Whether you’re working at a film studio, production company, or post house, supporting the creative team can be daunting—especially if you’re tasked with helping bring their vision to life in a collaborative environment where any technical or security snafus can bring your collective efforts to a screeching halt. 

When it comes to killing the momentum and excitement of a project, few obstacles hold a candle to the buffering delays and stuttering of live video streaming as you try to help your teammates remotely share their creativity, vision, and hard work with clients and/or collaborators. 

To save you the headache of figuring out the best way through this technical obstacle course, we compared the top low-latency video streaming protocols so you and your team can bring that vision to life—and avoid having your project look and sound like the modern equivalent of dial-up internet. 

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RTMP

Tenured production professionals are likely familiar with Real-Time Messaging Protocol (RTMP). Originally developed by Adobe, RTMP remotely transmits audio and video project data over the internet. 

With a persistent Transmission Control Protocol (TCP) connection that relays media in near real time, RTMP is often used for ingestion, transferring streams from encoders to servers like YouTube or Twitch. While primarily designed for live streaming, RTMP’s popularity has diminished as newer protocols emerged. 

Pros

• Generally stable and reliable for certain types of live streaming
• Boasts wide compatibility with encoders and streaming platforms
• Supports multiple audio and video codecs

Cons

• Relies on Flash for playback, which has become obsolete
• Not ideal for large-scale distribution or more modern devices
• Lacks adaptive bitrate streaming

Average latency

• 2–5 seconds

Best for

• Live streaming from encoders to streaming platforms
• Real-time applications requiring stable ingestion

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SRT

Secure Reliable Transport (SRT) is an open-source streaming protocol designed for delivering high-quality, low-latency video streams over unreliable or unpredictable networks. SRT utilizes advanced error correction and encryption, ensuring secure and reliable data transmission, even with challenging network conditions. 

Thanks to its highly adaptive design, SRT can optimize for latency and quality based on real-time feedback from the network, which can be crucial for intense projects and/or large group streams. Because of these features, SRT has become a popular choice for professional live streaming workflows.

Pros

• Powerful error correction for creating smooth and consistent streams, even with unstable bandwidths
• Features end-to-end AES (Advanced Encryption Standard) encryption to maximize security
• Open source and highly customizable

Cons

• Requires a more complex setup compared to HTTP-based protocols
• Limited native support in some consumer-grade devices
• May need additional infrastructure for large-scale distribution/streams

Average latency

• 2–5 seconds (though it’s adjustable based on the network conditions)

Best for

• Professional live streaming over unpredictable networks
• Secure transmission of high-quality video feeds, often required for sensitive and/or high-profile material
• Contribution feeds for broadcast and production workflows

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FTL

Anyone in game development over the last 10 years may be familiar with the Fast Transmission Layer (FTL) protocol developed by Microsoft and primarily used by Mixer, the company’s live streaming platform for video games. 

FTL was a popular choice for a half decade in the late 2010s, as its design could keep delays as low as 200 milliseconds or less, allowing real-time interactivity between streamers and viewers. Although it uses UDP (User Datagram Protocol) for fast data transfer, it requires robust network conditions to maintain quality and low latency. 

Pros

• Ultra-low latency (sub-200 milliseconds)
• Optimized for interactive streaming experiences
• Minimal delay for near-instant communication between the streamer and audience

Cons

• Requires robust and stable network conditions to reach ultra-low latency
• Has limited support outside of specific platforms like Mixer
• May sacrifice quality in poor or unstable network environments

Average latency

• Sub-200 milliseconds (with proper network capability and stability)

Best for

• Interactive live streaming (e.g., gaming, Q&A sessions, etc.)
• Applications requiring real-time viewer interaction

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HLS

Developed by Apple, HTTP Live Streaming (HLS) is one of the more widely used streaming protocols, with the company claiming a larger market share in the last decade. 

By dividing media into small segments and supporting adaptive bitrate streaming, HLS adjusts playback quality based on network conditions. Given its high compatibility, HLS has become an industry standard for on-demand and live streaming applications. However, though generally reliable and scalable, it tends to have higher latency than other protocols. 

Pros

• Broad compatibility between devices and platforms (cameras, computers, software, etc.)
• Supports adaptive bitrate streaming for varying or unstable network conditions
• Robust and scalable for large audiences and their viewing experience
• Uses Hypertext Transfer Protocol (HTTP) for easy integration with content delivery networks (CDNs)

Cons

• Higher latency (10–30 seconds), less than ideal for real-time applications
• Can be resource-intensive due to data packet segmenting and re-encoding
• Slower startup times compared to other protocols

Average latency

• 10–30 seconds

Best for

• On-demand video streaming
• Large-scale live streaming where instant feedback isn’t necessary or the focus is on a one-to-many broadcast for end users vs. an interactive environment
• Applications where latency is less critical

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MPEG-DASH

Like HLS, MPEG-Dynamic Adaptive Streaming over HTTP (MPEG-DASH) utilizes data segmentation and adaptive bit rates to deliver high-quality video streaming that matches the user’s network conditions. But unlike HLS, MPEG-DASH is platform-agnostic, making it highly versatile for a variety of devices and players. 

As the name suggests, MPEG-DASH also uses HTTP for transmission but supports more advanced features than HLS, such as multi-language audio and Digital Rights Management (DRM) integration. While that may sound ideal, the protocol’s adoption still tends to vary, largely due to differences in platform support.

Pros

• Platform-agnostic and codec-independent, making it one of the most versatile options
• Supports adaptive bitrate streaming, optimizing quality
• Advanced features like multi-language audio and DRM support
• Compatible with most modern devices and players

Cons

• Higher latency (10–30 seconds), making it unsuitable for real-time applications
• Less native support than HLS, requiring extra setup/development for compatibility
• Can be complicated to implement due to its wide customization options

Average latency

• 10–30 seconds

Best for

• On-demand streaming across a diverse range of devices
• Large-scale live streaming where instant feedback isn’t necessary or the focus is on a one-to-many broadcast for end users vs. an interactive environment
• Applications requiring a versatile, open standard for delivery

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WebRTC

Designed for ultra-low latency, Web Real-Time Communication (WebRTC) is an open-source protocol that facilitates direct communication between browsers and devices without needing plug-ins or other intermediaries. 

WebRTC’s focus on low latency makes it ideal for real-time video streaming and highly interactive applications, such as video conferencing and collaborative and/or feedback sessions, even with a larger number of users. 

While it requires stable network conditions to maintain its high standard of quality, WebRTC is quickly becoming an industry favorite, as it combines broad compatibility with one of the lowest latencies in the business. 

Pros

• Ultra-low latency streaming (under 500 milliseconds)
• Peer-to-peer architecture, reducing reliance on servers
• Supports real-time interactivity, including two-way audio and video, making it an ideal choice for collaborative environments where video quality is paramount
• Open standard with broad browser and device compatibility

Cons

• Sensitive to network instability, which can affect quality
• Limited scalability for large audiences, at least without additional infrastructure
• More complicated setup compared to HTTP-based protocols

Average latency

• Under 500 milliseconds

Best for

• Video conferencing and real-time communication
• Instantly interactive live streaming (gaming, collaborative sessions, education)
• Peer-to-peer data sharing and applications requiring immediate feedback

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Need ultra-low latency for creative collaboration?

Evercast is a video conferencing and video streaming tool that uses a combination of a custom WebRTC implementation and GPU rendering to achieve high quality and ultra low latency. Utilizing GPU rendering allows for increased performance when streaming 10-bit 4:4:4 color depth videos in real time up to 60fps, with an average latency below 100ms.

Evercast developed the Pixelweave™ library, a GPU video processing hub that lets video conversions happen on a secondary processing unit of the system, offloading computation from the CPU and increasing streaming performance on any modern device with no additional hardware needed. Moreover, everything works over the internet, avoiding proprietary networks and making it highly accessible.