Low latency streaming is always a compromise, but what can be done to keep QOE high?
This on-demand webinar looks at CMAF and presents some real-world data on this low latency technique. The webinar starts by explaining that CMAF is a low-latency streaming technology similar to HLS and other streaming protocols where the idea is to deliver the video as small files. Olivier and Alain from Harmonic explain how this is done and look at some of the trade-offs and compromises that are needed and introduce techniques to keep QOE high. They also look at deployment in cloud vs. on premise.
Pieter-Jan Speelmans talks about play tradeoffs and optimisations within the player. CMAF allows the buffer to be reduced and whilst a bad network may mean you buffer is similar to ‘normal’, but in good networks, this buffer can be brought down significantly. He also talks about how ABR switching is impacted by GOP length even in CMAF.
Viaccess-Orca explains how DRM works with CMAF and looks at some of the challenges including licences acquisition time and overloading licence servers at the beginning of events. Akamai’s Will Law explains some benefits of CMAF and the near-real-time of chunk-based transfer (HTTP 1.1) and how downloading chunks at full speed leads to problems when the same broadband link is used by several clients.
There are lots of good talks on CMAF, but this is one of the few which talks about CMAF not as theory, but as is deployable today.
CMAF brings low latency streams of less than 4 seconds into the realms of possibility, WebRTC pushes that below a second – but which is the right technology for you?
Date: June 12th 2019 Time: 11am PST / 2pm EST / 19:00 BST
CMAF represents an evolution of the tried and tested technologies HLS and DASH. With massive scalability and built upon the well-worn tenants of HTTP, Netflix and a whole industry was born and is thriving on these still-evolving technologies. The push to reduce latency further and further has resulted in CMAF which can be used to deliver streams with five to ten times lower latencies.
WebRTC is a Google-backed streaming protocol with the traditional meaning of streaming; it pushes a stream to you a opposed to the HLS-style methods of making small files available for download and reassembly into a stream. One benefit of this is extremely low bitrates of 1 second or less. Used widely by Google Hangouts and Facebook messenger, WebRTC is increasingly an option for more broadcast-style streaming services from live sports & music to gaming and gambling.
Both have advantages and draw-backs so Wowza’s Barry Owen and Anne Balistreri are here to help navigate the ins and outs of both technologies plus answer your questions.
JPEG XS is a brand-new, ultra-low latency standard delivering JPEG 2000 quality with 1000x lower latency; microseconds instead of milliseconds. This mezzanine compression standard promises compression ratios of up to 10:1, resolutions of up to 8K plus HDR and features frame rates from 24 to 120 fps.
Jean-Baptiste Lorent from intoPIX shows how JPEG-XS can be used with SMPTE ST-2110 stack. Part -22 of ST 2110 allows for transport of compressed video essence as an alternative to uncompressed essence – all the other elementary streams stay the same, just the video RTP payload changes. This approach saves a lot of bandwidth and keeps all the existing advantages of moving from SDI to IP at the same time.
Based on TICO which arrived in products four or more years ago allowing HD products to support UHD workflows, JPEG XS was also designed for visually lossless quality and maintaining that quality over multiple re-encoding stages. The combination of very-low microsecond-latency and relatively low bandwidth makes it ideal for remote production of live events.
ISO BMFF a standardised MPEG media container developed from Apple’s Quicktime and is the basis for cutting edge low-latency streaming as much as it is for tried and trusted mp4 video files. Here we look into why we have it, what it’s used for and how it works.
ISO BMFF provides a structure to place around timed media streams whilst accommodating the metadata we need for professional workflows. Key to its continued utility is its extensible nature allowing additional abilities to be added as they are developed such as adding new codecs and metadata types.
ATSC 3.0’s streaming mechanism MMT is based on ISO BMFF as well as the low-latency streaming format CMAF which shows that despite being over 18 years old, the ISO BMFF container is still highly relevant.
Thomas Stockhammer is the Director of Technical Standards at Qualcomm. He explains the container format in structure and origin before explaining why it’s ideal for CMAF’s low-latency streaming use case, finishing off with a look at immersive media in ISO BMFF.
There are two main modern approaches to low-latency live streaming, one is CMAF which used fragmented MP4s to allow frame by frame delivery of chunks of data. Similar to HLS, this is becoming a common ‘next step’ for companies already using HLS. Keeping the chunk size down reduces latency, but it remains doubtful if sub-second streaming is practical in real world situations.
Steve Miller Jones from Limelight explains the WebRTC solution to this problem. Being a protocol which is streamed from the source to the destination, this is capable of sub-second latency, too, and seems a better fit. Limelight differentiate themselves on offering a scalable WebRTC streaming service with Adaptive Bitrate (ABR). ABR is traditionally not available with WebRTC and Steve Miller Jones uses this as an example of where Limelight is helping this technology achieve its true potential.
Comparing and contrasting Limelight’s solution with HLS and CMAF, we can see the benefit of WebRTC and that it’s equally capable of supporting features like encryption, Geoblocking and the like.
Ultimately, the importance of latency and the scalability you require may be the biggest factor in deciding which way to go with your sub-second live streaming.
With live online viewing delayed by up to 30 seconds or more compared to broadcast TV, enriching the viewing experience with online content, while ensuring that all viewers see the action at the same time, is a significant challenge. To provide viewers with engaging online experiences that keep them coming back for more, service providers need true real-time streaming.
This webinar will cover questions such as:
How important is latency for live online streaming?
Which live streaming workflows offers the greatest opportunity to generate additional revenue?
What are the main challenges faced by online video service providers when live-streaming major events such as sports tournaments?
Being a webinar from Limelight, you will also hear
How Limelight realtime streaming minimizes latency
How to reach the widest audience with native browser support
How to enable new business models with interactivity
How to reach viewers everywhere
All this along with key findings from DTVE’s industry survey, showing that industry executives believe live streaming could ultimately supplant broadcast technology, but challenges remain.
Vice President of Product Strategy,
VP9 is a well-known codec, but it hasn’t seen many high-profile, live deployments which makes Twitch’s move to deliver their platform using VP9 in preference over AVC all the more interesting.
Here, Yueshi Shen from Twitch, explains the rationale for VP9 by explaining the scale of Twitch and looking at their AVC bitrate demands. He explains the patent issues with HEVC and VP9 then looks at decoder support across devices and platforms. Importantly, encoder implementation is examined leading to Twitch’s choice of FPGA to provide live encoding.
Yueshi then looks at the potential of AV1 to Switch_Frame to provide low-latency broadcast at scale.
There are two ways to stream video online, either pushing from the server to the device like WebRTC, MPEG transport streams and similar technologies, or allowing the receiving device to request chunks of the stream which is how the majority of internet streaming is done – using HLS and similar formats.
Chunk-based streaming is generally seen as more scalable of these two methods but suffers extra latency due to buffering several chunks each of which can represent between 1 and, typically, 10 seconds of video.
CMAF is one technology here to change that by allowing players to buffer less video. How does this achieve this? An, perhaps more important, can it really cut costs? Iraj Sodagar from NexTreams is here to explain how in this talk from Streaming Media West, 2018.
A brief history of CMAF (Common Media Format)
The core technologies (ISO BMFF, Codecs, captions etc.)
Limelight, Streaming Video Alliance and Videonet come together to discuss the introduction of WebRTC’s sub-second latency for live streaming which is opening the way for a ‘better-than-broadcast’ experience – enabling new ways to engage viewers and monetise them.
WebRTC provides real-time video delivery and can now be implemented in a CDN environment for large-scale distribution and has extremely low latency.
This webinar covers:
Making WebRTC part of your workflow
Compression, DRM & ad insertion
Innovation opportunities for broadcasters and challenger OTT providers
Special focus on increasing viewer engagement
creating new revenue streams.
New business partnerships
Optimisation for multiscreen television & connected TV devices