Video: Low Latency Streaming

There are two phases to reducing streaming latency. One is to optimise the system you already have, the other is to move to a new protocol. This talk looks at both approaches achieving parity with traditional broadcast media through optimisation and ‘better than’ by using CMAF.

In this video from the Northern Waves 2019 conference, Koen van Benschop from Deutsche Telekom examines the large and low-cost latency savings you can achieve by optimising your current HLS delivery. With the original chunk sizes recommended by Apple being 10 seconds, there are still many services out there which are starting from a very high latency so there are savings to be had.

Koen explains how the total latency is made up by looking at the decode, encode, packaging and other latencies. We quickly see that the player buffer is one of the largest, the second being the encode latency. We explore the pros and cons of reducing these and see that the overall latency can fall to or even below traditional broadcast latency depending, of course, on which type (and which country’s) you are comparing it too.

While optimising HLS/DASH gets you down to a few seconds, there’s a strong desire for some services to beat that. Whilst the broadcasters themselves may be reticent to do this, not wanting to deliver online services quicker than their over-the-air offerings, online sports services such as DAZN can make latency a USP and deliver better value to fans. After all, DAZN and similar services benefit from low-second latency as it helps bring them in line with social media which can have very low latency when it comes to key events such as goals and points being scored in live matches.

Stefan Arbanowski from Fraunhofer leads us through CMAF covering what it is, the upcoming second edition and how it works. He covers its ability to use .m3u8 (from HLS) and .mpd (from DASH) playlist/manifest files and that it works both with fMP4 and ISO BMFF. One benefit from DASH is it’s Common Encryption standard. Using this it can work with PlayReady DRM, Fairplay and others.

Stefan then takes a moment to consider WebRTC. Given it proposes latency of less than one second, it can sound like a much better idea. Stefan outlines concerns he has about the ability to scale above 200,000 users. He then turns his attention back to CMAF and outlines how the stream is composed and how the player logic works in order to successfully play at low latency.

Watch now!
Speakers

Koen van Benschop Koen van Benschop
Senior Manager TV Headend and DRM,
Deutsche Telekom
Stefan Arbanowski Stefan Arbanowski
Director Future Applications and Media,
Fraunhofer FOKUS

Video: From WebRTC to RTMP

With the demise of RTMP, what can WebRTC – its closest equivalent – learn from it? RTC stands for Real-Time Communications and hails from the video/voice teleconferencing world. RTC traditionally has ultra-low latency (think sub-second; real-time) so as broadcasters and streaming companies look to reduce latency it’s the obvious technology to look at. However, RTC comes from a background of small meetings, mixed resolutions, mixed bandwidths and so the protocols underpinning it can be lacking what broadcast-style streamers need.

Nick Chadwick from MUX looks at the pros and cons of the venerable RTMP (Real Time Messaging Protocol). What was in it that was used and unused? What did need that it didn’t have? What gap is being left by its phasing out?

Filling these increasing gaps is the focus of the streaming community and whether that comes through WebRTC, fragmented MP4 delivered over web sockets, Low-Latency HLS, Apple’s Low-Latency HLS, SASH, CMAF or something else…it still needs to be fulfilled.

Nick finishes with two demos which show the capabilities of WebRTC which outstrip RTMP – live mixing on a browser. WebRTC clearly has a future for more adventurous services which don’t simply want to deliver a linear channel to sofa-dwelling humans. But surely Nick’s message is WebRTC needs to step up to the plate for broadcasters in general to enable them to achieve <1 second end-to-end latency in a way which is compatible with broadcast workflows.

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Speaker

Nick Chadwick Nick Chadwick
Software Engineer,
Mux

Video: Deploying WebRTC In A Low-Latency Streaming Service

WebRTC is an under appreciated streaming protocol with sub-second latency. Several startups are working hard to harness this technology born by Google for use in video conferencing for live streaming.

When you look at the promised latencies, you can see why. CMAF, the lowest-latency protocol for live streaming using HLS-style chunked file delivery is gaining wider adoption and provides a very impressive latency reduction, however it typically stops at between 4 and 2 seconds. To get below a second, WebRTC is almost the only option out there.

In this talk, Millicast CTO Dr. Alex Gouaillard looks at the misunderstandings and misinformation are out there regarding WebRTC. Dr. Alex covers WebRTC now having ABR, using over multiple hops, the testing ecosystem and much more.

Dr. Alex also covers the lessons learnt over the last two years of development and implementation of the standard and finishes by looking to the future which will bring in QUIC, AV1 and Web ASM

Speaker

Alex Gouaillard Alex Gouaillard
Founder & CTO,
Millicast

Video: How to Identify Real-World Playout Options

There are so many ways to stream video, how can you find the one that suits you best? Weighing up the pros and cons in this talk is Robert Reindhardt from videoRx.

Taking each of the main protocols in turn, Robert explains the prevalence of each technology from HLS and DASH through to WebRTC and even Websockets. Commenting on each from his personal experience of implementing each with clients, we build up a picture of when the best situations to use each of them.

Speakers

Robert Reinhardt Robert Reinhardt
CTO,
videoRX