Introduction to capture handle - a new Chrome Origin Trial that lets a WebRTC screen sharing application communicate with the tab it is capturing. Examples use case discussed include detecting self-capture, improving the use of collaboration apps that are screen shared, and optimizing stream parameters of the captured content.
When running WebRTC at scale, you end up hitting issues and frequent regressions. Being able to quickly identify what exactly broke is key to either preventing a regression from landing in Chrome Stable or adapting your own code to avoid the problem. Chrome’s bisect-builds.py tool makes this process much easier than you would suspect. Arne from Whereby gives you an example of how he used this to workaround an issue that came up recently.
{“editor”, “Philipp Hancke“}
In this post I am going to provide a blow-by-blow account of how a change to Chrome triggered a bug in Whereby and how we went about determining exactly what that change was. ...
Echo cancellation is a cornerstone of the audio experience in WebRTC. Google has invested quite a bit in this area, first with the delay-agnostic echo cancellation in 2015 and now with a new echo cancellation system called AEC3. Debugging issues related to AEC3 is one of the hardest areas. Al Brooks from NewVoiceMedia ran into a case of seriously degraded audio reported from his customers’ contact center agents. After a lengthy investigation it turned out to be caused by a Chrome experiment that enabled the new AEC3 for a percentage of users in Chrome stable.
Al takes us through a recap of how he analyzed the problem and narrowed it down enough to file a bug with the WebRTC team at Google. ...
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?
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. ...
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. ...
Back in October 2013, the relative early days of WebRTC, I set out to get a better understanding of the getUserMedia API and camera constraints in one of my first and most popular posts. I discovered that working with getUserMedia constraints was not all that straight forward. A year later I gave an update after the situation with Chrome was greatly improved, but Firefox at the time effectively only supported a single resolution so constraints were not much help. Specifically, I am interested in understanding what happens when you ask for a specific resolution. You might want to have a specific resolution returned by getUserMedia if you want to match the camera resolution to a specific video area to have a 1 to 1 pixel correlation, in a computer vision application where each pixel represents a distance, or if you are dealing with non-standard video devices. ...
“Only Secure Origins Are Allowed”
– Chrome 47
Chrome 47 now forces secure origins (mostly) with HTTPS. This can be a pain to deal with, but Xander Dumaine is here to help with some guidance. Xander is a Senior Software Engineer who deals with the good and bad of WebRTC for Interactive Intelligence in Raleigh, NC. He is helping maintain simpleWebRTC and organises the Triangle WebRTC Meetup group in that area.
{“editor”: “chad hart“}
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There has been more noise about WebRTC making it possible to track users. We have covered some of the nefarious uses of WebRTC and look out for it before. After reading a blog post on this topic covering some allegedly new unaddressed issues a week ago I decided to ignore it after some discussion on the mozilla IRC channel. But this has some up on a the twitter-sphere again and Tsahi said ‘ouch’, here are my thoughts.
The blog post (available here) makes a number of claims about how certain Chrome behavior makes fingerprinting easier: ...
Let’s have some more fun with getUserMedia by creating a simple mirror application and determining its frame rate.
If your user is going to send their video, it is a general best practice to let them see what they are sending. To do this you simply route the local video stream capture by getUserMedia to a <video> element inside the DOM. That is pretty easy, but the challenge is the video you see is not mirrored. When you are looking at yourself you expect to see a mirrored image. When you don’t it feels off and leads to a poor user experience. Ok, so everyone mirrors their local video, so let’s do that. ...
Ring! Sometimes you need an alert to get your attention. Traditional phone systems make this easy – if someone calls you your phone rings. The traditional telephony model assumes the called device is always on an available to ring and this is how it generally works across analog phones, mobiles, VoIP phones, and even desktop calling replacements like Skype. The challenge in the web model is that you can no longer assume the remote device is available to run your program’s ring command. Even if the called party has a browser open, it does not mean they have a tab running your app. This means you need to find some other means of telling the called party to go to your URL. That can be limiting for a lot of apps. Fortunately there are solutions for this. ...
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