Video streaming protocols are essential for delivering smooth and reliable video playback across different devices and networks. This blog explores various video streaming protocols, including RTMP, HLS, MPEG-DASH, WebRTC, and SRT, comparing their latency, scalability, and compatibility. Learn how to choose the best video streaming protocol for your use case, whether for live streaming, on-demand content, or enterprise video delivery.
Video technology has come a long way since the first moving picture was created back in 1888. Today, you can share a video with millions of viewers across the world with just a couple of taps on your laptop or phone. On average, each person watches around 16 hours of video every week (a 52% increase over the last two years!).
But the ease with which we can now share videos belies the true complexity of streaming video. A fundamental part of being able to send and play video on your device is video streaming protocols.
Video streaming protocols make it possible to stream and deliver video on your device. They are essentially rules for sending and receiving data to enable a smooth playback experience. Understanding video streaming protocols is essential since they need to be compatible with both the input source and the endpoint.
To better understand how you can effectively use video, it’s important to understand how various protocols impact video streaming, so let’s get into it.
A video streaming protocol refers to a set of rules or standards that dictate how video data is transmitted from one device or system to another over the internet.
These protocols are essential for ensuring smooth, reliable video playback across various devices and networks.
Each video streaming protocol has its advantages and disadvantages. Depending on your use case—whether for live streaming, on-demand content, or enterprise video delivery—choosing the right protocol is key to delivering a seamless and high-quality video experience.
While we’ve discussed what a video streaming protocol is, it’s important to also clarify what it isn’t. There’s often confusion about the differences between video streaming protocols, codecs, and container formats, so let’s clear that up.
Codec vs. Streaming Protocol:
A codec is not a streaming protocol. While video streaming protocols control how video data is transmitted and streamed, a codec compresses video files to reduce their size. This makes it easier to store and transmit the video over networks.
Container Format vs. Streaming Protocol:
A container format is also different from a video streaming protocol. While streaming protocols define how data is transmitted, a container format is used to store that data.
In summary, video streaming protocols deal with the transmission of video data, while codecs reduce video file sizes and container formats store the video files. These components work together but serve different purposes in the video delivery process.
It’s important to consider what kind of protocol is supported by devices when choosing a video streaming protocol. Each protocol serves its purpose and varies in terms of factors, such as latency and compatibility.
Common video streaming protocols include:
Let’s take a look at each of these different video streaming protocols.
Created by Macromedia and later acquired by Adobe, this universally accepted protocol was designed for streaming live and on-demand videos using Adobe Flash applications.
This universally accepted protocol is designed to maintain low latency connections by transmitting video and audio data in small packets. RTMP is a protocol based on TCP that was designed with the purpose of maintaining low-latency connections.
However, this protocol was developed primarily to work with the Flash player. This video streaming protocol has various variants, such as:
Adobe’s announcement for ending support for the Flash player in 2020 has severely reduced its viability, though it is still used with an RTMP encoder; it is ingested and converted to another protocol for playback (usually HLS) for last-mile delivery or egress.
It is not recommended, however, since it is rapidly becoming obsolete.
Use Case Example: RTMP is commonly used for live streaming on social media platforms like Facebook and YouTube. These platforms rely on RTMP for quick video ingestion during live events like product launches, interviews, or gaming streams.
Its low latency makes it ideal for scenarios where audience interaction is required in real-time.
Example: RTMP is often used for streaming live events on social media platforms such as Facebook and YouTube, where immediate video delivery and interaction are crucial for engaging audiences in real time.
Why RTMP?: RTMP is perfect for environments where the video needs to be captured and broadcast with minimal delays, such as gaming live streams or interactive webinars, where real-time feedback is essential.
Learn more about RTMP and it's history in video streaming.
Another traditional, but lesser-known protocol, RTSP is slightly different from RTMP. A very low latency protocol, it’s mainly used as a remote control to pause, play or stop video sessions by commanding and controlling media servers.
This protocol is popular for communication and entertainment systems. Its low latency also makes it popular for surveillance systems, drone streaming and other IoT devices as well.
However, it’s much less popular than other protocols; while it’s still relatively prevalent among IP cameras, iOS and Android devices generally don’t offer players compatible with this protocol.
It also relies on the Real-Time Transport Protocol (RTP) and Real-Time Control Protocol (RTCP) for delivering the data.
Use Case Example: RTSP is often used in surveillance systems or IP cameras for streaming video feeds. Its ability to control playback (pause, fast forward, rewind) makes it an excellent choice for live monitoring and security surveillance.
However, due to its low scalability, it is less ideal for broader audience streaming, such as mass broadcasts or live events.
Example: RTSP is widely used for live streaming in security and surveillance systems, providing low-latency video feeds from IP cameras. It’s also used in drone streaming for live aerial views during events or missions.
Why RTSP?: RTSP is best suited for real-time, low-latency streaming in specific environments like surveillance, live security feeds, and remote control media servers, where playback control is needed.
Also known as Apple HLS, this protocol was developed in 2009 by Apple original for iOS devices. It has since become, however, the most widely popular video streaming protocol. HLS video streaming is compatible with a wide majority of devices and HTML5 players.
The death of Flash has made most users switch to HTML5 players, which is the primary reason for its increased popularity. Additionally, it is one of the most secure and scalable protocols out there.
It also supports adaptive bitrate streaming, so it automatically optimizes the stream according to the devices’ resolution and network condition. However, unlike traditional streaming protocols, it can have relatively high latency.
Use Case Example: HLS is the most widely used video streaming protocol, ideal for live sports streaming and news broadcasts.
It’s often used for large-scale events such as the Super Bowl or Olympics, where viewers access the stream on a variety of devices, including smartphones, tablets, smart TVs, and desktops.
Example: HLS works great for live sports streaming or online education platforms where a broad audience is expected to access the stream across multiple devices. HLS ensures that users on iOS, Android, or smart TVs have the best possible viewing experience regardless of their device or network speed.
Why HLS?: HLS supports adaptive bitrate streaming, allowing it to automatically adjust video quality based on the user’s device and network conditions. This makes it ideal for on-demand content and high-traffic events like webinars, corporate conferences, or sports events.
One of the newer streaming protocols, MPEG-DASH is a common alternative for HLS in the industry.
It was developed by the Moving Pictures Expert Group (MPEG), an international authority on digital audio and video standards, to deliver video and audio to devices over web servers using the DASH (Dynamic Adaptive Streaming over HTTP) protocol.
An open-source option, it can be customized to support any audio or video codec. It also provides adaptive bitrate streaming, but isn’t supported by Apple software, since Apple prefers prioritizing its own protocol.
Use Case Example: MPEG-DASH is gaining popularity for OTT services and on-demand streaming platforms. Companies like Netflix and Amazon Prime Video use MPEG-DASH for adaptive bitrate streaming to ensure a smooth viewing experience regardless of the user’s internet speed.
It is also used for enterprise video streaming, where different devices or geographies need to access video content.
Example: MPEG-DASH is often used for on-demand video services like Netflix or Amazon Prime Video to ensure smooth playback across a wide range of devices, from mobile phones to smart TVs. It’s especially popular for adaptive bitrate streaming, adjusting video quality based on the viewer’s internet connection.
Why MPEG-DASH?: MPEG-DASH is perfect for large-scale, video-on-demand streaming services where seamless, high-quality playback is essential, and compatibility with various codecs is needed.
Developed by Microsoft, this video streaming protocol was designed for applications with the Silverlight player, but is now compatible with a wider range of devices, including iOS.
Like the previously mentioned HTTP protocols, MSS also provides adaptive bitrate streaming and tools for protecting against piracy. This protocol hasn’t seen the same level of popularity as the other protocols, however.
HDS was Adobe’s answer for an adaptive bitrate streaming protocol that evolved as a successor to the RTMP protocol.
Already with a low latency, adding adaptive bitrate streaming only added to Adobe’s successive protocol. However, similar to RTMP, this protocol requires a Flash player and since Adobe has retired the Flash player, this protocol is well on its way to becoming obsolete as well.
WebRTC is one of the new latest video streaming protocols with the fastest video and audio streaming capabilities of any existing protocol today.
With near-instantaneous, this protocol is primarily used for peer-to-peer video and audio sharing between browsers. While considered the best video streaming protocol by some of today, this protocol is primarily designed for video conferencing and lacks scalability.
Use Case Example: WebRTC is ideal for video conferencing, webinars, or peer-to-peer communications, like those on Zoom or Google Meet.
It allows for low-latency video and audio streaming between users directly in the browser without requiring plugins, making it an excellent choice for business meetings and virtual collaboration.
Example: WebRTC is a go-to solution for video conferencing platforms like Zoom or Google Meet, where ultra-low latency is essential for seamless real-time communication.
Why WebRTC?: WebRTC is designed for real-time, browser-based communications that require instantaneous video and audio streaming, such as in online education, telehealth services, and remote team meetings.
This open-source protocol is capable of providing high quality video streaming regardless of the network conditions.
Created by the SRT Allliance (which includes prominent video technology experts such as Microsoft and Wowza), this video streaming protocol has been recognized as a competitive substitute to both RTMP and RTSP, SRT offers reliable live video streaming with low latency over suboptimal networks.
Moreover, this video streaming protocol is also codec-agnostic, meaning it can operate with any modern audio or video codecs. However, this protocol is still not widely supported yet due to being an emerging technology.
Use Case Example: SRT is gaining traction for live broadcasts in challenging network conditions, like satellite connections or rural areas.
It's used by news outlets and sports broadcasters who need to ensure a stable and high-quality connection, even when the network is unreliable or fluctuating.
Example: SRT is used in live broadcasting, especially in remote areas or situations where traditional networks may not provide a stable connection. It's particularly useful for sports events and news channels that rely on delivering high-quality video streams under tough conditions."
Why SRT?: SRT excels at reliable live streaming over unpredictable networks, making it ideal for enterprise broadcasts and field operations in real-time environments where the network can’t be guaranteed.
Selecting the right video streaming protocol is critical for ensuring smooth playback, minimizing buffering, and optimizing user experience.
Different protocols cater to specific streaming needs, and various factors influence which protocol is best suited for a particular application. Below, we explore the key considerations that impact protocol selection.
Latency refers to the time delay between when a video is captured and when it appears on a viewer’s screen. Depending on the use case, different latency levels are required:
Latency Performance by Protocol:
Key Considerations:
Not all video streaming protocols are compatible with every device and browser. Ensuring broad support across different platforms is critical for delivering a seamless experience to users.
Protocol Compatibility by Device Type:
Key Considerations:
Video streaming needs to adapt to different internet speeds and network conditions. Some protocols handle unstable networks better than others, ensuring uninterrupted playback.
Protocol Performance in Different Network Conditions:
Key Considerations:
When streaming to thousands or millions of viewers, scalability is a crucial factor. Some protocols are better suited for large-scale content delivery, while others excel in peer-to-peer interactions.
Scalability of Different Streaming Protocols:
Key Considerations:
When selecting a video streaming protocol, consider the following:
By analyzing latency, device compatibility, network conditions, and scalability, you can determine the best video streaming protocol to optimize video quality, performance, and viewer engagement.
Choosing the right video streaming protocol is essential for delivering seamless, high-quality video experiences across different devices and network conditions. The best protocol depends on latency requirements, device compatibility, network adaptability, and scalability needs.
By understanding the strengths and limitations of each protocol, organizations can optimize their streaming infrastructure to provide buffer-free playback, high-quality video, and global accessibility.
Understanding Video Streaming Protocols: Video streaming protocols define how video content is transmitted across networks to ensure smooth, uninterrupted playback. Protocols like RTMP, HLS, WebRTC, and SRT vary in latency, scalability, and device compatibility, making it essential to choose the right one based on your use case.
Latency and Quality: For real-time video communication, WebRTC and SRT offer ultra-low latency, making them ideal for live interactions. However, if quality is more important than latency, HLS and MPEG-DASH are preferred due to their scalability and adaptive bitrate capabilities.
Device and Network Compatibility: Choose streaming protocols that are compatible with the devices your audience uses. HLS is widely supported across platforms, including iOS, Android, and smart TVs, while MPEG-DASH is often preferred for Android and Windows devices. WebRTC excels in browser-based streaming for small groups but lacks scalability.
Scalability and Network Conditions: Protocols like HLS and MPEG-DASH are best suited for large-scale streaming as they work well with CDNs to distribute content. SRT ensures high-quality streaming even over unstable networks, making it a strong option for remote live streaming.
VIDIZMO as a Solution: For organizations requiring secure, scalable, and high-quality video streaming, VIDIZMO EnterpriseTube supports multiple protocols including RTMP, HLS, MPEG-DASH, and WebRTC, offering flexibility and enhanced performance.
VIDIZMO offers a Gartner-recognized end-to-end enterprise video platform that supports multiple streaming protocols, ensuring smooth live and on-demand video delivery across all devices.
With adaptive bitrate streaming, low-latency support, and AI-powered video management, VIDIZMO enables businesses to stream securely, scale effortlessly, and enhance audience engagement.
Start your free trial today and experience how VIDIZMO can optimize your video streaming for performance, security, and scalability.
Read our blog on the comparison of the top live video solutions.
What are video streaming protocols?
Video streaming protocols are a set of rules that define how video data is transmitted over the internet. They ensure smooth playback, low latency, and compatibility across devices. Common protocols include HLS, RTMP, WebRTC, and MPEG-DASH.
What is the best protocol for live streaming?
The best live streaming protocol depends on the use case. HLS and MPEG-DASH offer high-quality, adaptive bitrate streaming, while WebRTC and SRT provide ultra-low latency for real-time interactions. RTMP is still widely used for ingesting live streams before converting them to other formats.
How do video streaming protocols affect latency?
Different video streaming protocols have varying levels of latency. WebRTC offers near-instantaneous delivery, while HLS and MPEG-DASH prioritize quality over speed, resulting in higher latency. SRT and RTMP provide a balance between latency and reliability.
What is the difference between HLS and MPEG-DASH?
HLS (HTTP Live Streaming) is developed by Apple and widely supported across devices, including iOS. MPEG-DASH is an open-source alternative that supports multiple codecs but lacks native support on Apple devices. Both offer adaptive bitrate streaming for optimized playback.
Why is RTMP still used for video streaming?
Although RTMP (Real-Time Messaging Protocol) is outdated for playback, it remains popular for live stream ingestion. Many streaming platforms use RTMP encoders to send video data, which is then converted into HLS or DASH for broad compatibility.
What is adaptive bitrate streaming, and why is it important?
Adaptive bitrate streaming (ABR) dynamically adjusts video quality based on the viewer’s internet speed. This prevents buffering, ensures a smooth experience, and allows playback across different devices and network conditions. HLS and MPEG-DASH are the most common ABR protocols.
Which video streaming protocol is best for low-latency streaming?
For low-latency streaming, WebRTC is the fastest, offering real-time communication. SRT (Secure Reliable Transport) is another strong option for low-latency, high-quality video over unpredictable networks. LL-HLS (Low-Latency HLS) is an emerging solution improving HLS performance.
How does a CDN improve video streaming?
A Content Delivery Network (CDN) helps distribute video streams efficiently across multiple locations, reducing latency and buffering. CDNs cache content closer to viewers, ensuring faster load times and better performance for live and on-demand video streaming.
What factors should I consider when choosing a video streaming protocol?
When selecting a video streaming protocol, consider:
How does VIDIZMO support multiple video streaming protocols?
VIDIZMO EnterpriseTube supports multiple video streaming protocols including RTMP for ingestion and HLS for delivery, ensuring high-quality, scalable, and secure video streaming. It provides adaptive bitrate streaming, low-latency options, and AI-powered video management for enterprises.