If you’re new to WebRTC, Jitsi was the first open source Selective Forwarding Unit (SFU) and continues to be one of the most popular WebRTC platforms. They were in the news last week because their parent group inside Atlassian was sold off to Slack but the team clarified this does not have any impact on the Jitsi team. Helping to show they are still chugging along, they released a new feature they wanted to talk about – off-stage layer suspension. This is a technique for minimizing bandwidth and CPU consumption when using simulcast. Simulcast is a common technique used in multi-party video scenarios. See Oscar Divorra’s post on this topic and that Fippo post just last week for more on that. Even if you are not implementing a simulcast, this is a good post for understanding how to control bandwidth and to see some follow-along reverse-engineering on how Google does things in its Hangouts upgrade called Meet. ...
Simulcast is one of the more interesting aspects of WebRTC for multiparty conferencing. In a nutshell, it means sending three different resolution (spatial scalability) and different frame rates (temporal scalability) at the same time. Oscar Divorra’s post contains the full details.
Usually, one needs a SFU to take advantage of simulcast. But there is a hack to make the effect visible between two browsers — or inside a single page. This is very helpful for single-page tests or fiddling with simulcast features, particular the ability to enable only certain spatial layers or to control the target bitrate of a particular stream. ...
Dealing with multi-party video infrastructure can be pretty daunting. The good news is the technology, products, and standards to enable economical multiparty video in WebRTC has matured quite a bit in the past few years. One of the key underlying technologies enabling some of this change is called simulcast. Simulcast has been an occasional sub-topic here at webrtcHacks in the past and it is time we gave it more dedicated attention.
To do that we asked Oscar Divorra Escoda, Tokbox’s Senior Media Scientist and Media Cloud Engineering Lead to walk us through it. Tokbox was one of the first to market with a SFU and Oscar shares some of his learnings below. ...
During this year’s WWDC keynote, Apple announced the availability of FaceTime in web browsers, making it available to Android and Windows users. It has been six years since the last time we looked at FaceTime (FaceTime Doesn’t Face WebRTC) so it was about time for an update. It had to be WebRTC and as I’ll show – it is very much WebRTC.
FaceTime Web does use WebRTC for media and it uses the Insertable Streams API for end-to-end encryption. It also uses an interesting approach to avoid simulcast.
Interview with WebRTC standards co-chair and author, Bernard Aboba. We cover the current status of WebRTC and where it is headed including WebRTC-NV, Simulcast, SVC, AV1, WebTransport, WebCodecs, ML and more.
Walkthrough and deep analysis of how Azure Communications Service makes use of WebRTC by Gustavo Garcia
A couple of weeks ago, the Chrome team announced an interesting Intent to Experiment on the blink-dev list about an API to do some custom processing on top of WebRTC. The intent comes with an explainer document written by Harald Alvestrand which shows the basic API usage. As I mentioned in my last post, this is the sort of thing that maybe able to help add End-to-End Encryption (e2ee) in middlebox scenarios to WebRTC.
I had been watching the implementation progress with quite some interest when former webrtcHacks guest author Emil Ivov of jitsi.org reached out to discuss collaborating on exploring what this API is capable of. Specifically, we wanted to see if WebRTC Insertable Streams could solve the problem of end-to-end encryption for middlebox devices outside of the user’s control like Selective Forwarding Units (SFUs) used for media routing. ...
Time for another opinionated post. This time on… end-to-end encryption (e2ee). Zoom apparently claims it supports e2ee while it can not satisfy that promise. Is WebRTC any better?
Zoom does not have End to End Encryption
Let’s get to the bottom of things fast: Boo Zoom!
I reviewed how Zoom’s implements their web client last year.
I’m not really surprised of their general lack of e2ee given that their web client did not provide any encryption on top of TLS or WebRTC’s DataChannel. For reasons we will discuss below, this means they weren’t doing any obvious e2ee there. ...
SDP has been a frequent topic, both here on webrtcHacks as well as in the discussion about the standard itself. Modifying the SDP in arcane ways is referred to as SDP munging. This post gives an introduction into what SDP munging is, why its done and why it should not be done. This is not a guide to SDP munging.
Review of Chrome's migration to WebRTC's Unified Plan, how false metrics may have misguided this effort, and what that means moving forward.