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.
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Series preface: We generally lean toward long posts here at webrtcHacks, but not all interesting topics warrant a lot of new text. Sometimes briefer is better. So to better address the many topics that fit into this category, we are starting a new Minimum Duration series. Here is our first post under this set covering Perfect Negotiation.
What is Perfect Negotiation and why do we need it?
Long ago the WebRTC specification designers settled on leaving the signaling communication mechanism between two WebRTC peers up to the application. This means your code needs to handle passing Session Description Protocol (SDP) back and forth and giving that to the peerConnection API. Today WebRTC implementations also almost universally use Trickle-ICE, a form of Interactive Connectivity Establishment (ICE), which passes potential network paths between those peers asynchronously so a connection can be established as soon as possible. The asynchronous but time sensitive nature of all this means it is possible for glare conditions to occur – situations where both sides are making updates at the same time causing their state machines to get out of sync. Differences in how developers implement their code and browsers variances make this worse.
WebRTC has a new browser – kind of. Yesterday Microsoft’s “new” Edge browser based on Chromium – commonly referred to Edgium – went GA. This certainly will make life easier for WebRTC developers since the previous Edge had many differences from other implementations. The big question is how different is Edgium from Chrome for WebRTC usage?
The short answer is there is no real difference, but you can read below for background details on the tests I ran. If you’re new around WebRTC the rundown may give you some ideas for testing your own product.
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 asked Dan to comment on recent spec changes, ORTC, the next version of WebRTC, codecs, Apple, when the 1.0 spec might ever be finalized, and a whole lot more.
{“editor”, “chad hart“}
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.
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.
Last week I attended the Illinois Institute of Technology Real-Time Communications (IIT-RTC) Conference in Chicago. This event has a history of attracting key players from around the RTC world. It features discussion that is distilled down to the key trends and technology challenges in the industry, with very little “fluff” on top. This year the IIT-RTC conference was co-located with IPTComm as well, adding to the quality of the content.
Topics at the conference touched on many segments of RTC, including IMS, RCS, E-911, OTT, and more. Our own Victor Pascual sits on the steering committee for the Web and Emerging Technologies track, where WebRTC was given particular focus. It began with a fantastic WebRTC tutorial from Alan Johnston (co-author of the SIP specification and a dozen other IETF RFCs) and Dan Burnett (co-editor of the W3C WebRTC specification). They are also both co-authors of “WebRTC: APIs and RTCWeb Protocols of the HTML5 Real-Time Web”, and provided a fantastic expert introduction to WebRTC APIs and methodologies. This set the tone for lots of excellent presentation, expert perspective, demonstrations, and discussion on WebRTC over the next few days. Here are some discussions I found particularly interesting:
