Keeping It Realtime

One of the most talked about presentations at RealtimeConf 2013 was yeti Lance Stout and Philipp Hancke's demonstration of WebRTC signaling over XMPP, resulting in a federated video call within the browser.

This demo caught the attention of The VoIP Users Conference, or VUC a weekly, live discussion about all the telephony things, which began in 2007. Lance and Philipp will be joining the VUC community on December 27 at noon (Eastern Time) to discuss the potential of WebRTC as an interoperable tool to communicate within established protocols.

Philipp and Lance have been working together for some time on projects (strophe.jingle and Stanza.io, respectively) that push forward the ability of developers to utilize XMPP in tandem with the web, specifically WebRTC.

Join their conversation two weeks from today on Friday, December 27, at the VoIPUsersConference.org, or track it on their Google+ community.

Today, yeti Henrik Joreteg, will be discussing WebRTC on the weekly podcast, The Web Ahead. The hour long show allows host Jen Simmons the opportunity to chat one on one with each week's guest on the burgeoning technologies and platforms that are pushing the web forward.

Henrik is author of the popular library, SimpleWebRTC and lead developer for Talky, a tool we built for simple video chat and screen sharing. He's also heading up the effort to push WebRTC forward at IsWebRTCReadyYet.com.

Henrik first spoke on WebRTC at JSConf Brazil and built AT&T's WebRTC focused att.js, which was showcased by AT&T at CES earlier this year. He spoke recently about WebRTC, first on a Realtime Data panel at EdgeConf this past September, and again at RealtimeConf 2013, in Portland a few weeks ago. Henrik will also be speaking at Cascadia.js later this month.

Go to 5by5's website to hear today's podcast with Henrik and Jen.

People have a very special distaste for telecoms.

Between wiretapping, egregious data roaming and text messaging charges, the unreadability of our phone bills, and carrier lock-in, we have plenty of reasons. Add in the fact that we consider ourselves entirely dependent on them and we're all the more resentful and cynical.

But forget the traditional telecoms for a minute.

Google, Microsoft, Apple, and Facebook are truly the new telecoms—each within the last few years has built or bought a communication platform of their own.

Wouldn't it be cool if someone built a federated, interoperable, open source, and super hackable "Skype killer" built on WebRTC and XMPP and released it at RealtimeConf?

I had the privilege to attend EdgeConf 2013 as a panelist and opening speaker for the Realtime Data discussion.

It was an incredible, deeply technical conference with an interesting discussion/debate format.

Here's the video from the panel:

First—some important news: We've changed the dates.

Early this year, we announced dates for RealtimeConf, RedisConf, and a brand new, WebRTC-focused event. A short time later, our friends at LxJS announced their dates—which were unfortunately the same. We immediately reached out, discussed the mix-up, and determined that we were in the best position to change our dates.

After getting back to the states after being part of a wonderful first-time RealtimeConf EU smashingly organized by Julien Genestoux, we've reset dates and, in the meantime, we've made tremendous headway on conference planning.

Last year, we had a full week of Realtime events and we're just ramping that up a notch, by adding a full day focused on the most exciting realtime technology to hit the web: WebRTC.

Today, we're open sourcing signalmaster.

Paired with SimpleWebRTC.js, this is all you need to set up and serve WebRTC video calls.

Happy WebRTCing!

We're announcing two WebRTC projects today: SimpleWebRTC.js and conversat.io.

WebRTC is one of the most exciting additions to HTML5, enabling direct peer-to-peer connections for video and audio streaming. We’ve been playing with WebRTC for almost a year now, including helping clients like AT&T put it into play.

One conclusion we've come to is that WebRTC should be easier for developers to work with if it's going to gain more adoption. More people should be playing with this new amazing technology, but there's a lot of annoying complexity when working with it. As Mikeal Rogers put it, "It’s about 10x more complicated than WebSockets, and it’s taken 3 years to be where we are with WebSockets."

SimpleWebRTC.js

One of the main focuses of AT&T's Developer Summit at CES in Las Vegas this year was WebRTC.

The AT&T WebRTC API truly untethers user phone numbers from their mobile device. When your phone number on your mobile rings, so can your browser and your tablet. It's the kind of thing that could make answering your phone or making a phone call from anywhere as easy as checking your email is today.

&yet was privileged to play a central role in AT&T's work with WebRTC this year, alongside other partners like Ericsson and Voxeo Labs.

&yet has teamed up with the AT&T Foundry in Palo Alto to deliver several projects this past year. Some were shared at CES and most of those are featured at js.att.io and on AT&T's GitHub account. (Note that at this time, you have to be a registered AT&T Alpha Developer in order to use the API.)