In this post, we’re diving into a frequently asked question: what is low latency streaming and why does it matter? In the world of live television and video streaming, latency is a fundamental concept to understand. Latency can have a major impact not only on broadcast production workflows but also on the final viewing experience. While video processing technology has greatly advanced since the beginning of digital television and OTT (over the top) video, we’re still closing the latency gap with solutions that promise low or ultra-low latency streaming.
Read on for a deep dive into the concept of latency, what causes it, why it matters, as well as tips on how it can be minimized.
What is Video Latency?
Simply explained, video latency is the time it takes for live content to travel from one destination to another. Broadcasters often refer to this as “glass-to-glass” or “end-to-end” latency. These are all terms that describe the total amount of time it takes for a single frame of video to travel from the camera to the final display. This time or delay can vary widely for many reasons. Delays are typically between 15-30 seconds for online content (also referred to as OTT video streaming) and under 10 seconds for broadcast television. However, in the case of live video production, latency needs to be much lower — well under a second — before content goes online or on the air in order for the content to be useable for live production, bi-directional interviews, and monitoring.
Why Does Low Latency Streaming Matter?
While no absolute value defines low latency, it’s often considered as less than a few seconds, with ultra-low latency being under a second. Although some solutions may claim to support no latency, this is technically not possible, unless you manage to circumvent the laws of physics!
For a long time, studies suggested that the lowest perceptible limit for humans to correctly identify an image was around 100ms. However, more recent studies suggest that the fastest rate at which humans appear to be able to process incoming visual stimuli is as low as 13ms.
For the purposes of this post, we’re using the terms broadcast, low, and ultra-low latency to illustrate the differences for live video streaming.
While unreasonably high video latency presents a serious annoyance to users, especially when watching live sports, ultra-low latency is not always critical — it simply depends on your application. For certain use cases such as streaming previously recorded events, higher latency is perfectly acceptable especially if it results in better picture quality through robust prevention of packet loss. A short delay between broadcast production and playout may even be intentional to facilitate live subtitling, closed captioning, and prevent obscenities from airing.
In linear broadcast workflows, for example, the delay between the feed and the actual live feed is typically somewhere between three to seven seconds. For OTT workflows, video latency is typically 15 seconds to as much as a minute. In the case of live video production, especially when taking into account the increasingly decentralized workflow requirements of talent and staff working remotely, latency needs to be kept as low as possible, under half a second and lower.
When is Low Latency Streaming Critical?
Keeping latency as low as possible is a must for live video production and streaming. Whether you’re producing live sporting events, esports, or interviews, nothing kills the viewing experience like high latency. We’ve all watched a live broadcast on location with long, awkward pauses, or people talking over each other in interviews because of latency issues. Perhaps you’ve watched a hockey game online while your neighbor watches live over the air and you hear them celebrate the winning shot 10 seconds before you see it. Worse still, imagine watching election results and they appear on your Twitter feed before you even get to see it on your TV screen. In these cases, low latency ensures an optimal viewing experience with great interactivity and engagement.
The key to low latency video streaming within seconds is ultra-low latency video produced within milliseconds.
Use cases where low latency streaming is especially critical:
Live sporting events: Capturing all the camera angles from a remote venue and including commentators and interviews requires close coordination of ultra-low latency video streams.
Esports and gaming: Every millisecond matters when covering events with players dispersed across the globe. In this case, ultra-low latency video is critical.
Bi-directional interviews: To keep conversations fluid and natural, video streams need to be kept at very low latency for both the interviewer and interviewee so that the total return time is under half a second.
Real-time monitoring: With decentralized broadcast production increasingly the norm, remote production staff, including assistant directors, rely on ultra-low latency video to make quick adjustments seconds before content goes to air.
Security and surveillance: For ISR and public safety applications, low latency video along with metadata is critical for making split-second decisions.
Remote operations: For video equipment operators (with functions such as replay and graphics) to be able to work remotely, video streams over IP must be as close as possible to equipment monitors onsite.
Decentralized workflows and remote collaboration: With talent and staff dispersed across geographies, having access to ultra-low video production and monitoring feeds is needed for seamless real-time collaboration.
What Causes Video Latency?
The issue of video latency isn’t caused by how quickly a signal travels from A to B alone, but also the time it takes to process video from raw camera input to encoded then decoded video streams. Many factors can contribute to latency depending on your delivery chain and the number of video processing steps involved. While individually these delays might be minimal, cumulatively they can quickly add up. Some of the key contributors to video latency include:
Network type and speed: Whether it be by public internet, satellite, MPLS, or other type of IP network, the network you choose to use to transmit your video will impact both latency and quality. The speed of a network is defined by throughput or how many megabits or gigabits it can handle in a second as well as by the distance traveled. On an IP network, the total round-trip time (RTT) can be determined at any given moment by measuring how long it takes for a packet to travel back and forth to a specific destination by pinging an IP address.
Individual components in the streaming workflow: From the camera to video encoders, video decoders, production switch, and then to the final display, each of the individual components in streaming workflows create processing delays which contribute to latency in varying degrees. OTT latency, for example, is usually much higher than digital TV because video needs to go through additional steps such as ABR transcoding before being viewed on a device.
Streaming protocols and output formats: The choice of video protocol used for broadcast production contribution and the outgoing viewing device delivery formats used also affects video latency. Not all protocols are equal and the type of error correction used by the selected protocol to counter packet loss and jitter can also add to latency as well as firewall traversal.
There are several ways to minimize video latency without having to compromise picture quality. The first is to choose a hardware encoder and decoder combination engineered to keep latency as low as possible, even when using a standard internet connection. The latest generation of video encoders and video decoders can maintain low latency (under 50ms in some cases) and have enough processing power to use HEVC to compress video to extremely low bitrates (down to under 3 Mbps) all while maintaining high picture quality. Another key factor in achieving lower levels of latency is to select a video transport protocol that will deliver high-quality video at low latency over noisy public networks like the internet. Successfully streaming video over the internet without compromising picture quality requires some form of error correction as part of a streaming protocol to prevent packet loss. Different types of error correction will all introduce latency, but some do so more than others. The Secure Reliable Transport (SRT) open-source protocol leverages ARQ error correction to help prevent packet loss while introducing less latency than other approaches including FEC and RTMP, though SRT can also support FEC when needed. With over 600 leading broadcast technology providers now supporting the SRT protocol, it has become the industry standard for low latency streaming.
When looking at minimizing latency, it’s important to carefully consider the impact configuration of different components can have depending on the use case. As you can tell by now, there are always trade-offs.
Achieving a higher quality for the end-user usually means higher resolutions and frame rates and therefore higher bandwidth requirements. While new technology and advanced codecs strive to improve latency, finding the right balance will always be important.
Ultimately, the individual targeted use case will determine the best balance within this triangle of video encoding and streaming considerations. For applications where low latency streaming is critical such as video surveillance and ISR, picture quality can often be traded in favor of minimal latency. However, for use cases where pristine broadcast-quality video matters, latency can be increased slightly in order to support advanced video processing and error correction.
The SRT streaming protocol, unlike RTMP, is codec agnostic and can support HEVC video for high-quality content, including 4K UHD video, at low bitrates and low latency. By delivering the optimal combination of bandwidth efficiency, high picture quality, and low latency, viewers can enjoy a great live experience over any network – with no spoilers.
Haivision’s powerful Makito X4 video encoder and decoder pair offer broadcast-quality streaming at ultra-low latency. Fueled by the SRT low latency protocol, the pair is relied upon by organizations worldwide for ultra-low latency, pristine quality, and rock-solid reliability for live content streaming over IP networks, including the internet. In addition, by taking advantage of slice-based encoding of HEVC and H.264 video, the latest release of both the Makito X4 video encoder and decoder can reduce overall end-to-end latency by up to 25%.
For mobile contribution, low latency is also possible. With Haivision’s Pro460 mobile transmitter, broadcasters can still realize the benefits of mobile contribution and low latency thanks to mobile IP-based video technology. To meet the demands of modern broadcast, each Haivision Pro mobile transmitter also features low latency video returns and remote latency configuration for efficient and effective live video production from anywhere.
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