Can any stream be too low-latency? For some matching broadcast latency, is all they need. But for others, particularly for gaming, gambling or more interactive services, sub-second is a must and they are happy to swap out parts of their technology stack to make that happen. WebRTC is often seen as the best choice for anyone wanting to go achieve an almost instant stream. Started by Google in 2011 for video conferencing applications, WebRTC hit a 1.0 release in 2018 and has been adopted by a number of companies catering to the broadcast market.
WebRTC stands out among the plethora of streaming protocols since it is an actual stream of data and not a series of files transferred just in time. Traditionally buffers have been heavily used in streaming because it was so hard to get data to the player when the mainstream internet was starting out in the 90s and as the mobile internet was establishing itself 10 years later. Whilst those buffers are very helpful in dealing with delayed data, they are a big set back in delivering a low-latency stream. With WebRTC, there is very little buffering, so when using the protocol you have to understand that you may not get all your data delivered and if packets are missing glitches will be seen. This is one significant difference since MPEG DASH and HLS will either show you a blank screen or a perfect rendition of the file chunk that was sent thanks to TCP. This is an example of the compromises of going to sub-second latency; there are no second chances to get the packet again. And whilst this compromise may be a great exchange for an auction site or betting service, for other streaming services, it may be better to use CMAF with 3-second latency.
In this talk, Limelight Networks Video Architect Andrew Crowe introduces WebRTC and explains how it can be deployed. He starts by talking about the video codecs it contains. VP9 has recently been added to the options and for a long time, it was a VP8 technology. Andrew explains how the codecs it carries does have a knock-on effect on its compatibility with browsers. UDP is the underlying technology to all low-latency technologies since the bureaucracy of TCP/IP gets in the way of real-time media streams. Andrew also explains how security pervades WebRTC from its use of DTLS (which is like HTTPS/TLS for UDP) to secure RTP and SRTCP.
The last part of the talk discusses the architectures that CDN LimeLight uses to enable large-scale WebRTC streams including the need to get through firewalls. Andrew discusses how some features of the technology suit small-scale events, but can’t be used with thousands of viewers. He also discusses how adaptive bitrate streams can be delivered, although not within WebRTC itself, there is enough information to implement ABR in addition to the standard stream.
Latency seems to be the new battleground for streaming services. While optimising bandwidth and quality are still highly important, they are becoming mature parts of the business of streaming where as latency, and technologies to minimise it – as Apple showed this month – are still developing and vying for position.
Here, the Streaming Video Alliance brings together people from large streaming services to explore this topic finding out what they’ve been doing to reduce it, the problems they’ve faced and the solutions which are on the table.
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,
Digital TV Europe
Subscribe to get daily updates
Views and opinions expressed on this website are those of the author(s) and do not necessarily reflect those of SMPTE or SMPTE Members.
This website is presented for informational purposes only. Any reference to specific companies, products or services does not represent promotion, recommendation, or endorsement by SMPTE