WebRTC is used on a massive scale thanks to Facebook messenger and Google, but when it comes to video streaming services, some find its open source codec VP8 too restrictive. WebRTC is actively evolving to adapt and become codec agnostic though this work is ongoing. In the meantime, Comcast is here to show us there is a way to inject the codec of your choice into WebRTC.
Finding that many of their video capture devices, CCTV cameras and the like, had hardware AVC encoders, Bryan Meissner explains Comcast didn’t feel it had much of a choice in codec, therefore they looked for a way to make WebRTC to carry AVC.
While forcing an unsupported codec into a protocol wasn’t ideal, they were able to leave much of WebRTC unchanged. The RTP and Data channels were established as normal and peering continued to work as ever. With control of both the send and receive side, the team found they could pick out the data from the WebRTC stack ahead of the normal decoder and feed that into Exoplayer using its API. This allowed playback on Android devices. Bryan goes on to explain the approach for iOS and web browsers. As WebRTC is ‘baked in’ to browsers, there really are very few ways to change the signal flow.
At the end of the day, Comcast made this work and used it in production or many years, Jeff Cardillo explains as he wraps up this video. But he also takes time to talk through some of the problems. Having to bypass parts of a program with parts of another library does increase complexity. Not only does the code become more complex but the code becomes platform specific, you need control over the source and keeping the individual parts synchronously up to date can be a balancing act.
Jeff finishes this talk from Demuxed SF 2019 by elaborating on the mobile and browser tradeoffs at play.
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.
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.
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
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