’t really competitors as much as tools built for opposite ends of the same problem. Most broadcasters end up using both.
A quick high level comparison:
|
HLS |
WebRTC |
|
|
Origin |
Apple |
Google-backed open web standard (W3C/IETF) |
|
Latency |
15-30s standard |
Sub-second (typically less than 500ms) |
|
Scalability |
Millions of viewers via CDN caching |
~300-500 viewers per SFU instance before scaling infra is needed |
|
Transport |
HTTP/TCP |
UDP (via ICE/STUN/TURN) |
|
Network Compatibility |
Passes through nearly any firewall or proxy |
Can be blocked by restrictive corporate firewalls without TURN relay |
|
Best for |
Broadcast, VOD, large live audiences |
Interactive live, video calls, low-latency publishing |
|
Video quality |
Optimized for adaptive bitrate and buffering resilience |
Comparable codecs (H.264/VP8/VP9/AV1). Tuned to favor low latency over buffering under poor network conditions |
What is HLS?
HLS breaks video into tiny segments delivered over standard HTTP, which is what makes it so easy to distribute at scale. CDNs can cache and serve it without specialized infrastructure. It supports adaptive bitrate streaming and will automatically adjust quality to match a viewer’s device and connection. It’s built into virtually every modern browser and Apple device by default.
The tradeoff is latency. Since HLS buffers segments to protect playback smoothness, standard HLS runs 15-30 seconds behind the actual live stream. It’s also playback only, meaning it’s intended for consumers to receive the stream and not publish a stream out.
What is WebRTC
WebRTC isn’t strictly a protocol. It’s an open-source project and web standard maintained through the W3C and IETF but functions as a protocol. It’s main defining trait and attribute is real-time latency, typically under 500 milliseconds. This is why it powers many video calls and interactive platforms like WhatsApp, Discord, and Slack, among other live auction, betting, and watch-party features.
WebRTC also supports publishing directly from a browser, not just playback. It uses the same modern codecs as HLS. The real tradeoff isn’t inherent video quality, it’s that WebRTC’s congestion control is tuned to protect latency. This means it will degrade resolution faster than HLS under poor network conditions. It also requires much more infrastructure to scale. Past a few hundred concurrent viewers per server and you will want a Selective Forwarding Unit (SFU) and NAT traversal (STUN/TURN), which restrictive enterprise firewalls can still easily block.
LL-HLS and the New Low-Latency Field
Apple introduced Low-Latency HLS in 2019, using smaller, partial segments to cut down latency from 15-30 seconds down to a realistic 2-5 seconds in production. All while keeping HLS’s CDN scalability. That’s closed most of the gap with WebRTC for many live use cases.
A few newer protocols also worth mentioning in 2026:
- SRT (Secure Reliable Transport) - commonly used for contribution feeds (camera to cloud) and not typically for end of pipeline or last mile delivery to viewers.
- DASH - HLS’s open, codec agnostic counterpart. This is common outside of the Apple ecosystem.
- HESP (High-Efficiency Streaming Protocol) - utilizes a dual stream approach to hit 1-2 second latency with adaptive bitrate and nearly instant channel switching. Positioned as broadcast style television.
- Media over QUIC (MoQ) - a newer protocol, IETF standard running oer HTTP/3’s QUIC transport and aims to deliver low latency and CDN-scale distribution in one protocol. Still relatively new on the scene, standards and library are actively being developed.
FAQ
Is WebRTC better than HLS?
Neither is strictly better and they are optimized for different goals. WebRTC wins on latency and interactibely while HLS wins on scale, reliability, and reach.
Can you use HLS and WebRTC together?
Yes, and it’s increasingly the default pattern. WebRTC for interactive front end, HLS for scaled delivery to the broader audience.
Does LL-HLS make WebRTC unnecessary?
Not for the interactive use cases. LL-HLS closes the latency gap for one-to-many broadcast, but WebRTC is still better fit for two way or highlyu interactive experiences like calls, auctions, co-streaming, etc.
Choosing the Right Protocol for Your Platform
If you’re building a broadcast, FAST channel, or VOD plus live platform, HLS will almost always be your foundation. It’s what Zype’s Content Delievery is built around, with adaptive bitrate adn global CDN reach out of the box. If your platform also needs real-time interactive, then layering WebRTC on top can be worthwhile.
Not sure which mix fits your audience and use case? Talk to our team about how Zype supports video distribution at scale.