HLS and RTMP come up together in almost every streaming setup, which makes it easy to assume you have to pick one. In practice they are not rivals: RTMP is built for ingest (getting a stream from your encoder to a server), while HLS is built for delivery (getting that stream out to viewers). This guide explains what each protocol does, how they differ, how they work together in a real streaming pipeline, and where WebRTC fits for real-time, interactive video.
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HTTP Live Streaming (HLS) is a video delivery protocol created by Apple. It breaks a stream into small file segments and creates a playlist that tells the player how to find and play them in order. A player downloads the segments one after another over standard HTTP or HTTPS and reassembles them into smooth playback.
HLS's best-known strength is adaptive bitrate streaming: it offers each segment at several quality levels and the player switches between them based on the viewer's current bandwidth, which keeps playback steady and limits buffering. Because it runs on ordinary web servers and HTTP, it works across almost all modern devices and browsers without plugins, and it supports encryption and captions out of the box.
HLS is the default choice for delivering video to a large audience. Its main strengths:
The main trade-off:
RTMP (Real-Time Messaging Protocol) is a way to push live audio and video from an encoder to a streaming server over a persistent TCP connection. Introduced by Macromedia in 2002, it was originally used for direct playback in Flash. Flash is gone, so RTMP is no longer used to play video in the browser, but it remains a dependable way to get a stream into a server.
RTMP sends a single encoded stream with low delay, which makes it well suited to the first stage of a live workflow. It does not adjust quality on its own: adaptive bitrate is added later, when the stream is packaged into a delivery format such as HLS or DASH. Modern platforms generally use RTMPS (RTMP over TLS) so the ingest link is encrypted.
RTMP earns its place as an ingest protocol. Its main strengths:
Its main limitations:
The core difference is their job: RTMP gets a stream into a server, and HLS gets it out to viewers. The table below sums up how they compare.
| Feature | RTMP (or RTMPS) | HLS |
|---|---|---|
| Primary role | Ingest (encoder to server) | Delivery (server to viewers) |
| Transport | TCP | HTTP / HTTPS |
| Typical latency | 1 to 3 seconds | 6 to 30 seconds (3 to 6 with LL-HLS) |
| Adaptive bitrate (ABR) | No (added downstream) | Yes, native |
| Browser playback | No native playback | Native on Apple; hls.js elsewhere |
| Scalability | Moderate (ingest only) | Very high (CDN-based) |
| Security and DRM | RTMPS (TLS); no DRM | HTTPS plus DRM (FairPlay, Widevine) |
In short, RTMP wins on ingest latency and encoder support, while HLS wins on reach, adaptability and scale. They answer different questions, which is why most live setups use both.
In a typical live streaming pipeline, you don't choose one instead of the other. You combine them:
RTMP handles the first mile, HLS handles the last mile. Seeing them as two stages of one workflow, rather than competitors, is the key to designing a reliable stream.
If HLS is for large-scale delivery and RTMP is for ingest, WebRTC is for real-time interaction. It is built into modern browsers, needs no plugins, and delivers sub-second latency with two-way audio and video, along with built-in encryption.
That makes WebRTC the right tool for a different job: video calls, virtual classrooms, webinars with live Q&A and interactive streaming, where people need to talk back and forth without delay. It is not a drop-in replacement for one-way broadcast to millions, which is where HLS and its CDN model still lead. For a one-way audience at scale, HLS fed by RTMP ingest remains the standard; for interactive, low-latency communication, WebRTC is the better fit. (For a closer look at that trade-off, see our comparison of WebRTC and HLS.)
What WebRTC does well:
Where it is harder:
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HLS and RTMP are not competitors but partners: RTMP carries a stream into the server, and HLS carries it out to viewers, usually in the same workflow. If you are delivering on-demand or broadcast video to a wide audience, HLS is the standard; RTMP (or RTMPS) is how you get the stream there. And when your priority is real-time, two-way interaction rather than one-way broadcast, WebRTC is the protocol to reach for, which is exactly what Digital Samba is built on.
Neither is better, because they do different jobs. RTMP is built to ingest a stream from an encoder to a server with low latency, while HLS is built to deliver that stream to viewers at scale with adaptive quality. Most live setups use RTMP for ingest and HLS for delivery.
RTMP is still the most widely supported way to push a live stream from software like OBS or a hardware encoder into a server, with low latency and reliable TCP delivery. It is no longer used for browser playback, but it remains the standard for the ingest stage.
Yes, and most live streams do. The encoder sends an RTMP or RTMPS stream to a server, the server transcodes it into several bitrates, and viewers receive it as HLS through a CDN. RTMP handles the first mile and HLS the last mile.
RTMPS is RTMP sent over a TLS-encrypted connection. It gives the same low-latency ingest as RTMP while protecting the stream in transit, which is why most modern platforms require it.
It depends on the use case. HLS is better for one-way delivery to a large audience, where a few seconds of latency is acceptable. WebRTC is better for real-time, interactive video such as calls and webinars, where sub-second latency and two-way communication matter most.