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Scan for outdated or missing drivers - takes under a minuteDriver Scan →Repair Windows errors before they cause bigger problemsFix Now →Fix the driver behind crashes, sound loss and screen glitchesFind Drivers →RTMP, HLS, SRT, WebRTC, and MPEG-DASH do different jobs in a streaming workflow. HLS and DASH are chiefly associated with delivering adaptive video to viewers; WebRTC serves interactive real-time communications; SRT is a transport for moving live media between endpoints over challenging networks. RTMP is a name you may encounter in contribution or ingest workflows, but its exact behavior and current support depend on the implementation, so check the target platform’s documentation before relying on it.
What is the difference between RTMP, HLS, SRT, WebRTC, and DASH?
The key distinction is where each technology fits. Sending a live feed to an ingest endpoint is not the same task as distributing playback to a large audience, and neither is the same as a two-way video conversation. A platform may also accept one format at ingest and package or deliver the stream another way.
| Technology | Typical role | What it is useful for | Important qualification |
|---|---|---|---|
| HLS | HTTP-based viewer delivery | Live or prerecorded playback, adaptive quality, and distribution through web servers and CDNs | Latency depends on the playlist, segment or part settings, player buffering, and the rest of the delivery path. |
| MPEG-DASH | HTTP-based adaptive viewer delivery | Internet multimedia streaming using an adaptive presentation | Implementations and device support vary; the standard does not mean every service exposes identical features. |
| WebRTC | Real-time communications | Interactive audio and video where participants need to communicate with each other | End-to-end delay depends on the complete implementation and network path. |
| SRT | Live-media transport between contribution or distribution endpoints | Moving a feed across a network where packet loss, jitter, or bandwidth variation is a concern | Its retransmission and buffering can improve resilience, but the buffer adds transport delay. |
| RTMP | Often encountered in contribution or ingest discussions | Check whether a particular encoder and destination support it for the intended workflow. | The sources cited here do not establish detailed RTMP behavior, latency, security, codecs, or current platform support. |
These descriptions reflect the roles described in Apple’s HLS documentation, MPEG’s DASH standards information, IETF material on WebRTC and streaming operations, and Haivision’s SRT documentation. They are not a universal compatibility matrix for every service or device.
Where protocols fit in a live streaming workflow
A useful way to compare protocol names is to trace the video from its source to the viewer:
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- Capture and encode: A camera, screen capture, or prerecorded file becomes audio and video data in an encoder or playback system.
- Contribution or ingest: The source sends a feed to a receiving endpoint. SRT is documented for transport between contribution and distribution endpoints; RTMP is also a name encountered in ingest discussions, but check the destination’s current requirements rather than assuming support.
- Packaging and distribution: A service may prepare the stream for delivery over HTTP. HLS and MPEG-DASH are associated with this adaptive viewer-delivery stage.
- Playback or interaction: A viewer’s player receives and buffers a delivered stream. In a two-way session, WebRTC is the more relevant comparison because it is a real-time communications suite, not simply another CDN-oriented segmented playback format.
One service can bridge or convert between stages. As a result, asking “Which protocol does this platform use?” may have more than one answer: the encoder-to-ingest connection and the viewer’s playback format can differ.
How each protocol works and when it makes sense
HLS: adaptive delivery over HTTP
Apple describes HTTP Live Streaming (HLS) as sending audio and video over HTTP from an ordinary web server. It supports live broadcasts and prerecorded video, alternate bitrate variants, and switching between variants as connection conditions change. Optional encryption and authentication are also part of the described HLS capabilities. The use of web servers and CDNs makes HLS a viewer-delivery option for broad web and device distribution.
Apple’s authoring requirements for its platforms cover codecs, containers, playlists, security, and low-latency behavior. Its Low-Latency HLS material recommends a one-second Part Target Duration, while also requiring the part duration to account for expected client-to-server round-trip time. That is an authoring parameter—not a promise that a viewer will see the camera image one second after it is captured.
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MPEG-DASH: an adaptive HTTP streaming standard
MPEG-DASH is MPEG’s standard for multimedia streaming over the internet. At a practical level, compare it with HLS as an HTTP-based approach to adaptive delivery, not as a guarantee that the same player features, codecs, or service behavior will be available everywhere. Apple’s CMAF overview notes that segmented CMAF media can support both HLS and MPEG-DASH presentations; that does not make every DASH and HLS implementation interchangeable.
WebRTC: interactive real-time media
WebRTC is the relevant category to consider when people need to interact through audio or video, rather than simply watch a broadcast. IETF documents describe RTP as the media transport framework used in WebRTC and specify how WebRTC endpoints work with transports, NATs, relays, and firewalls. Those network arrangements are part of why implementation matters: the protocol family alone does not determine the end-to-end delay or guarantee that a particular network path will work.
SRT: resilient contribution transport
Haivision describes Secure Reliable Transport (SRT) as open-source transport technology intended for streaming over unpredictable networks. Its documented mechanisms detect network conditions, compensate for jitter and bandwidth fluctuation, retransmit missing packets, and support AES encryption. This makes SRT a candidate for contribution or distribution links where a little more buffer time can help a feed withstand network variation.
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That buffer is a trade-off. Retransmission can only help if there is time for a missing packet to arrive before the receiver needs it. Haivision’s documentation for SRT 1.5.4, updated in February 2026, gives four times the round-trip time (RTT) as a rule of thumb for SRT latency on a fairly good network with 0.1–0.2% packet loss and no significant burst loss. Treat that as guidance for the stated conditions, not a universal setting or an end-to-end latency estimate.
RTMP: verify the specific ingest path
RTMP appears in streaming contribution and ingest discussions, but the sources available here do not establish an authoritative RTMP specification or enough detail to make broad claims about its latency, codecs, security, or present-day platform support. If you are choosing an encoder setting, confirm the destination’s current official ingest documentation and match its stated requirements. Do not infer viewer playback format from the ingest protocol name.
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There is no single best protocol across every stage. Start with the job and the actual endpoint support, then weigh latency, resilience, audience delivery, compatibility, and operational complexity.
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For a large audience watching a broadcast
Consider HLS or DASH for viewer playback when the service and target devices support the intended implementation. Their adaptive delivery model can offer alternate quality levels as network conditions change. Confirm the platform’s requirements and player/device support rather than assuming the standards guarantee identical behavior across products.
For an interactive call or live conversation
Consider WebRTC when participants need real-time interaction. Check how the particular service handles endpoint connectivity, relays, firewalls, and the full media path; the name alone does not specify the delay users will experience.
For contribution over a variable or lossy link
Consider SRT where its retransmission and buffering mechanisms fit the link and the receiver supports it. Decide how much transport buffer the application can tolerate, and test with the network conditions you actually expect. A more resilient feed may require more delay.
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When a platform asks for RTMP
Use RTMP only when the selected platform and sending software document support for that specific workflow. Follow the destination’s current ingest instructions for the required settings. The protocol label does not, by itself, establish a codec, security arrangement, latency, or playback format.
Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.Why protocol latency is not viewer delay
Latency accumulates through the whole chain: capture, encoding, packaging or multiplexing, network transport, player buffering, decoding, and display. A number attached to one stage cannot be treated as camera-to-screen delay unless it measures that entire path under stated conditions.
Haivision’s SRT documentation defines SRT “latency” as the delay introduced by sending over the network. Its guidance distinguishes that transport delay from overall camera-to-screen delay, which also includes capture, encoding, multiplexing, splitting, decoding, and display. Similarly, Apple’s one-second Low-Latency HLS Part Target Duration is an authoring target with round-trip-time and buffering considerations, not a one-second viewing guarantee.
- Ask what the latency figure measures: transport, packaging, player buffer, or capture-to-display.
- Check the conditions attached to the figure, including network RTT, packet loss, and burst loss.
- For interactive use, test the complete route between participants rather than relying on a protocol’s advertised role.
- For broadcast delivery, balance a lower delay against enough buffering to avoid interruptions under real network conditions.
Compatibility and deployment checks before you commit
Protocol standards describe capabilities, but your actual workflow depends on the encoder, ingest endpoint, packaging service, player, and target devices. Before production, verify each part of that chain.
- Endpoint support: Confirm the sender and receiving service both support the protocol for the specific workflow—ingest, contribution, delivery, or interaction.
- Platform requirements: Use the destination’s current official ingest documentation for its required settings; support can vary by service and change over time.
- Player and device behavior: Check the intended browsers, apps, and devices. Apple’s HLS authoring rules are specific to Apple platforms, and other implementations may differ.
- Network conditions: Evaluate packet loss, jitter, bandwidth variation, round-trip time, and whether the route includes firewalls or relays.
- Latency budget: Decide how much delay the use case can accept before setting buffers or selecting a delivery approach.
- Security and configuration: Verify the service’s supported encryption, authentication, codec, packaging, and playlist requirements rather than assuming they follow automatically from a protocol name.
- Operational complexity: Account for configuration, monitoring, network traversal, packaging, and the consequences of buffering or retransmission.
Keeping a prerecorded YouTube stream live without managing protocol settings
If your actual goal is to keep uploaded videos playing as a 24/7 YouTube channel, that is a different problem from selecting a protocol for a live camera or interactive call. StreamNeo is a cloud service for looping uploaded videos to YouTube: upload a recording or make a playlist, add your YouTube stream key once, and go live. The cloud keeps the loop running without a computer, OBS, or home connection staying on. StreamNeo’s stated product information does not identify its underlying streaming protocol, so this is not a protocol compatibility claim.
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