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Clear out junk files and repair common Windows errorsFree Scan →Fix the driver behind crashes, sound loss and screen glitchesFind Drivers →RTMP and SRT are chiefly used to send live video from an encoder to another system (contribution or ingest); HLS is chiefly used to deliver streams to viewers over HTTP servers and CDNs. They are therefore not three interchangeable choices for the same job. Choose based on where the protocol sits in your workflow, what the endpoints support, and the latency and reliability trade-offs you need.
What each protocol does
RTMP: a widely used ingest option
RTMP is a mature protocol originally developed for Flash that remains in use for live-stream ingest. It runs over TCP, which includes retransmission behavior. Whether a particular RTMP workflow supports a codec such as HEVC depends on the encoder and receiving endpoint; check the destination’s current ingest and codec requirements rather than assuming compatibility. Haivision’s 2022 protocol comparison discusses RTMP’s history and limitations.
SRT: 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 from loss, with a configurable latency buffer. That buffer can improve delivery over an unpredictable connection, but it adds delay; the result depends on configuration and network conditions. SRT transports media between systems; it does not, by itself, provide the viewer-facing web playback layer. See the SRT project documentation and SRT draft.
HLS: HTTP delivery to viewers
HLS delivers audio and video over HTTP using ordinary web servers and CDNs. It supports live and on-demand video, multiple bitrate renditions, and switching between renditions as bandwidth changes. Apple also documents encryption and authentication support. HLS is generally the audience-delivery layer, not a direct substitute for a contribution link from an encoder. Apple’s HLS overview describes its delivery model.
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RTMP vs SRT vs HLS at a glance
| Decision point | RTMP | SRT | HLS |
|---|---|---|---|
| Typical role | Contribution or ingest when the destination accepts it | Contribution between encoders, gateways, and receivers | Viewer-facing delivery through HTTP servers and CDNs |
| Transport and network behavior | TCP-based delivery with retransmission; performance depends on the path and buffering | UDP with ARQ recovery and a configurable latency buffer | HTTP delivery; adaptive renditions and CDN distribution |
| Latency considerations | Often used for relatively low-latency ingest, but encoding, servers, players, and buffers add delay | Recovery buffer trades some delay for resilience; no universal latency is guaranteed | Traditional HLS favors reliability; Low-Latency HLS can reduce delay when the full chain supports it |
| Compatibility to verify | Ingest endpoint and codec support | SRT support and agreed configuration at both endpoints | Player, origin/CDN, packaging, and Low-Latency HLS support if needed |
How to choose for your workflow
Choose SRT for a contribution link across an unpredictable network
Consider SRT when sending a contribution feed over a variable public IP path and recovery from packet loss matters. Configure the latency buffer to suit the connection and the delay your workflow can tolerate. Verify both endpoints support SRT and agree on the necessary settings.
Choose RTMP when the ingest endpoint expects it
Use RTMP when the platform or production workflow calls for RTMP ingest and the endpoint supports the codec and settings you intend to send. Confirm compatibility with the destination before configuring the encoder.
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Choose HLS for broad web and CDN delivery
Use HLS when the goal is to distribute playback to an audience through standard web infrastructure and CDNs, including adaptive bitrate delivery. Consider Low-Latency HLS when lower delay matters, but only if the production, packaging, delivery, and playback systems all support the required behavior.
What latency figures do—and do not—tell you
A protocol label alone cannot promise an end-to-end delay. Encoding, network conditions, server processing, player behavior, and buffering all contribute. In a 2022 comparison, Haivision reported SRT as more than twice as fast as RTMP in its compared setup, and five to twelve times faster in tests using dedicated hardware encoding and decoding. Those vendor-reported measurements included encoding, travel to and from test destinations, decoding, and display/server/player buffering. They describe Haivision’s particular test setup, not a universal or independently established result. Read the Haivision comparison for its context.
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Low-Latency HLS is a mode of delivery, not a different workflow role
Apple says, “Historically, HLS has favored stream reliability over latency.” Low-Latency HLS extends HLS to reduce delay while retaining scalability. Apple documents mechanisms such as partial segments, playlist delta updates, blocking playlist reload, preload hints, and rendition reports; they require compatible production and delivery infrastructure. A player or CDN that lacks the necessary support can prevent the intended low-latency behavior. See Apple’s Low-Latency HLS overview.
Compatibility and planning checklist
- Map the workflow: decide whether you are moving video from an encoder to another system or delivering it to viewers.
- Check every endpoint: confirm protocol and codec support at the sender, receiver, platform, player, and any relevant packaging or CDN layer.
- Set a latency target: account for encoding, network travel, server processing, player buffering, and any SRT recovery buffer.
- Assess the network: packet loss and round-trip time affect contribution performance; test the actual path under realistic conditions.
- Plan for audience delivery: consider scale, adaptive bitrate needs, and whether your web delivery infrastructure supports the HLS features you intend to use.
- Keep the chain coherent: a contribution protocol and an audience-delivery protocol can both appear in one workflow; select each for its own stage.
Where StreamNeo fits: always-on YouTube playback
StreamNeo is a cloud service for keeping a YouTube channel live 24/7 from uploaded videos. It plays uploaded videos rather than broadcasting a live camera feed, and it streams to YouTube only. This is a managed way to keep a prerecorded-video YouTube stream running, not a replacement for choosing an ingest or audience-delivery protocol throughout a production chain. See StreamNeo for details.
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To start, upload a recording or build a playlist, add your YouTube stream key once, and go live. StreamNeo loops the video from the cloud, so your computer, OBS, and home connection do not have to stay on. Each slot includes one always-on stream, 24/7 looping and playlists, 10 GB of storage per slot pooled across active slots, automatic recovery if YouTube drops the stream, and support from the StreamNeo team. Uploaded video streams as made, up to 4K 60fps, with no re-encode and no quality tiers; all plans have the same product and differ only in duration.
The first day is free, with no card required, once per account. Billing options are:
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- Daily: $0.99 per day
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You can cancel any time. UPI and cards are accepted in India; card checkout is available worldwide. For five or more slots, contact support. Before streaming, check that your uploaded material complies with YouTube’s copyright and reused-content rules. Use the copyright safety checklist to review your plans.
Quick Recap
Start your free first day with StreamNeo.
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