Video: There and back again: reinventing UDP streaming with QUIC

QUIC is a encrypted transport protocol for increased performance compared to HTTP but will this help video streaming platforms? Often conflated with HTTP/3, QUIC is a UDP-based way evolution of HTTP/2 which, in turn, was a shake-up of the standard HTTP/1.1 delivery method of websites. HTTP/3 uses the same well-known security handshake from TLS 1.3 that is well adopted now in websites around the world to provide encryption by default. Importantly, it creates a connection between the two endpoints into which data streams are multiplexed. This prevents the need to constantly be negotiating new connections as found in HTTP/1.x so helping with speed and efficiency. These are known as QUIC streams.

QUIC streams provide reliable delivery, explains Lucas Pardue from Cloudflare, meaning it will recover packets when they are lost. Moreover, says Lucas, this is done in an extensible way with the standard specifying a basic model, but this is extensible. Indeed, the benefit of basing this technology on UDP is that changes can be done, programmatically, in user-space in lieu of the kernel changes that are typically needed for improved TCP handling on which HTTP/1.1, for example, is based.

QUIC hailed from a project of the same name created by Google which has been taken in by the IETF and, in the open community, honed and rounded into the QUIC we are hearing about today which is notably different from the original but maintaining the improvements proved in the first release. HTTP/3 is the syntax which is a development on from HTTP/2 which uses the QUIC transport protocol underneath or as Lucas would say, “HTTP/3 is the HTTP application mapping to the QUIC transport layer.” Lucas is heavily involved with in the IETF effort to standardise HTTP/3 and QUIC so he continues in this talk to explain how QUIC streams are managed, identified and used.

It’s clear that QUIC and HTTP/3 are being carefully created to be tools for future, unforeseen applications with clear knowledge that they have wide applicability. For that reason we are already seeing projects to add datagrams and RTP into the mix, to add multiparty or multicast. In many ways mimicking what we already have in our local networks. Putting them on QUIC can enable them to work on the internet and open up new ways of delivering streamed video.

The talk finishes with a nod to the fact that SRT and RIST also deliver many of the things QUIC delivers and Lucas leaves open the question of which will prosper in which segments of the broadcast market.

The Broadcast Knowledge has well over 500 talks/videos on many topics so to delve further into anything discussed above, just type into the search bar on the right. Or, for those who like URLs, just add your search query to the end of this URL http://thebroadcastknowledge.com/tag/.

Lucas has already written in detail about his work and what HTTP 3 is on his Cloudflare blog post.

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Speaker

Lucas Pardue Lucas Pardue
Senior Software Engineer,
Cloudflare

Video: 2019 What did I miss? – Introducing Reliable Internet Streaming Transport

By far the most visited video of 2019 was the Merrick Ackermans’ review of RIST first release. RIST, the Reliable Internet Stream Transport protocol, aims to be an interoperable protocol allowing even lossy networks to be used for mission-critical broadcast contribution. Using RIST can change a bade internet link into a reliable circuit for live programme material, so it’s quite a game changer in terms of cost for links.

An increasing amount of broadcast video is travelling over the public internet which is currently enabled by SRT, Zixi and other protocols. Here, Merrick Ackermans explains the new RIST specification which aims to allow interoperable internet-based video contribution. RIST, which stands for Reliable Internet Stream Transport, ensures reliable transmission of video and other data over lossy networks. This enables broadcast-grade contribution at a much lower cost as well as a number of other benefits.

Many of the protocols which do similar are based on ARQ (Automatic Repeat-reQuest) which, as you can read on wikipedia, allows for recovery of lost data. This is the core functionality needed to bring unreliable or lossy connections into the realm of usable for broadcast contribution. Indeed, RIST is an interesting merging of technologies from around the industry. Many people use Zixi, SRT, and VideoFlow all of which can allow safe contribution of media. Safe meaning it gets to the other end intact and un-corrupted. However, if your encoder only supports Zixi and you use it to deliver to a decoder which only supports SRT, it’s not going to work out. The industry as accepted that these formats should be reconciled into a shared standard. This is RIST.

File-based workflows are mainly based on TCP (Transmission Control Protocol) although, notably, some file transfer service just as Aspera are based on UDP where packet recovery, not unlike RIST, is managed as part of the the protocol. This is unlike web sites where all data is transferred using TCP which sends an acknowledgement for each packet which arrives. Whilst this is great for ensuring files are uncorrupted, it can impact arrival times which can lead to live media being corrupted.

RIST is being created by the VSF – the Video Standards Forum – who were key in introducing VS-03 and VS-04 into the AIMS group on which SMPTE ST 2022-6 was then based. So their move now into a specification for reliable transmission of media over the internet has many anticipating great things. At the point that this talk was given the simple profile has been formed. Whist Merrick gives the details, it’s worth pointing out that this doesn’t include intrinsic encryption. It can, of course, be delivered over a separately encrypted tunnel, but an intrinsic part of SRT is the security that is provided from within the protocol.

Despite Zixi, a proprietary solution, and Haivision’s open source SRT being in competition, they are both part of the VSF working group creating RIST along with VideoFlow. This is because they see the benefit of having a widely accepted, interoperable method of exchanging media data. This can’t be achieved by any single company alone but can benefit all players in the market.

This talk remains true for the simple profile which just aims to recover packets. The main protocol, as opposed to ‘simple’, has since been released and you can hear about it in a separate video here. This protocol adds FEC, encryption and other aspects. Those who are familiar with the basics may whoosh to start there.

Speaker

Merrick Ackermans Merrick Ackermans
Chair,
VSF RIST Activity Group

Video: The challenges of deploying Apple’s Low Latency HLS In Real Life

HLS has taken the world by storm since its first release 10 years ago. Capitalising on the already widely understood and deployed technologise already underpinning websites at the time, it brought with it great scalability and the ability to seamlessly move between different bitrate streams to help deal with varying network performance (and computer performance!). In the beginning, streaming latency wasn’t a big deal, but with multi-million pound sports events being routinely streamed, this has changed and is one of the biggest challenges for streaming media now.

Low-Latency HLS (LL-HLS) is Apple’s way of bringing down latency to be comparable with broadcast television for those live broadcast where immediacy really matters. The release of LL-HLS came as a blow to the community-driven moves to deliver lower latency and, indeed, to adoption of MPEG-DASH’s CMAF. But as more light was shone on the detail, the more questions arose in how this was actually going to work in practice.

Marina Kalkanis from M2A Media explains how they have been working with DAZN and Akamai to get LL-HLS working and what they are learning in this pilot project. Choosing the new segment sizes and how they are delivered is a key first step in ensuring low latency. M2A are testing 320ms sizes which means very frequent requests for playlists and quickly growing playlist files; both are issues which need to be managed.

Marina explains the use of playlist shortening, use of HTTP Push in HTTP2 to reduce latency, integration into the CDN and what the CDN is required to do. Marina finishes by explaining how they are conducting the testing and the status of the project.

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Speaker

Marina Kalkanis Marina Kalkanis
CEO,
M2A Media

Webinar: Networking Fundamentals


Date: Thursday 12th December, 1pm EST / 18:00 GMT

Networking is increasingly important throughout the broadcast chain. This webcast picks out the fundamentals that underpin SMPTE ST 2110 and that help deliver video streaming services. We’ll piece them together and explain how they work, leaving you with more confidence in talking about and working with technologies such as multicast video and HTTP Live Streaming (HLS).

Register now!
Speaker

Russell Trafford-Jones Russell Trafford-Jones
Editor, https://TheBroadcastKnowledge.com
Manager, Support & Services, Techex