In 2026, video is a default way people learn, buy, and communicate online. Multiple industry roundups report that over 90% of internet users watch online video weekly, which matches what most broadcasters see in practice: if your stream doesn’t start fast and play smoothly, viewers leave.

That’s where streaming protocols matter.

A streaming protocol (sometimes called a broadcast protocol or video stream protocol) is the set of rules that moves video/audio across the internet—from your encoder or camera to a platform, CDN, and player. Protocols don’t decide “quality” by themselves; they decide how media is transported, how latency behaves, and how reliably playback works at scale.

In this guide, we’ll cover:

• The core protocol types (ingest, contribution, delivery, real-time)
• The 7 most common streaming protocols used by professional broadcasters
• Hybrid workflows (because in 2026, most pipelines use more than one)
• What’s new: Low-Latency HLS, CMAF, WHIP/WHEP, and HTTP/3 trends
• Security essentials: encryption, tokenization, DRM, and common pitfalls

TL;DR : 

In 2026, most professional workflows are hybrid:

• RTMP or SRT for ingest/contribution (easy encoder support vs. more resilient transport)
• HLS or DASH for large-scale delivery (CDN-friendly ABR)
• WebRTC for ultra-low latency interactivity (sub-second)
• Low-latency HLS/CMAF sits in the middle (near-real-time without WebRTC complexity)
• For security: HTTPS + token auth + DRM for delivery, and SRT (AES) / WebRTC (DTLS-SRTP) for real-time paths.

Table of Contents

• The Basics of Streaming Protocols
• 7 Common Streaming Protocols
• Best Streaming Protocols by Industry Use Case
• Protocol Interoperability and Hybrid Workflows
• Protocol Innovations in 2026
• Streaming Protocols and Security: What You Need to Know
• Best Practices for Choosing a Streaming Protocol
• The Future of Streaming Protocols: AI, Automation, and Real-Time Adaptation
• Conclusion
• Glossary of Streaming Protocol Terms

The Basics of Streaming Protocols

Streaming protocols are the backbone of professional broadcasting in 2026, enabling seamless delivery of live and on-demand content across a rapidly evolving digital landscape. A streaming protocol, often referred to as a broadcast protocol, is a standardized method for transmitting media, typically video or audio, over the internet. Essentially, video streaming protocols break content into small “chunks” and send them from one device to another, defining how these chunks are reassembled into playable content on the receiving end.

To explain how the protocol works: the video is broken into small segments, known as video segments, which are then downloaded and played sequentially by the player. These video segments are assembled in real time to create a smooth viewing experience.

This process highlights a critical feature of streaming protocols: compatibility. Both the sending device and the viewer’s device must support the same protocol for the stream to function. For instance, if you’re broadcasting using MPEG-DASH but the viewer’s player doesn’t support it, the stream will fail. Standardization is therefore vital, ensuring reliability across devices and platforms. HLS works by converting videos into small segments delivered over HTTP, enabling adaptive bitrate streaming and smooth playback on the user’s device.

Six widely used protocols dominate professional broadcasting today:

• HTTP Live Streaming (HLS)
• Real-Time Messaging Protocol (RTMP)
• Secure Reliable Transport (SRT)
• Dynamic Adaptive Streaming over HTTP (MPEG-DASH)
• Microsoft Smooth Streaming (MSS)
• Web Real-Time Communication (WebRTC)

When content is delivered, video data is transmitted continuously from the server to the user’s device, allowing for seamless playback without requiring a full download.

Before going into these live streaming protocols, let’s clarify two common points of confusion: the difference between protocols and codecs, and protocols versus stream formats.

Live Streaming Protocol vs. Codec

• Codec = compress/decompress (e.g., H.264, HEVC, AV1)
• Protocol = transport rules (e.g., HLS, DASH, WebRTC, SRT)

Streaming Protocols vs. Video Streaming Formats

Another area of potential confusion is the video stream format, which refers to the “container” or “package” that holds the transmitted media. A stream format typically includes compressed video, compressed audio, and metadata like subtitles or timing information. 

This data is transported via a streaming protocol, which dictates how the content is organized within the chunks being streamed. In protocols like HLS, a playlist file (such as an M3U8 master or media playlist) instructs the player on which media segments are available, their order, and enables features like adaptive streaming. Common stream formats for video include MP4 (fragmented) and MPEG-TS, each designed to work seamlessly with specific live streaming protocols.

7 Common Streaming Protocols

Different video streaming protocols are used for different use cases. Certain protocols are better suited for some streaming setups and others. The best protocol for live streaming depends on the situation.

There are seven common streaming protocols that professional broadcasters should be familiar with. Here is a brief overview of each that can serve as a decision matrix to help you easily make the right choice.

Other protocols such as MPEG-DASH, HDS, RTMP, and Flash have also been used alongside HLS, each offering unique features and applications for adaptive bitrate streaming and live video delivery, though some (like Flash and HDS) have seen a decline in support.

Now, take some time to explore some of the background and technical requirements for the most popular protocols.

1. HTTP Live Streaming (HLS)

HLS remains the “plays almost everywhere” delivery standard—especially because of Apple ecosystem support.

How it works: playlists + segments over HTTP. For low latency, LL-HLS adds partial segments, preload hints, and blocking playlist reload behaviors. Apple documents these LL-HLS extensions directly.

Best for (2026):

• Large-scale live events, OTT delivery, VOD
• Broad device coverage (iOS, smart TVs, browsers via players)

Where HLS struggles:

True real-time experiences (sub-second) still favor WebRTC

LL-HLS adoption is much better than early years, but your player/CDN/origin must support the full chain.

Practical tip :
Replace “HLS works by converting videos…” repeats with one concise explanation + a bullet list of settings (segment duration, part duration, keyframe cadence).

2. Real-Time Messaging Protocol (RTMP)

Real-Time Messaging Protocol (RTMP) is a legacy streaming protocol originally developed by Macromedia (later acquired by Adobe) to deliver video content to the now-defunct Adobe Flash Player. At its peak, RTMP powered a vast share of online video, as Adobe Flash Player was installed on nearly 99% of desktop computers in Western markets. Because of their tight integration, RTMP and Flash were often confused as one and the same. However, it’s important to clarify: Flash is dead, but RTMP lives on.

While RTMP is no longer used for playback due to Flash’s obsolescence, it has found a second life in video streaming methods as a robust and widely supported ingest protocol. In this new role, RTMP is used to transmit video from an encoder to an online video platform (OVP), where the content is then repackaged—typically into HLS or DASH—for end-user delivery.

One of the most common streaming types today is RTMP ingest combined with HLS delivery. This hybrid model leverages RTMP’s efficiency for real-time encoding and HLS’s scalability for viewer playback. While HLS ingest is slowly gaining traction, its support across web streaming protocols and OVPs is still limited, reinforcing RTMP’s staying power.

Best for (2026):

• Encoder → platform ingest (OBS, hardware encoders, legacy workflows)
• Simple setup for live events/webinars

RTMP may no longer be the face of streaming, but behind the scenes, it’s still doing the heavy lifting. If you need a stream with low latency, where there is a minimal delay in the processing of the data, RTMP ingest is one of the best video streaming standards to use.

3. Secure Reliable Transport (SRT)

Secure Reliable Transport (SRT) is an innovative, open-source streaming protocol developed by Haivision—an established leader in the video streaming industry. Designed for secure, low-latency, and reliable transmission over unpredictable networks (like the public internet), SRT is quickly becoming a top contender among modern live streaming protocols.

Initially created to address the limitations of traditional streaming types like RTMP and HLS, SRT excels in environments where stability and performance are critical. SRT is already being embraced in protocol streaming workflows that demand excellence, whether it’s remote production or streaming data for broadcasters in high-stakes situations.

SRT is media-agnostic, meaning it supports any codec, streaming formats, and transport types. However, broad adoption has been gradual due to limited support from legacy streaming infrastructure. Many encoders, decoders, and web streaming protocols still require updates to integrate SRT natively.

Some of the key advantages of SRT include the following:

• Security: Built-in 128/256-bit AES encryption keeps content safe during transmission.
• Low Latency: Adaptive retransmission and error correction enable real-time performance, making SRT one of the best options for low-latency event streaming protocol needs.
• Resilience: Handles jitter, packet loss, and bandwidth fluctuations, ensuring smooth delivery, even on less-than-stable networks.

Best for (2026):

• Remote production
• Field contribution over public internet
• High-quality contribution to cloud media servers

SRT offers a future-proof solution to your live video needs for broadcasters, media companies, or developers looking to stay ahead of the curve. The broader ecosystem continues to catch up in terms of support. However, SRT is already setting the benchmark for what’s possible in the next generation of audio-video protocol design.

4. Microsoft Smooth Streaming (MSS)

Microsoft Smooth Streaming (MSS) was once a pioneering adaptive bitrate streaming protocol, but as of 2022, it is no longer in active use. That said, MSS holds an important place in the history of streaming protocols, and understanding its rise and fall is a valuable case study in how even major platforms can fade when foundational technologies become obsolete.

Launched in 2008 by Microsoft, MSS was developed to support adaptive streaming formats that could optimize bandwidth and reduce buffering. It was a significant step forward at the time, known for delivering smooth playback, cost efficiency, and improved quality for viewers on variable internet connections.

MSS was the stream protocol behind some major early streaming events. Notably, it powered NBC’s online coverage of the 2008 Summer Olympics and was integral to streaming on Xbox 360, Windows Phone 7, and other platforms. It also worked with Microsoft’s Silverlight plugin, a now-defunct framework once used for web-based video experiences.

The downfall of Microsoft Smooth Streaming was tied closely to the decline of Silverlight, which was officially discontinued in late 2021. Since MSS was deeply dependent on this proprietary plugin, it lost its relevance. The industry simply moved toward HTML5-based web streaming protocols and more modern, plug-in-free media protocols like MPEG-DASH and HLS.

Despite its eventual decline, MSS did support PlayReady DRM, making it a go-to option in its day for content protection and secure protocol streaming. Microsoft has since shifted its support toward MPEG-DASH, which is now among the most widely adopted streaming media protocols.

Supported Codecs and Formats:

• Video Codecs Supported: H.264, VC-1
• Audio Codecs Supported: AAC, WMA
• Transport/Package Format: MP4 fragments
• Segment Duration: 2–4 seconds

Playback Support: 

• Browsers with Silverlight plugin
• Xbox consoles
• Windows Phone
• iOS devices
• Windows-based computers
• Various Smart TVs

If you’re exploring older devices or maintaining legacy systems, MSS might still surface in your research. However, for anyone seeking modern compatibility, scalability, and performance, newer internet stream protocols like DASH, HLS, or SRT are far better suited for today’s digital landscape.

5. Dynamic Adaptive Streaming over HTTP (MPEG-DASH)

MPEG-DASH is the live-streaming protocol of the future.

MPEG-DASH is an open ABR standard used heavily in OTT and smart TV ecosystems. It commonly pairs with DRM(Widevine/PlayReady).

Best for (2026):

• OTT distribution (many smart TVs, Android ecosystems)
• ABR at scale, multi-DRM delivery

DASH still isn’t natively supported by iOS Safari in the same way HLS is, so many stacks run HLS + DASH for full reach.

6. WebRTC

WebRTC is the go-to for “feels truly live” experiences: auctions, two-way classrooms, betting-style interactivity, live shopping, instant reactions.

Best for (2026):

• Sub-second interactive streaming
• Browser-based real-time audio/video

WHIP/WHEP 

To reduce custom WebRTC plumbing, the industry is standardizing WebRTC ingest/egress over HTTP:

• WHIP is now an RFC (WebRTC-HTTP Ingestion Protocol).
• WHEP is still progressing as an IETF draft (status can change, but it’s not shown as an RFC in the sources we pulled).

Why this matters: WHIP/WHEP makes WebRTC easier to integrate into streaming services using standard HTTP semantics, lowering operational complexity.

7. RTSP (Real-Time Streaming Protocol)

RTSP is best explained as a session control protocol, commonly used with RTP for media transport in camera/monitoring systems. It’s still everywhere in:

• IP cameras and surveillance
• Private monitoring feeds
• Some low-latency enterprise video paths

RTSP endpoints are frequently misconfigured (default passwords, exposed ports). If you keep RTSP, strongly recommend network isolation + secure variants when available.

Best Streaming Protocols by Industry Use Case

Choosing the right streaming protocol isn’t just about technical specs. It’s about meeting the demands of specific industries and audiences. Different sectors have unique requirements for latency, scalability, quality, and interactivity. 

Here’s a breakdown of the best live streaming protocols by use case, along with the ideal streaming formats and methods for each scenario.

Virtual Events and Conferences

HLS, DASH, and RTMP (for ingest) are the recommended protocols for these events. Virtual events prioritize scalability, compatibility, and reliability. HLS, a widely supported HTTP streaming protocol, is ideal for reaching broad audiences across devices. Combine with RTMP for stream ingestion due to its simplicity and low setup overhead. Use cloud-based platforms with built-in streaming protocols in cloud computing for global reach.

Online Education and e-Learning

The recommended protocols here are WebRTC, RTMP, and HLS. Online learning requires real-time interactivity. WebRTC, a data streaming protocol, enables ultra-low latency two-way communication, perfect for virtual classrooms and live Q&A. HLS can serve as a backup for playback. Use web streaming protocols that support adaptive bitrate streaming for variable internet connections.

Sports and Esports Streaming

We recommend using SRT (Secure Reliable Transport), WebRTC, and DASH for high-definition video with minimal delay is crucial. SRT provides secure, low-latency streaming over unpredictable networks. For ultra-low latency, WebRTC excels, especially for esports with chat interaction. For high viewership events, combine multimedia streaming protocols with a robust CDN strategy.

Church Services and Religious Broadcasts

With many faith-based organizations streaming weekly services, the recommended protocols for them are RTMP, HLS, and MPEG-TS. RTMP is easy to implement and stream to platforms like YouTube or Facebook. HLS ensures reliable playback across devices. Pre-recorded sermons can be scheduled using streaming media protocols via automation tools.

E-commerce Live Demos

We recommend WebRTC, RTMP, and LL-HLS (Low Latency HLS) for real-time interaction that drives conversions in live shopping. WebRTC is the best protocol for streaming with instant feedback. For broader delivery, LL-HLS combines low latency with wide device support. Optimize your stream format and video compression to keep load times low on mobile.

Protocol Interoperability and Hybrid Workflows

In 2026, most pro stacks are multi-protocol:

Common hybrid workflows

Workflow A: Classic scalable live

• Encoder (RTMP) → Platform → HLS/DASH via CDN → HTML5 player

Workflow B: Resilient contribution + real-time viewing

• Camera/encoder (SRT) → Media server → WebRTC (or LL-HLS) → viewers

Workflow C: OTT delivery

• Contribution → Packaging (DASH + DRM, plus HLS for iOS) → CDN → device apps/TVs

Protocol matching by role

• Ingest/Contribution: RTMP, SRT, RTSP
• Delivery: HLS, MPEG-DASH (often CMAF/fMP4)
• Real-time interactivity: WebRTC (increasingly via WHIP/WHEP)

Below, you will find an overview of the protocol matching by role:

Protocol Innovations in 2026

With the significant advancements we are seeing in the streaming protocols, there are enhancements that have emerged to meet the expectations of real-time applications, OTT platforms, and streaming data for broadcasters. The demand for ultra-low latency, scalability, and compatibility with modern network infrastructures drives these. 

These innovations signal a shift in how streaming types are chosen, not just for video quality but for speed, interactivity, and scale. Let’s take a look at some of the most promising innovations that are reshaping live streaming protocols.

Low-Latency CMAF for HLS and DASH

The Low-Latency Common Media Application Format (LL-CMAF) is a key enabler of low-latency streaming via HLS and MPEG-DASH. Introduced to reduce latency while maintaining adaptive bitrate support, LL-CMAF slices video into smaller chunks that can be streamed before the segment is fully encoded.

By using chunked transfer encoding and smaller segment sizes, LL-CMAF reduces end-to-end latency to 3-7 seconds, compared to traditional HLS/DASH’s 10-30 seconds. This makes it ideal for live events where near-real-time delivery is critical, such as sports, auctions, gaming, or news broadcasts. 

LL-CMAF maintains compatibility with existing CDNs and browser-native HTML5 players. It enhances the scalability of live streaming protocols without the complexity of ultra-low latency tech like WebRTC. Its ability to package a single set of media files for both HLS and DASH stream formats enhances efficiency, reducing encoding costs and server footprints.

QUIC-Based Protocols: WHIP/WHEP for WebRTC over HTTP/3

QUIC, the foundation of HTTP/3, has spurred innovation in live streaming protocols with protocols like WebRTC-HTTP Ingestion Protocol (WHIP) and WebRTC-HTTP Egress Protocol (WHEP). These protocols standardize WebRTC’s ingest and egress processes, simplifying ultra-low-latency streaming over HTTP/3. 

WHIP enables high-quality video input from devices like webcams, while WHEP ensures seamless delivery to non-WebRTC clients, such as smart TVs. With QUIC’s faster connection setup and stream multiplexing, WHIP/WHEP achieve sub-second latency, making them perfect for interactive applications like video conferencing or live auctions. Their reliance on HTTP infrastructure enhances interoperability, positioning them as a top choice for scalable, low-latency streaming data for broadcasters.

High-Efficiency Streaming Protocol (HESP)

The High-Efficiency Streaming Protocol (HESP), developed by the HESP Alliance, is designed for ultra-low latency and optimized scalability. HESP combines the benefits of adaptive bitrate streaming with latency as low as 1-2 seconds, rivaling traditional broadcast TV. 

Unlike HLS/DASH, HESP uses a dual-stream approach, low-latency and high-quality streams, allowing viewers to switch seamlessly without buffering. Optimized for modern CDNs, HESP reduces bandwidth costs while supporting high-quality playback across devices. Its open-standard approach ensures compatibility with various media protocols. It’s one of the reasons it’s a versatile solution for large-scale live events, such as gaming, global concerts, sports betting, interactive shows, financial trading dashboards, etc.

With features like ABR support, server-side ad insertion (SSAI), and DRM compatibility, it’s far superior to WebRTC or LL-HLS and capable of reaching millions. HESP is positioned as a bridge between real-time interactivity and broadcast protocols, removing the trade-off between latency and reach.

These innovations in live streaming protocols reflect the industry’s push toward real-time, scalable, and efficient content delivery in 2026, catering to diverse use cases with unmatched performance.

Streaming Protocols and Security: What You Need to Know

Security is a top priority for organizations distributing premium content or user-sensitive data. Whether you’re streaming a live event, powering a webcam protocol, or delivering OTT video at scale, protecting your video streams from hijacking, piracy, and unauthorized access is mission-critical.

Below, we break down the security and compliance considerations you need to understand when choosing between today’s leading live streaming protocols and media protocols.

Core Security Measures in Streaming Protocols

When evaluating streaming formats and protocol streaming options, consider whether they support the following security features:

• Encryption (e.g., TLS, SRTP): Protects stream content from interception.
• Token Authentication: Ensures only authorized users access the stream.
• Digital Rights Management (DRM): Prevents unauthorized reproduction or redistribution of content.
• Firewall/NAT Traversal: Helps ensure safe delivery across networks.
• Vulnerability Exposure: Determines susceptibility to injection, hijacking, or packet sniffing.

Below, you can find a security comparison of popular streaming protocols.

Compliance Considerations

When handling user data or protected media (e.g., healthcare, finance, e-learning, or licensed content), compliance with regulations such as GDPR, HIPAA, or DMCA may require:

• End-to-end encryption (SSL/TLS or AES)
• Audit trails and secure logging
• Geo-blocking and IP whitelisting
• Multi-DRM and adaptive access policies

These compliance requirements often dictate which streaming protocols in cloud computing environments are chosen for streaming data for broadcasters and enterprise use.

Best Practices for Choosing a Streaming Protocol

Selecting the right protocol is about more than just speed or video quality. It’s a strategic decision that affects latency, compatibility, scalability, and security. As the number of streaming formats, methods, and devices continues to grow, understanding how to align your use case with the right protocol is essential.

These best practices will help you decide regardless of whether you’re building a global live streaming protocol pipeline or setting up a webcam protocol for secure enterprise use.

Match Protocol to Platform and Device Support

Protocol compatibility can make or break your stream’s accessibility. Make sure your chosen stream protocol fits your audience’s devices and playback environments.

Identify Your Use Case

Choosing the right streaming protocol begins with understanding your specific use case, as different streaming types serve different needs. If you’re delivering live content, such as real-time events, low-latency protocols like SRT, WebRTC, or HESP are ideal for minimizing delay and a smooth viewer experience. For on-demand (VOD) or globally distributed live streaming, protocols like HLS or MPEG-DASH are better suited due to their scalability and broad compatibility. 

Next, consider your audience. If your viewers are internal users, such as within a company or private network, you’ll want to prioritize firewall-friendly, low-latency streaming media protocols. For public-facing broadcasts, opt for protocols that support high scalability and CDN distribution, ensuring consistent performance across a wide geographic range. 

Finally, assess whether your stream requires interactivity. WebRTC is the go-to live protocol for features like live chat, Q&A, or video call-ins. It offers real-time, bi-directional communication directly within web browsers.

Test Under Real-World Conditions

Monitoring tools and test environments help ensure you’re delivering the best live format experience possible. Simulate various network conditions, devices, and geographies to:

• Validate protocol performance
• Identify fallback needs (e.g., switching from WebRTC to HLS)
• Ensure optimal stream format selection

There’s No One-Size-Fits-All

Every protocol, whether it’s RTMP, SRT, HLS, WebRTC, or MPEG-DASH, has strengths and tradeoffs. Evaluate them in terms of your goals, not trends alone. The best data streaming protocol is the one that balances quality, reach, interactivity, security, and cost.

The Future of Streaming Protocols: AI, Automation, and Real-Time Adaptation

Artificial intelligence is driving a new era of efficiency, quality, and responsiveness, whether we are talking about smarter encoding or adaptive delivery.

AI-based encoding is now streamlining workflows by automatically adjusting bitrate, resolution, and compression based on the content type and audience network conditions. This minimizes bandwidth usage while maximizing visual fidelity.

Dynamic protocol switching, guided by AI, enables platforms to shift between streaming protocols. It’s switching from SRT to HLS or WebRTC mid-stream based on real-time performance metrics such as latency, buffering, or device compatibility.

Additionally, AI-driven quality optimization tools can monitor stream health, predict viewer drop-off points, and proactively adjust streams to avoid playback issues. Combined with automation and cloud infrastructure, this allows broadcasters to deliver more resilient and responsive live and VOD experiences at scale.

FAQs

What is a streaming protocol?

A streaming protocol defines how video and audio are transmitted over the internet from an encoder or server to a viewer’s device in real time or near real time.

What is the best streaming protocol for live broadcasting in 2026?

There is no single best option. Most professional workflows use RTMP or SRT for ingest and HLS or MPEG-DASH for delivery, with WebRTC added for ultra-low-latency interaction.

What is the difference between RTMP and HLS?

RTMP is primarily used for ingesting live streams from encoders, while HLS is used for delivering streams at scale to viewers across browsers, mobile devices, and smart TVs.

Which streaming protocol has the lowest latency?

WebRTC offers the lowest latency (often under one second). SRT and Low-Latency HLS provide near-real-time performance with better scalability.

Is MPEG-DASH better than HLS?

MPEG-DASH and HLS offer similar adaptive bitrate capabilities. DASH is widely used on Android and smart TVs, while HLS remains essential for iOS and Apple devices.

Are streaming protocols secure?

Yes—when properly configured. Modern protocols support encryption, token authentication, and DRM, with WebRTC and SRT offering built-in security for real-time streams.

Conclusion

Streaming protocols continue to serve as the foundation of modern digital media delivery. They ensure seamless, high-quality content transmission across various platforms. Web streaming protocols like HLS and DASH, alongside more traditional broadcasting protocols such as RTSP, play a crucial role in facilitating the real-time video streaming so popular today. These and real-time video protocols support a wide range of video stream formats and codecs, ensuring high compatibility and efficiency. 

Understanding the capabilities and limitations of each video stream protocol—and choosing the right one for your specific application—is essential for broadcasters and content creators looking to optimize performance, scalability, and user experience. As digital media evolves, so too will these protocols, adapting to support emerging technologies like AI-powered encoding, low-latency delivery, and real-time interactivity.

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Glossary of Streaming Protocol Terms

• ABR (Adaptive Bitrate Streaming): A technique that adjusts video quality in real time based on a viewer’s internet speed, minimizing buffering and enhancing playback.
• CDN (Content Delivery Network): A distributed network of servers that deliver media content to users based on their geographic location to reduce latency and improve reliability.
• CMAF (Common Media Application Format): A media packaging format optimized for low-latency delivery, designed to work across HLS and MPEG-DASH using fragmented MP4 files.
• DRM (Digital Rights Management): A technology used to control how digital content is accessed, copied, or shared, often used in media protocols to prevent piracy.
• HESP (High-Efficiency Streaming Protocol): A low-latency streaming protocol designed for large-scale live events, offering sub-second latency and fast start times.
• HLS (HTTP Live Streaming): A widely used HTTP streaming protocol developed by Apple that breaks video into small, HTTP-delivered segments for reliable playback.
• HTTP3 / QUIC: A modern internet transport protocol (QUIC) built on UDP and used in HTTP3 for faster, more secure streaming over web protocols like WHIP/WHEP.
• Ingest: The process of bringing a video stream into a streaming platform from an encoder or camera. Common ingest protocols include RTMP and SRT.
• MPEG-DASH (Dynamic Adaptive Streaming over HTTP): An open-standard, codec-agnostic streaming format that supports ABR and is commonly used for on-demand and live streaming.
• RTMP (Real-Time Messaging Protocol): An older streaming protocol used for ingesting streams into video platforms; now often paired with HLS for delivery.
• RTSP (Real-Time Streaming Protocol): A network protocol used primarily in IP cameras and surveillance systems for low-latency, real-time video streaming.
• RTP (Real-Time Transport Protocol): A foundational protocol for delivering audio and video over IP networks, commonly used with RTSP and SIP.
• SRT (Secure Reliable Transport): An open-source live streaming protocol that delivers secure, low-latency streams over unreliable networks, popular for contribution workflows.
• Segment Duration: The length of each video segment or chunk in a streaming protocol. Shorter segments generally mean lower latency.
• Transcoding: The process of converting a video from one format, resolution, or bitrate to another—essential for ABR streaming and cross-device support.
• TS (Transport Stream): A container format used to deliver audio, video, and metadata over streaming systems, commonly seen in HLS and MPEG-DASH workflows.
• WebRTC (Web Real-Time Communication): A peer-to-peer protocol suite enabling low-latency, browser-based video/audio streaming, ideal for real-time communication like video calls.
• WHIP/WHEP: WebRTC-HTTP ingestion/protocols based on QUIC that streamline WebRTC workflows for streaming protocols in cloud computing environments.