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webrtcH4cKS: ~ Guide to WebRTC with Safari in the Wild (Chad Phillips)

Posted by Chad Phillips on September 7, 2018
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It has been more than a year since Apple first added WebRTC support to Safari. My original post reviewing the implementation continues to be popular here, but it does not reflect some of the updates since the first limited release. More importantly, given its differences and limitations, many questions still remained on how to best develop WebRTC applications for Safari.

I ran into Chad Phillips at Cluecon (again) this year and we ended up talking about his arduous experience making WebRTC work on Safari. He had a great, recent list of tips and tricks so I asked him to share it here. ...

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webrtcH4cKS: ~ VR Video Calling with WebRTC and WebVR (Dan Jenkins)

Posted by Dan Jenkins on August 27, 2018
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WebRTC isn’t the only cool media API on the Web Platform. The Web Virtual Reality (WebVR) spec was introduced a few years ago to bring support for virtual reality devices in a web browser. It has since been migrated to the newer WebXR Device API Specification.

I was at ClueCon earlier this summer where Dan Jenkins gave a talk showing that it is relatively easy to add a WebRTC video conference streams into a virtual reality environment using WebVR using FreeSWITCH. FreeSWITCH is one of the more popular open source telephony platforms and has had WebRTC for a few years. WebRTC; WebVR; Open Source – obviously this was good webrtcHacks material. ...

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webrtcH4cKS: ~ Suspending Simulcast Streams for Savvy Streamlining (Brian Baldino)

Posted by Brian Baldino on August 6, 2018
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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. ...

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webrtcH4cKS: ~ A playground for Simulcast without an SFU

Posted by Philipp Hancke on July 31, 2018
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You don’t need to fit an SFU opponent when testing simulcast. Image: Hall of Mirrors scene from Bruce Lee’s Enter the Dragon

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. ...

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webrtcH4cKS: ~ Chrome Screensharing Blues – preparing for getDisplayMedia

Posted by Philipp Hancke on June 14, 2018
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What happens when you screen share on a computer that's already sharing your screen

The Chrome Webstore has decided to stop allowing inline installation for Chrome extensions. This has quite an impact on WebRTC applications since screensharing in Chrome currently requires an extension. Will the getDisplayMedia API come to the rescue?

Screensharing in Chrome

When screensharing was introduced in Chrome 33, it required implementation via an extension as a way to address the security concerns. This was better than the previous experience of putting this capability behind a flag which lead to sites asking their users to change that flag… that got those sites an official yikes. ...

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webrtcH4cKS: ~ Smile, You’re on WebRTC – Using ML Kit for Smile Detection

Posted by Gustavo Garcia on June 11, 2018
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Now that it is getting relatively easy to setup video calls (most of the time), we can move on to doing fun things with the video stream. With new advancements in Machine Learning (ML) and a growing number of API’s and libraries out there, computer vision is also getting  easier to do. Google’s ML Kit is a recent example of a new machine learning based library that makes gives quick access to computer vision outputs.

To show how to use Google’s new ML Kit to detect user smiles on a live WebRTC stream, I would like to welcome back past webrtcHacks author and WebRTC video master  Gustavo Garcia Bernardo of Houseparty. Joining him I would like to also welcome mobile WebRTC expert, Roberto Perez of TokBox.  They give some background on doing facial detection, show some code samples, but more importantly share their learnings for optimum configuration of smile detection inside a Real Time Communications (RTC) app. ...

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webrtcH4cKS: ~ Autoplay restrictions and WebRTC (Dag-Inge Aas)

Posted by Dag-Inge Aas on May 7, 2018
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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. ...

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webrtcH4cKS: ~ So your VPN is leaking because of Chrome’s WebRTC…

Posted by Philipp Hancke on April 2, 2018
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Brussel’s Mannneken Pis. Original photo by Flickr user Francisco Antunes (CC BY 2.0)

We have covered the “WebRTC is leaking your IP address” topic a few times, like when I reported what the NY Times was doing and in my WebRTC-Notifier. Periodically this topic comes up now and again in the blogosphere, generally with great shock and horror. This happened again recently, so here is an updated look into this alleged issue.

The recent blog post titled VPN Leak by voidsec highlighting how 19 out of more than 100 VPN services tested “leak” IP addresses via WebRTC is a quite interesting read. Some of the details about WebRTC are not quite correct the results are interesting nonetheless. At is core this is someone who sat down to test a long list of services and their behaviour, one by one. This is not the most exciting research task, but exhaustive studies like this often find something interesting. ...

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webrtcH4cKS: ~ Progressive Web Apps (PWA) for WebRTC (Trond Kjetil Bremnes)

Posted by Trond Bremnes on March 28, 2018
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One of WebRTC’s biggest challenges has been providing consistent, reliable support across platforms. For most apps, especially those that started on the web, this generally means developing a native or hybrid mobile app in addition to supporting the web app.  Progressive Web Apps (PWA) is a new concept that promises to unify the web for many applications by allowing web-based apps to look and act like native mobile ones without introducing an intermediary hybrid framework. As will be discussed, this approach has a lot of advantages, but does it make any sense for a WebRTC app? ...

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webrtcH4cKS: ~ YouTube Does WebRTC – Here’s How

Posted by Philipp Hancke on March 24, 2018
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I logged into YouTube on Tuesday and noticed this new camera icon in the upper right corner, with a “Go Live (New)” option, so I clicked on it to try. It turns out you can now live stream directly from the browser. This smelled a lot like WebRTC, so I loaded up chrome://webrtc-internals to see and sure enough, it was WebRTC. We are always curious here to see how large scale deployments are implemented, so I immediately asked WebRTC reverse engineering master Philipp “Fippo” Hancke to investigate deeper. The rest here is his analysis. ...

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