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One free scan finds every outdated or missing driver and matches the right update for your exact hardware.Free scan · exact hardware matchRTMP, HLS and WebRTC do different jobs in many streaming workflows. RTMP commonly carries a feed from an encoder to a streaming platform; HLS distributes video to viewers over HTTP; and WebRTC supports real-time, interactive communication. A platform may accept a stream through one protocol and deliver it to viewers through another, so the right comparison starts with the role each protocol plays.
What each protocol is for
RTMP: sending a feed to a platform
RTMP is commonly used for contribution, or ingest: an encoder sends its live feed to a streaming platform. The platform can then repackage that incoming stream for viewer delivery. RTMP should not be mistaken for a standard browser playback format; browsers no longer play it natively, according to Wowza’s protocol documentation.
HLS: delivering video to viewers
HTTP Live Streaming (HLS) is a delivery method for live and on-demand audio and video. Apple describes HLS as working with ordinary web servers and content delivery networks (CDNs), with reliability and adaptation to changing network conditions among its design goals. Wowza describes HLS as segmented, adaptive-bitrate delivery over HTTP. That combination makes it a common fit when broad reach and playback that can adjust to a viewer’s connection matter. See Apple’s HLS overview and Wowza’s protocol documentation.
WebRTC: real-time communication
WebRTC is a set of technologies and browser APIs for real-time communication. Its media transport uses RTP: the IETF’s WebRTC media transport requirements specify RTP as required. WebRTC suits interactive situations where participants need to communicate with little delay, though actual performance depends on the implementation, network and service architecture. See the W3C WebRTC specification and IETF RFC 8834.
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RTMP vs. HLS vs. WebRTC at a glance
| Protocol | Typical role | Strength | Important qualification |
|---|---|---|---|
| RTMP | Encoder-to-platform contribution or ingest | Commonly used to send a live feed to a platform | Not native browser playback; a platform may repackage it for delivery to viewers. Source: Wowza protocol documentation. |
| HLS | Viewer delivery over HTTP | Uses web servers and CDNs; supports adaptation to network conditions and adaptive-bitrate delivery | Standard HLS and Low-Latency HLS are distinct variants. Source: Apple HLS overview; Wowza protocol documentation. |
| WebRTC | Browser-oriented real-time communication | Designed for interactive communication with low delay | Performance depends on implementation, network and service architecture. Source: W3C; IETF RFC 8834. |
Latency: useful comparisons, not guarantees
There is no single delay figure that applies to every deployment of a protocol. Latency varies with configuration and workflow, and vendor comparisons should be read in their stated context rather than as protocol-wide benchmarks.
- On its 2024 WebRTC workflows in Wowza Video page, Wowza describes HLS as typically 15–20 seconds and WebRTC as 500 milliseconds or less for the workflows discussed there.
- Wowza’s current WebRTC overview, accessed in 2026, compares WebRTC at under 500 ms, Low-Latency HLS and RTMP at 2–5 seconds, and HLS or MPEG-DASH at 10–30 seconds. These are Wowza’s comparison figures, not standardized or universal guarantees.
The distinction between standard HLS and Low-Latency HLS matters: do not apply a figure for one variant to the other. For broader operational considerations around streaming latency, see IETF RFC 9317.
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How to choose for your workflow
- Choose around interactivity first. If viewers need to talk, respond or participate in real time, WebRTC is the most natural fit among these three. If they primarily watch a one-way broadcast, HLS is often a better delivery model.
- Separate ingest from playback. If an encoder sends a feed to a platform, RTMP may be the ingest path; viewers may receive a repackaged stream using a different delivery protocol. Asking “Which protocol does the service use?” is incomplete unless you know whether the answer refers to ingest or playback.
- Consider reach and network variability. HLS’s HTTP and CDN delivery model and adaptive-bitrate approach suit broad viewer distribution across changing connection conditions.
- Check the actual player and device path. Confirm what the platform accepts from encoders and what its player supports on viewers’ target browsers and devices. Do not assume that an ingest protocol can also be played directly in a browser.
- Set an acceptable delay. A live event where a few seconds of delay is acceptable has different requirements from a conversation or remote interaction. Compare the chosen service’s implementation and configuration, not only the protocol label.
- Account for features and operations. Adaptive-bitrate behavior, CDN delivery, player support and the complexity of operating the media path can matter as much as nominal latency. The IETF’s RFC 9317 discusses operational considerations for streaming media.
What the protocol names do not tell you
The labels alone do not establish a service’s actual end-to-end delay, device compatibility, reliability or scale. Those depend on the implementation and delivery architecture. Likewise, choosing a protocol does not remove the need to confirm the full path: encoder, ingest endpoint, any repackaging or media server, delivery network, and viewer player.
RTMP, HLS and WebRTC are therefore not three interchangeable settings. They occupy different parts of many workflows: contribution, broad HTTP delivery and real-time communication. Some platforms support multiple protocols precisely because one can be appropriate for ingest while another is better for playback; check the platform’s documentation for its specific supported paths. Wowza’s vendor documentation describes its supported protocols.
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