23-05-2025
Bridging the Latency Gap with Broadcast in a Streaming World
Even with advancements in delivery platforms, it's not uncommon for digital audiences to experience delays of tens of seconds or more compared to traditional broadcast of the same content.
As media companies in the Middle East and beyond navigate the transition from traditional broadcast to IP- and cloud-based operations, one challenge continues to stand out: latency. In the world of live production and streaming, timing is everything. Delay can create a disconnect between the content and the audience, affecting engagement, brand trust and ultimately monetisation. So whether it's sports, news or interactive programming, latency now plays a central role in determining the quality of experience for streaming viewers and the value of the content they're consuming.
Traditional television broadcast has long been the benchmark for low-latency performance. Over-the-air signals historically were near-instant, synchronised and predictable. Streaming delivery, on the other hand, is still catching up. Even with advancements in delivery platforms, it's not uncommon for digital audiences to experience delays of tens of seconds or more compared to traditional broadcast of the same content. These highly engaged audiences – who are increasingly viewing sports and other live content on mobile devices – expect real-time delivery, and significant delay is considered unacceptable.
So what's the hold-up in content delivery with streaming? Well, there isn't a single culprit – latency is a cumulative result of multiple workflow steps, including cloud-based operations. Transmitting content over long-distance networks introduces latency, as does processing at remote cloud servers, encoding and decoding processing, and the monitoring and feedback loop. All of these delays add up fast. For media companies, the challenge is to address these issues without sacrificing quality.
Fortunately, the media industry around the globe is making progress on multiple fronts when it comes to bridging the latency gap with traditional broadcast. Low-latency protocols like LL-HLS and CMAF are moving from theory to development, reducing sources of delay in the delivery systems. Meanwhile, JPEG XS is being integrated into live production environments to reduce latency in the contribution and production networks while maintaining origination quality.
JPEG XS stands apart from other codecs because it wasn't designed with the goal of maximising compression efficiency. Instead, its main objective is to retain the benefits associated with an uncompressed stream – visually lossless quality, super-low latency and low complexity. JPEG XS achieves visually lossless quality through modern intraframe encoding. In addition, it supports a constant bitrate and high dynamic range (HDR), and aligns well with SMPTE ST 2110 for IP network transmission. And while other codecs used in live production workflows can accumulate significant latency in the encoding and decoding stages, JPEG XS delivers extremely low latency – a handful of milliseconds for combined encoding and decoding.
Furthermore, despite the acceleration of cloud-based playout, media companies are realising that not all workloads belong in the cloud and not all latency is created equal. As a result, hybrid 'ground and cloud' architectures are gaining traction as a way to optimise performance. By running latency-sensitive processes at the edge or on-prem while leveraging the cloud for scale and elasticity, media companies can better balance speed and flexibility. This balance is essential in live environments where the ability to deliver real-time content is now a baseline expectation.
