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
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.
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.
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.