Video Streaming Technology

Video streaming technology powers how video content is delivered, viewed, and interacted with in real time or on demand. It combines multiple components  that ensure smooth playback, quality delivery, and reliable performance across different devices and networks. From ultra-low latency delivery to adaptive bitrate encoding, it supports a wide range of live streaming use cases. This section provides an overview of commonly used streaming protocol s, video and audio codecs, and container formats.

Video Streaming Protocol

Video streaming protocols define how video and audio data are transmitted from the source to the viewer over the internet or intranet. They determine the speed, reliability, and latency of delivery, enabling seamless playback across different devices and live streaming use cases .

WebRTC

Web Real-Time Communication standard that enables real-time video, audio, and data streaming with sub-250 ms latency directly between browsers, apps, devices.

SRT

Secure Reliable Transport protocol designed for real-time streaming with sub-250 ms latency, high-quality video delivery over unpredictable networks.

HLS

HTTP Live Streaming protocol by Apple that delivers adaptive bitrate video by segmenting streams into small media chunks. Latency between 2-10 seconds.

WHIP and WHEP

WebRTC-HTTP protocols for ingest (WHIP) and playback (WHEP) simplify live publishing and subscribing to media servers while achieving ultra-low sub-250 ms latency streaming.

RTMP

Real-Time Messaging Protocol widely used for live video ingestion to media servers and SDNs. Latency differs from sub-250 ms to 3 seconds.

Zixi

Proprietary transport protocol using ARQ and FEC for reliable, ultra-low latency video over the public internet, often used for broadcast contribution and distribution. Latency near 300 ms.

RTSP

Real-Time Streaming Protocol establishes and manages media sessions, commonly used with IP cameras and surveillance systems. Latency commonly under 300ms.

ETMP

Enhanced Real-Time Messaging Protocol built on RTMP to improve stability and efficiency for modern live streaming work-flows. Latency around 200-500 ms.

MOQ

Media over QUIC protocol enables real-time streaming with sub-250 ms latency and supports both live and non-real-time scenarios such as caching, time-shied playback, and VOD.

HESP

High Efficiency Streaming Protocol by Dolby reduces playback delay at startup and offers near-real-time performance, but its closed, licensed nature limits wide adoption. Latency under 400ms.

Read these blogs to learn how the protocols compare to one another :

RTMP vs RTSP

MOQ v s WebRTC

RTSP v s WebRTC

WebRTC vs RTMP

CMAF vs WebRTC

WebRTC vs HESP

LL-HLS vs WebRTC

Video Codecs

Video codecs compress and decompress raw video data to make it suitable for streaming and playback without sacrificing quality.

They balance file size, efficiency , and visual performance, ensuring smooth delivery for various resolutions and livestreaming use cases.

h.264 (AVC)

Advanced Video Coding standard offering a strong balance of compression efficiency, playback quality, and universal compatibility across devices and browsers.

h. 265 ( HEVC)

High Efficiency Video Coding standard provides up to 50% better compression efficiency than H.264, lowering bandwidth use while requiring more processing power and hardware support.

VP9

Open-source video codec developed by Google, providing compression efficiency similar to H.265 while maintaining broad browser and platform support.

AV1

Next-generation open, royalty-free codec developed by the Alliance for Open Media, offering superior compression and quality efficiency across modern devices.

Read these blogs to learn how the video codecs compare to one another

H.264 vs H.265 vs VP9

AV1 vs VP9 vs VP8

AV1 vs H.264

AV1 vs H.265

Audio Codecs

Audio codecs encode and decode sound data to efficiently transmit high-quality audio over a public or private connection. They reduce bandwidth usage while preserving clarity, making them es sential for smooth, synchronized sound in live streaming use cases .

Opus Codec

Versatile, open audio codec optimized for real-time applications, combining ultra-low sub-250 ms latency with high-quality voice and music streaming.

AAC

Advanced Audio Coding format delivering higher-quality sound than MP3 at lower bit-rates, combining efficient compression with broad support across streaming platforms.

Want to use these video streaming technologies in your app?

You can use Red5 live streaming products ranging from open-source solutions to advanced platforms like Red5 Pro, Red5 Cloud, and Red5 SDKs. Red5 is committed to providing reliable software and tools that help bring our vision to life through innovation and patented technology. Read our case studies to see what customers and partners say about working with Red5. Visit the blog and download whitepapers to explore industry trends and news, educational content, and release announcements about our products.

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FAQs

What is video streaming technology?

Video streaming technology is a method that delivers video content over public (internet) or private (intranet) connection. This technology powers everything from live broadcasts and video calls to on-demand platforms like YouTube or Netflix.

What technology is used for streaming?

Video streaming relies on a combination of key technologies including media servers, streaming protocols (such as WebRTC, RTMP, HLS, and SRT), and codecs like h.264, h.265, VP8, VP9, AV1, or Opus for compression. These technologies work together to send video and audio data efficiently across networks, ensuring smooth playback, low latency, and consistent quality for users on any device or connection speed.

How does video streaming work?

Video streaming works by breaking video files into small pieces and sending them over the internet or intranet using streaming protocols. As these packets reach the viewer’s device, a media player decodes and displays them almost instantly. The process continues in a loop, so playback feels continuous. Factors like bandwidth, latency, and codec efficiency affect how smoothly the stream plays.

What streaming technology lets you stream and watch video without an internet connection?

Air-gapped streaming allows video to be streamed and watched without any internet connection. It uses a closed, isolated network where data is transferred locally between devices for maximum security. This method is often used by government, defense, or enterprise organizations to ensure real-time video transmission while completely preventing external network access or data leaks.

What protocol is best for reliable streaming?

There’s no universal protocol that fits every streaming use case. Each protocol has strengths depending on your use case, requirements, and goals. For instance, WebRTC is ideal for real-time, ultra-low latency communication. HLS works best for scalability and adaptive playback. SRT ensures reliable delivery over unstable networks. 

What codec is best for reliable streaming?

There’s no single “best” codec for reliable streaming. It depends on your use case, requirements, and goals. For example, if you need 4K video quality, you should use the AV1 codec. But if your setup requires compatibility with SRT streaming, you’ll need to choose the H.264 or AVC codec.

What protocol and codec Red5 supports?

Red5, as a protocol-agnostic company, supports multiple streaming protocols like WebRTC, RTMP, RTSP, SRT, ERTMP, Zixi, HLS, WHIP and WHEP, and codecs including h.264, h.265, VP8, VP9, AV1, and Opus. You can also use multiple protocols in a single workflow. For example, combine WebRTC and HLS. This flexibility allows you to optimize for quality, efficiency, or compatibility based on your specific real-time streaming requirements.