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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. ...  Continue reading

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. ...  Continue reading

Hear No Evil picture

One of the great things about WebRTC is that it is built right into the web platform. The web platform is generally great for WebRTC, but occasionally it can cause huge headaches when specific WebRTC needs do not exactly align with more general browser usage requirements. The latest example of this is has to do with the autoplay of media where sound(s) suddenly went missing for many users. Former webrtcHacks guest author Dag-Inge Aas has been dealing with this first hand. See below for his write-up on browser expectations around the playback of media, the recent Chrome 66+ changes, and some tips and tricks for working around these issues. ...  Continue reading

As the year 2017 comes to an end, there was a small present. Hangouts started to support Firefox with WebRTC instead of rejecting access – plugin access had been unavailable since Firefox 53 removed NPAPI in April 2017. While it had been public for a while that the Firefox WebRTC team had been testing this, it was a nice Christmas present to see this shipped. Tsahi Levent-Levi was one of the first people to notice.
This comes at a time where other Google teams are being criticized for promoting Chrome-only experiences. Kudos to the Hangouts team for showing that you still care about the web as an open platform! ...  Continue reading

Two weeks ago Microsoft’s Bernard Aboba (and former webrtcHack’s interviewee) gave an update on Edge’s ORTC and WebRTC at the Microsoft Build conference. He covered some big topics including VP8 and WebRTC 1.0 support. You can see the update video at the link above or read the follow-up post for details. Then last week Microsoft announced plug-in free Skype on the Edge browser.

I had some questions; Fippo had some questions; so we asked Bernard if he could publicly respond here. It turned out Bernard and his teammate on the Edge Browser team, Shijun Sun, were building a running list of questions they wanted to address too. Here it is. ...  Continue reading

There have been many major WebRTC launches in the past months including Facebook and KimDotCom. Before those, Mozilla started bundling a new WebRTC calling service right into Firefox. Of course we wanted to check out to see how it worked.

To help do this we called on the big guns – webrtcHacks guest columnist Philipp Hancke. Philipp is one of the smartest guys in WebRTC outside of Google. In addition to his paid work for &yet he is the leading non-googler to contribute to the webrtc demos and samples and is also a major contributor to the Jitsi Meet and strophe.jingle projects. Google even asks him to proof-read their WebRTC release notes...  Continue reading

Biggie vs. Tupac. Gates vs. Jobs. Apple vs. Samsung.  Nothing catches people’s attention for no legitimate reason like a feud. Unfortunately this isn’t just a celebrity phenomenon. Feuds have been endemic even to real communications as well. From the very beginning, Elisha Gray’s dispute with Alexander Graham Bell over the original telephone patent showed the industry has a propensity for squabbles. Unfortunately we have become so accustomed to feuds that we sometimes fabricate battles that do not really exist. I fear that this is often the case with one of the most important, but misunderstood efforts affecting WebRTC’s future – Object Real Time Communications (ORTC). ...  Continue reading

Update: Philipp continues to reverse engineer Hangouts using chrome://webrtc-internals. Please see the bottom section for new analysis he just put together in the past couple of days based on Chrome 38.

As initiators and major drivers of WebRTC, Google was often given a hard time for not supporting WebRTC in its core collaboration product. This recently changed when WebRTC support for Hangouts was added with Chrome 36.

So obviously we wanted to check out how this worked. We also were curious to see how a non-googler could make some practical use of chrome://webrtc-internals. Soon thereafter I came across a message from Philipp Hancke (aka Hornsby Cornflower) saying he had already starting looking at the new WebRTC hangouts with webrtc-internals. Fortunately I was able to convince him to share his findings and thorough analysis. ...  Continue reading

As I mentioned in my ‘WebRTC meets telecom’ article a couple of weeks ago, at Quobis we’re currently involved in 30+ WebRTC field trials/POCs which involve in one way or another a telco network. In most cases service providers are trying to provide WebRTC-based access to their existing/legacy infrastructure and services (fortunately, in some cases it’s not limited to do only that). To achieve all this, one of the pieces they need to deploy is a WebRTC Gateway. But, what is a WebRTC Gateway anyway? A year ago I had the chance to provide a first answer during the Kamailio World Conference 2013 (see my presentation WebRTC and VoIP: bridging the gap) but, since Lorenzo Miniero has recently released an open source, modular and general purpose WebRTC gateway called Janus, I thought it would be great to get him to share his experience here. ...  Continue reading

As WebRTC implementations and field trials evolve, field experience is telling us there are still a number of open issues to make this technology deployable in the real world and the fact that we would probably do some things differently if we started all over again. As an example, see the recent W3C discussion What is missing for building (WebRTC) real services or Quobis‘ CTO post on WebRTC use of SDP.

Tim Panton, contextual communications consultant at Westhawk Ltd,  has gone through some of these issues. During the last couple of years we had the chance to run some workshops together and have some good discussions in the IETF and W3C context. Tim’s expertise is very valuable and I thought it would be a good idea to have him here to share some of his experiences with our readers. It ended up as a rant. ...  Continue reading