Imaged modified from Window in Seattle airport by Flickr user Robert Scoble (CC BY-2.0)
Want to keep up on our latest posts? Please click here to subscribe to our mailing list if you have not already. We only email post updates. You can also follow us on twitter at @webrtcHacks for blog updates.
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.
If you are new to WebRTC then you have missed out on years of drama in the standards bodies over various issues like SDP and codecs. These standards dictate what vendors must implement so they ultimately dictate the industry roadmap. To get a deep perspective and appreciation of the issues, we like to ask Dan Burnett, W3C editor to comment on where we are at with the standardization process. I caught up with Dan at this year’s IIT Real Time Communications Conference and had the more detailed Q&A with him shortly thereafter.
We have been waiting a long time for Microsoft to add WebRTC to its browser portfolio. That day finally came last month when Microsoft announced its new Windows 10 Edge browser had ORTC. This certainly does not immediately address the Internet Explorer population and ORTC is still new to many (which is why we cover it often). On the positive side, interoperability between Edge, Chrome, and Firefox on the audio side was proven within days by multiple parties. Much of ORTC is finding its way into the WebRTC 1.0 specification and browser implementations.
Atlassian’s HipChat acquired BlueJimp, the company behind the Jitsi open source project. Other than for positive motivation, why should WebRTC developers care? Well, Jitsi had its Jitsi Video Bridge (JVB) which was one of the few open source Selective Forwarding Units (SFU) projects out there. Jitsi’s founder and past webrtcHacks guest author, Emil Ivov, was a major advocate for this architecture in both the standards bodies and in the public. As we have covered in the past, SFU’s are an effective way to add multiparty video to WebRTC. Beyond this one component, Jitsi was also a popular open source project for its VoIP client, XMPP components, and much more.
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.
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.
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.
Last year we interviewed Oleg Moskalenko and presented the rfc5766-turn-server project, which is a free open source and extremely popular implementation of TURN and STURN server. A few months later we even discovered Amazon is using this project to power its Mayday service. Since then, a number of features beyond the original RFC 5766 have been defined at the IETF and a new open-source project was born: the coTURN project.
Today we are catching up with Oleg again to see what’s new and to learn what coTURN is about.
I’m at the IIT RTC Conference this week in Chicago which is an excellent, no-BS conference that featured many WebRTC luminaries and one of best events I have attended in a long time.
On Tuesday I moderated a panel with WebRTC contributors and ORTC promoters, Robin Raymond of Hookflash, Bernard Aboba of Microsoft, and Peter Thatcher of Google, asking many of the same questions I did on the ORTC Q&A several weeks ago.
Dan Burnett was in the room, asking a lot of questions. If you don’t know Dan, he is a long time W3C author and editor. He is also one of the Godfathers of WebRTC who was there right at the beginning. He also has a highly regarded book on WebRTC coauthored with Alan Johnston.
