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.
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.
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.)
Date: Thursday February 28th 2019, 10am PT / 1PM ET / 18:00 GMT
Streaming continues to grow, in amount streamed, in people consuming it and in importance within this and other industries. One things which has always been an enabler yet made streaming harder to deploy is its rapid evolution. Whilst this has been a boon for smaller, nimbler companies – both content producers and service providers – the streaming has now arrived at most companies in one way or another and this breadth of use-cases has kept streaming tech moving forward and showing no signs of abatement.
Some aspects are changing. For instance we are seeing the first patent-free MPEG standard proposals (EVC, which has basic patent-free functionality and a better performing patent-controlled profile) on the heels of AV1. We’re seeing low-latency efforts such as CMAF taking hold as an alternative to WebRTC. With CMAF being much closer to the ever popular HLS, this may well beat out WebRTC in deployments at the cost of a slightly higher, but much improved latency.
To bring all of this in to focus for 2019, Jason Thibeault from the Streaming Video Alliance is bringing together a panel of experts to look at the coming trends and to give us an idea of what to look out for, and how to make sense, of 2019’s year of video delivery.
Nobody wants to find out about a big play or major news event on Twitter before they see it in their video stream, so reducing latency is crucial for OTT services’ success. Likewise, ultra-low latency is crucial for interactive streaming applications. Depending on your use case, a few seconds of latency might be fine, or you might need to try to hit that sub-second target.
Learn which technologies and solutions are best for your business, and make sure your viewers get their video on time, every time. In this webinar, you’ll learn the following:
Why it’s important to evaluate and improve latency end-to-end, including software and services, encoder, platform, and player
How to decide which technology and solution is best for your use case (e.g. CMAF, HLS/DASH, WebRTC, Websocket)
How chunked CMAF offers a standards-based approach that allows latency to be decoupled from segment duration
How chunked CMAF leverages existing CDN HTTP capacity to provide low-latency solutions at high scale
How WebRTC can be used to deliver live video sub-second latency at scale, and provide rich, interactive experiences for live streaming applications
How a single misconfigured component can undo any other effort to achieve low latency
How integrated solutions create new business opportunities for low latency interactive use cases
How to achieve low latency across all platforms and devices
VP of Product Strategy,
Moderator: Eric Schumacher-Rasmussen
A step by walk through fuboTV’s FIFA World Cup streaming system. Delivering to FOX, Billy Romero and Thomas Symborski explain the challenges and the successes streaming from Russia to the US in UHD HDR.
This talk from Demuxed 2018 shows examples of how social media can help get reports back of problems and the real life usage stats. Billy and Thomas explain their system from commissioning through vendor choice and all the way to CMAF delivery.
A refreshingly transparent talk which is well worth watching!
Will Law from Akamai proves his chunky credentials by telling us how to achive very low-latency streaming in his talk at Demuxed 2018.
In the jungle of solutions for low latency live streaming, there are many current options ranging from WebRTC, to proprietary UDP protocols to standard segmented media with ever-shortening segments. This session highlights one of these – chunked-encoded chunked-transferred CMAF – as a optimal and practical confluence of both reach and performance. On the technical side we’ll investigate the underlying technology, the latency regimes possible, compatibility with legacy players, cachability on delivery networks and player behavior requirements. Including live demonstrations of several streams on a production network. This talks brings a standards perspective from DVB and DASH as well as CDN support. As a sweetener, Will points you at open source code on both the encoder and player side for doing this all yourself.
Chief Architect, Media Cloud Engineering