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The world of browsers and how they work is both complex and fascinating. For those that are new to the browser engine landscape, Google, Apple, and many others collaborated on an open source web rendering engine for many years known as WebKit.  WebKit has active community with many less well known browsers that use it, so the WebKit community was shocked when Google announced they would fork WebKit into a new engine for Chrome called Blink.

Emphasis for implementing WebRTC shifted with Google into Blink at the expense of WebKit. To date, Apple has not given any indications it was going to add  WebRTC into WebKit (see this post for an idea on nudging them). This is not good for the eclectic WebKit development community that would like to start working with WebRTC or those hoping for WebRTC support in Apple’s browsers.

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Two weeks ago Philipp Hancke,  lead WebRTC developer of Talky and part of the &yet‘s WebRTC consulting team, started a series of posts about detailed examinations he is doing on several major VoIP deployments to see if and how they may be using WebRTC. Please see that post on WhatsApp for some background on the series and below for another great analysis – this time on Facebook Messenger. {“editor”: “chad“}

Last week, Facebook announced support for video chats in their Messenger app. Given that Messenger claims to account for 10% of global mobile VoIP traffic, this made in a very interesting target for further investigation. As part of the series of deconstructions, the full analysis (another fifteen pages, using the full range of analysis techniques demonstrated earlier) is available for download here, including the wireshark dumps.

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One of our first posts was a Wireshark analysis of Amazon’s Mayday service to see if it was actually using WebRTC. In the very early days of WebRTC, verifying a major deployment like this was an important milestone for the WebRTC community. More recently, Philipp Hancke – aka Fippo – did several great posts analyzing Google Hangouts and Mozilla’s Hello service in Firefox. These analyses validate that WebRTC can be successfully deployed by major companies at scale. They also provide valuable insight for developers and architects on how to build a WebRTC service.

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WebRTC-based services are seeing new and larger deployments every week. One of the challenges I’m personally facing is troubleshooting as many different problems might occur (network, device, components…) and it’s not always easy to get useful diagnostic data from users.

troubleshooting (Image source: google)

troubleshooting (Image source: google)

Earlier this week, Tsahi, Chad and I participated at the WebRTC Global Summit in London and had the chance to catch up with some friends from Google, who publicly announced the launch of test.webrtc.org. This is great diagnostic tool but, to me, the best thing is that it can be easily integrated into your own applications; in fact, we are already integrating this in some of our WebRTC apps.

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Apple Feast

Apple Feast photo courtesy of flikr user Overduebook. Licensed under Creative Commons NC2.0.

One of the biggest complaints about WebRTC is the lack of support for it inside Safari and iOS’s webview. Sure you can use a SDK or build your own native iOS app, but that is a lot of work compared to Android which has Chrome and WebRTC inside the native webview on Android 5 (Lollipop) today. Apple being Apple provides no external indications on what it plans to do with WebRTC. It is unlikely they will completely ignore a W3C standard, but who knows if iOS support is coming tomorrow or in 2 years.

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Unnatural shrinkage. Photo courtesy Flikr user Ed Schipul

Unnatural shrinkage. Photo courtesy Flikr user Ed Schipul

One evening last week, I was nerd-sniped by a question Max Ogden asked:

maxodgen-tweet

That is quite an interesting question. I somewhat dislike using Session Description Protocol (SDP)  in the signaling protocol anyway and prefer nice JSON objects for the API and ugly XML blobs on the wire to the ugly SDP blobs used by the WebRTC API.

The question is really about the minimum amount of information that needs to be exchanged for a WebRTC connection to succeed.

 WebRTC uses ICE and DTLS to establish a secure connection between peers. This mandates two constraints:

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A couple of decades ago if you bought something of any reasonable complexity, odds are it came with a call center number you had to call in case something went wrong. Perhaps like the airline industry, economic pressures on contact centers shifted their modus operandi from customer delight to cost reduction. Unsurprisingly this has not done well for contact center public sentiment. Its no wonder the web came along to augment and replace much of this experience –  but no where near all it. Today, WebRTC offers a unique opportunity for contact centers to combine their two primary means of customer interaction – the web and phone calls – and entirely change the dynamic to the benefit of both sides.

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I was starting to work with a big dataset and was dreading the idea of bogging down my machine with MySQL or SQLServer, so I decided to give Google’s BigQuery a try. Before I got into my project, I was pleasantly surprised to a public GitHub dataset was readily available. Tsahi does a cursory analysis of WebRTC projects on GitHub by manually counting search results every month. I was also inspired by Billy Chia‘s great NoJitter post analyzing WebRTC topics on Stack Overflow.

I was curious to see if I could extract some details from GitHub to see:

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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.

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Sorry. We really wanted to do a post-cap of the W3C WebRTC and IETF RTCweb meetings that took place at the end of October and November, but we did not get to it. Victor and Reid provided some commentary on the codec debate prior to the IETF discussion. The outcome of that discussion was widely publicized and we did not have a lot of value to add to this for the developer community.

Importantly, codecs were not the only thing discussed in this latest rounds of standards meetings. There were a couple items like the move to JavaScript promises, output device enumeration, and discussions of security implications that are very relevant to the average WebRTC developer that have gone under the general media radar. To get the whole on standards right from the horse’s mouth, I asked W3C WebRTC editor and founding author Dan Burnett for an update on the recent WebRTC standards meetings and for some details on some of the more significant issues like promises and screen sharing.

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