RTMP and SRT are mainly used to send a live feed from an encoder to an ingest point; HLS is mainly used to deliver video from servers and CDNs to viewers. They are not three interchangeable ways to do the same job. Choose based on where the protocol sits in your workflow, what the endpoints support, network conditions, and the latency your full system can sustain.
Where each protocol fits in a live-video workflow
A typical live stream has two broad stages. First, a production system contributes video to a receiving service or gateway. Then the video is distributed to viewers. RTMP and SRT are commonly used in the contribution stage; HLS is an HTTP-based audience-delivery format. A system may use one protocol for contribution and another for delivery.
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- RTMP: a mature, TCP-based option still used for live-stream ingest when the receiving endpoint accepts it.
- SRT: a UDP-based transport for sending media between compatible encoders, gateways, and receivers, with packet recovery and a configurable latency buffer.
- HLS: an HTTP-based format for live or on-demand playback through web servers and CDNs, with support for adaptive-bitrate renditions.
RTMP: use it when the ingest endpoint expects it
RTMP was developed in the Flash era and remains in use for live-stream ingest. It uses TCP and retransmission behavior. That makes it a practical choice when a platform or production workflow supports RTMP and its codec requirements fit your feed. Haivision’s 2022 comparison notes limitations around newer codecs such as HEVC, but codec support depends on the specific implementation and endpoint; check the destination’s current ingest requirements rather than assuming every RTMP service accepts the same formats. Haivision’s RTMP-versus-SRT comparison
RTMP trade-offs
- Useful for: contribution to an ingest service that expects RTMP.
- Network behavior: TCP delivery and retransmission; actual performance also depends on the path and buffering.
- Check before using: destination support, codec compatibility, and the latency of the whole production-to-viewer chain.
SRT: choose it for resilient contribution over variable networks
SRT is an open-source video transport that runs over UDP. It uses packet acknowledgments and retransmission (ARQ) to recover lost data, along with a configurable latency buffer. On an unpredictable public-IP path, those recovery mechanisms can help maintain delivery. The buffer creates a trade-off: allowing time for recovery can add delay, and the actual amount depends on configuration and network conditions. SRT transports media; it does not, by itself, define the web player or viewer-facing delivery layer. Haivision SRT project documentation · SRT Internet-Draft
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SRT trade-offs
- Useful for: contribution between compatible encoders, gateways, and receivers when recovery over a variable network matters.
- Network behavior: UDP transport with ARQ recovery; the recovery buffer trades added delay for resilience.
- Check before using: support at both endpoints, agreed configuration, packet loss, round-trip time, and acceptable latency.
HLS: use it for audience delivery over HTTP and CDNs
HLS sends audio and video over HTTP and can use ordinary web servers and CDNs. It supports live and on-demand streams, alternate bitrate renditions, and switching between them as bandwidth changes. Apple also documents encryption and authentication support. Those delivery characteristics make HLS relevant when the goal is to serve viewers at scale using standard web infrastructure. Apple’s HTTP Live Streaming overview
Traditional HLS and Low-Latency HLS
Apple’s Low-Latency HLS documentation puts the trade-off plainly: “Historically, HLS has favored stream reliability over latency.” Low-Latency HLS extends the protocol to reduce delay while retaining scalability. It uses mechanisms such as partial segments, playlist delta updates, blocking playlist reload, preload hints, and rendition reports. Lower-delay operation requires compatible production and delivery infrastructure; labeling a stream HLS does not mean it is using Low-Latency HLS. Apple’s Low-Latency HLS overview
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HLS trade-offs
- Useful for: audience-facing playback through HTTP servers and CDNs.
- Network behavior: HTTP delivery and adaptive renditions, with CDN distribution where configured.
- Check before using: player, origin, CDN, packaging, and—if lower delay matters—support for Low-Latency HLS across the full chain.
Compare the protocols by workflow need
| Decision | RTMP | SRT | HLS |
|---|---|---|---|
| Typical role | Contribution or ingest where the destination accepts it | Contribution between compatible encoders, gateways, and receivers | Audience-facing delivery through HTTP servers and CDNs |
| Transport and delivery behavior | TCP-based delivery with retransmission | UDP with acknowledgments, ARQ recovery, and a configurable latency buffer | HTTP delivery with adaptive-bitrate renditions and CDN distribution |
| Latency considerations | Often used for relatively low-latency ingest, but encoding, servers, players, and buffers add delay | Recovery buffer adds configurable delay; no universal latency follows from the protocol name | Traditional HLS favors reliability; Low-Latency HLS can reduce delay with compatible infrastructure |
| Compatibility to verify | Ingest endpoint and codec support | SRT support and matching configuration at both endpoints | Player, origin/CDN, packaging, and Low-Latency HLS support where applicable |
How to choose for your workflow
- Identify the job. If you are sending a feed to an ingest service, compare contribution options such as RTMP and SRT. If you are deciding how viewers receive the stream, evaluate HLS and the delivery infrastructure.
- Confirm endpoint compatibility. Check the current destination or receiver requirements for protocol, codecs, and configuration. For SRT, verify support and matching settings at both ends.
- Set a latency target for the whole chain. Account for encoding, network travel, recovery buffers, servers, packaging, player behavior, and display. No protocol name guarantees a particular end-to-end delay.
- Assess network conditions and scale. For a variable public-IP contribution path, SRT is worth considering when packet recovery and a configurable buffer are useful. For broad delivery through web infrastructure and CDNs, HLS is the relevant layer.
- Test the actual route. Measure behavior with the intended encoder, network, receiver, packaging, and player. A protocol comparison cannot replace testing the complete workflow.
What latency comparisons can—and cannot—tell you
Haivision’s 2022 comparison reports that SRT was more than twice as fast as RTMP in its compared setup, and 5 to 12 times faster in tests using dedicated hardware encoding and decoding. The vendor says its measurements included encoding, travel to and from test destinations, decoding, and display/server/player buffering. These are results from Haivision’s particular test setups, not independent universal benchmarks or a prediction for a different workflow. Haivision’s comparison and test qualifications
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For your own system, compare like with like: use the same endpoints, content, network path, encoder settings, and playback conditions. Include the player and buffering in the measurement; otherwise, a transport-only number may not describe what viewers experience.
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Common selection mistakes
- Treating all three as substitutes: RTMP and SRT commonly carry a contribution feed; HLS commonly delivers playback to viewers.
- Assuming UDP means lower end-to-end latency: SRT’s recovery buffer can add delay, and the rest of the chain still matters.
- Assuming HLS always means high latency: Low-Latency HLS exists, but the production, server, and delivery chain must support its mechanisms.
- Choosing by protocol name alone: codec support, endpoint compatibility, network conditions, player behavior, and configuration determine whether a particular workflow works well.
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