Zero Latency in Live Video — Myth or Reality? What Is Happening in the World of Live Video

The phrase “zero latency” is often used in the world of live video like a magic spell—promising real-time performance with no delay whatsoever. But what does zero latency actually mean? Is it technically achievable? Or is it a myth, exaggerated by marketing departments? This article breaks down the concept, separates facts from fiction, and explores what’s truly happening in the evolving world of live video, streaming, and AV-over-IP.

To begin, it’s essential to understand what latency is. In the context of live video, latency is the time it takes for a signal to travel from the source—such as a camera lens—to the destination, typically a display or a viewer’s device. This “glass-to-glass” delay includes multiple components: sensor readout, video processing, compression, network transport, buffering, decoding, and rendering. Even the fastest systems in the world cannot escape the laws of physics and computational delays.

So when companies or systems promise “zero latency,” they often mean “perceived zero latency”—a delay low enough that human users cannot detect it. In practical terms, this typically means anything below 100 milliseconds. For most users, a delay of 30–80 ms feels instantaneous. But that’s not the same as actual zero latency, which is physically unachievable in digital systems.

In broadcast, esports, live auctions, and remote collaboration, latency is a crucial factor. A delay of even one or two seconds can cause disconnection between audio and video, missed cues, or awkward conversation delays. In gambling, financial trading, or sports commentary, real-time accuracy is essential. That’s why so much effort has gone into minimizing latency at every stage of the live video pipeline.

There are different “tiers” of latency, depending on the use case. Let’s define the practical ranges:

• Conversational latency falls below 150 ms. This is required for natural dialogue in video conferencing, virtual production, and remote interviews. Anything above 250–300 ms starts to cause noticeable conversational lag. Achieving sub-150 ms glass-to-glass delay requires fine-tuned control over every part of the AV chain, from uncompressed video capture to WebRTC or SRT transport protocols.
• Interactive latency ranges from 150 ms to about 1 second. This is common in social live streaming, gaming, and e-commerce streams where users send emojis, comments, or make purchases in real time. Here, responsiveness is vital, but it can tolerate slightly more delay than one-on-one conversation.
• Broadcast parity latency is usually between 3 to 7 seconds. This is the range most common in cable TV and traditional IPTV, and is acceptable for watching sports or news, where syncing to other viewers is important but interaction is minimal.
• Buffered or quality-first latency exceeds 8–10 seconds. This is where platforms prioritize high resolution, stable bitrate, and seamless playback—often in OTT streaming services like video-on-demand or high-quality event broadcasting.

Now, let’s examine the technologies that enable ultra-low latency or near-zero latency performance.

One of the most discussed real-time protocols is WebRTC. Originally developed for peer-to-peer communication, WebRTC enables sub-200 ms latencies in live video by eliminating segment-based buffering and allowing direct connections between browsers and video sources. However, WebRTC has limitations in terms of scaling and codec flexibility.

Another popular protocol is SRT (Secure Reliable Transport), which delivers high-quality, low-latency video over unpredictable networks by using ARQ (Automatic Repeat reQuest) mechanisms. In live production environments, SRT can achieve glass-to-glass delays of under 500 ms, depending on encoder performance and network stability.

RIST (Reliable Internet Stream Transport) offers similar capabilities, focusing on interoperability, minimal overhead, and compatibility with existing broadcast infrastructure.

On the HTTP-based streaming side, new-generation formats like Low-Latency HLS and Low-Latency DASH reduce traditional 6–30 second delays to 1–2 seconds. These protocols maintain scalability and compatibility with CDNs, but depend heavily on player and segment timing optimizations.

Meanwhile, hardware vendors have developed AV-over-IP technologies such as SDVoE (Software Defined Video over Ethernet), which claim “zero frame latency” in local networks. This typically means sub-frame delays (under 1 ms) across 10 GbE networks using uncompressed or lightly compressed video. In practice, it translates to about 5–20 ms of end-to-end delay—still remarkably low, though not absolute zero.

In enterprise AV, real-time telepresence systems now integrate high-speed codecs and direct switching fabrics to support near-instantaneous signal delivery. In broadcast, robotic camera systems, remote control rooms, and IP-based routers have made ultra-low-latency workflows more viable than ever before.
 

Trade-offs and engineering considerations

Reducing latency often means lowering video resolution, cutting compression efficiency, increasing bandwidth requirements, or accepting less error correction. When latency is pushed too low, the risk of dropped packets, jitter, and video artifacts increases—especially on unstable networks.

To balance these factors, engineers must carefully tune parameters like:

• Encoder settings (bitrate control, GOP structure, buffer sizes)
• Transport mode (TCP vs UDP, segment size, latency buffer)
• Network topology (wired vs wireless, edge caching, redundancy)
• Playback behavior (buffer length, decoder delay, retry logic)

Real-world applications reveal just how difficult true zero latency would be.

Even on a closed network, using direct SDI loopback and ultra-fast decoders, there’s still a measurable processing delay. In large-scale streaming platforms, delays are compounded by CDN edge nodes, adaptive bitrate packaging, and user device variability.

Nonetheless, expectations are rising. Viewers now compare live streams to phone calls or FaceTime—both of which operate on low-latency architectures. Users in sports betting, esports, and live commerce increasingly demand sub-second response. As a result, service providers, hardware manufacturers, and software vendors are racing to optimize latency without compromising quality or scalability.

So, is zero latency a myth?

Yes—if interpreted literally. No digital system can have zero delay due to the inherent time needed for processing and transmission.

But also no—if we define it as sub-100 ms delay that is imperceptible to the human eye. In this sense, “zero latency” becomes a relative concept, describing systems that deliver real-time performance within human tolerance thresholds.

This redefinition of “zero latency” is what drives innovation in protocols, codecs, and system design. AV professionals now focus on achieving “application-level real-time” rather than theoretical absolutes.

Common engineering questions

Here are some of the most frequent long-tail questions from engineers and AV architects:

• What protocols support sub-100 ms latency for global live video delivery?
• How can AV-over-IP systems achieve zero-frame latency without compromising quality?
• What is the trade-off between video resolution and transport delay in low-latency systems?
• How do mobile edge computing and 5G networks reduce glass-to-glass delay?
• Which codecs are most effective in ultra-low latency environments?
• What role does jitter buffering play in balancing reliability and responsiveness?

These questions reflect growing awareness among integrators, system designers, and streaming providers. Latency is no longer a backend issue—it’s a user experience issue, affecting engagement, monetization, and satisfaction.

As we move into a hybrid world of physical and digital interaction—remote events, live-streamed concerts, cloud-controlled production—the latency challenge becomes central to performance.

In conclusion, zero latency in live video is not an achievable physical goal, but it is a valuable design target. Striving for sub-100 ms delay pushes technological boundaries and enables new types of content interaction. Whether in AV-over-IP systems, broadcast-grade workflows, or interactive mobile apps, understanding and managing latency is now critical.

The future of live video will not be defined by reaching absolute zero, but by optimizing for the right latency tier—conversational, interactive, broadcast-grade, or quality-first—depending on the use case. The companies that master this balance will define the next generation of real-time communication.