Disruptive Telephony

How does the Jitsi softphone work with IPv6? And what role could DNSSEC play with VoIP? At IETF86 earlier this month, I sat down with Emil Ivov, project leader of the Jitsi Project to talk about a wide range of topics including how Jitsi got started and why it does so much with IPv6 (interesting reason!), what they are looking to do with Jitsi now, the role of DNSSEC and why they added that support to Jitsi... and much, much more... I quite enjoyed talking to Emil and the Jitsi project is certainly one that I will continue to watch - and use!

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In about 15 minutes, at 5:30pm US Eastern At around 6:00pm US EDT, Henning Schulzrinne, CTO of the US Federal Communications Commission (FCC) will be speaking on "The End of Plain Old Telephone System (POTS): Transitioning the PSTN to IP" at the technical plenary of the 86th IETF meeting happening this week in Orlando, Florida.  You can listen or watch here:

• Audio stream – either Caribbean 3 or Caribbean 4
• Jabber chat: [email protected]
• Meetecho conferencing  (webinar)

Henning's slides are also available for download.

It should be quite an interesting session!

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What is new with the Jitsi softphone these days? What new capabilities does it have as it continues to expand its support of SIP, XMPP and other protocols?

I've long been a fan and user of Jitsi, in part because it supports IPv6 and is the only VoIP softphone I know of right now that supports DNSSEC, something I'm continuing to experiment with, so I'm looking forward to today's "VoIP Users Conference (VUC) call at 12 noon US Eastern - about 2.5 hours from now.

You can watch it live via a Google+ Hangout On Air, or call in (potentially using Jitsi!) via:

• sip:[email protected]
• +1 (646) 475-2098
• Skype:vuc.me

There's also an IRC backchannel where links are shared, questions are answered and other comments occur.

And for those of you using Google+, there is a Google+ Event you can join.

It should be a good show! (And yes, you can watch it / listen to it later...)

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Want to learn more about what I'm doing these days? Both with my work with the Internet Society Deploy360 Programme as well as other various projects? And what any of that has to do with VoIP and real-time communications these days?

If so, join us tomorrow, Friday, March 1, at 12 noon US Eastern on the "VoIP Users Conference (VUC)". You can join a Google+ Hangout, or call in via:

• sip:[email protected]
• +1 (646) 475-2098
• Skype:vuc.me

There's also an IRC backchannel where links are shared, questions are answered and other comments occur.

As long time readers know, I've been a huge fan of - and participant in, when I've been able to - the VUC calls and community that Randy Resnick has been spearheading since March 2007.

I don't remember when precisely I first started joining in... maybe sometime in 2007 or 2008. In those early days (and still today) it was a great place to discuss open source telephony solutions like Asterisk, Freeswitch and others. The VUC became a place where many of us gathered weekly to learn about the latest technologies from various projects or vendors... and just to chat about various topics relating to VoIP, Unified Communications, SIP and whatever else. We'd get down in the weeds in really geeky discussions about wideband audio... and then get into higher level discussions about trends in VoIP and UC.

When I was at Voxeo for four years, the VUC calls were a way to connect with the telephony developer community - and in fact there were several VUC sessions related to various Voxeo Labs products and services. There were also a couple of VUC sessions relating to IPv6 and several relating to security in which I was a participant (including discussion of my UC security book at one point). It's been a great community and several good friendships have evolved out of meeting people within the VUC world.

When I joined the Internet Society back in September 2011, my attention shifted rather dramatically away from VoIP as I focused more on IPv6, DNSSEC and now routing resiliency/security (and the drop-off in posts here at Disruptive Telephony was also apparent), but VoIP has always been a strong interest and I continued to dip into VUC episodes now and then.

For the last, oh, 6 or 7 months (or maybe 9 or 10!) Randy's been repeatedly asking me if I would be the guest on a VUC episode to talk about all the various stuff I've been up to. His persistence finally wore me down and I agreed to a date... hence the show tomorrow. :-)

Given the conversational nature of the VUC shows, I suspect we'll probably travel all over various topics. There's an abstract up that says this:

Dan York has been a member of the VUC community for many years now but he’s been away for a bit and in this episode we’ll learn about what he’s doing now and how it relates to VoIP and UC. We’ll talk about his work with the Internet Society Deploy360 Programme related to IPv6, DNSSEC and routing resiliency (and touch on what exactly the Internet Society is all about) as well as his views on WebRTC and some of the other standards relating to real-time communications being developed within the Internet Engineering Task Force (IETF). We’ll also cover Dan’s favorite topic of VoIP security and some of the changes that are being seen there. We’ll probably also talk about some of Dan’s other activities and interests related to podcasting and social media.

How much of that we actually discuss will be an open question.

You're welcome to join us... we always do take questions from people calling in to the show and also via the IRC channel. This time there will also be a Google+ Hangout so people will get to see my smiling face! ;-)

And yes, for better or worse it will be archived for posterity for later listening/viewing...

P.S. If you'd like to join the VUC "community" yourself, a great way beyond attending the calls is to join the "IP Communications & VoIP" Community on Google+, as a lot of good interaction happens there.

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This week the WebRTC/RTCWEB initiative passed a HUGE milestone in adding a real-time communications layer to the Web with achieving interoperability between Google Chrome and Mozilla Firefox. Google and Mozilla celebrated with a pair of blog posts:

• Hello Firefox, this is Chrome calling!
• Hello Chrome, it's Firefox calling!

They also published the video I've embedded below. On the surface, the video doesn't appear terribly exciting: two guys having a basic conversation over video. But consider this:

• The video conversation was initiated from within web browsers.
• There were NO plugins used... no Flash, Java or anything else.
• The entire conversation was securely encrypted.
• The call used "wideband audio" (also called "HD audio") to provide a much richer experience that far exceeds any kind of conversation you can have on traditional telecom and mobile networks.
• The call did not have to involve any external telecom networks or services and could have been initiated directly from one browser to the other. (I don't know exactly how they set up this call.)

And perhaps most importantly:

Any web developer can now create this kind of real-time communication using a few lines of JavaScript and other web programming languages.

As I'm said before, WebRTC will fundamentally disrupt telecommunications and add a real-time communications layer to the Internet that is based on open standards and is interoperable between systems. Creating applications that use voice, video and chat is being removed from the realm of "telecom developers" and made truly accessible to the zillions of "web developers" out there.

Congrats to the Google and Mozilla teams... this is a huge step forward for WebRTC!

You can see the video below... and if you are a developer interested in playing with WebRTC further, both the Google and Mozilla blog posts offer pointers to source code. The team over at Voxeo Labs also released a new version of their Phono SDK yesterday with WebRTC support that may be helpful as well.

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UPDATE #1: The discussion threads on Hacker News related to the Google and Chrome blog posts make for quite interesting reading and provide many additional links for exploration:

• Hacker News thread about Google blog post (long thread about relation to Skype)
• Hacker News thread about Mozilla blog post

UPDATE #2: Over at Forbes, Anthony Wing Kosner weighed in with a similar piece and proved he can write far more poetic headlines than mine: Google And Mozilla Strike The Golden Spike On The Tracks Of The Real Time Web

UPDATE #3: And over on No Jitter, Tsahi Levant-Levi gets the "wet blanket" award for dampening enthusiasm with his post: WebRTC Browser Interoperability: Heroic. Important. And...Expected

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Fascinating news today out of Oracle that they have purchased Acme Packet in a transaction estimated to be around $2 billion US. For those of you not really tracking the VoIP security space, Acme Packet is probably the world's largest vendor of "session border controllers (SBCs)", devices that are used to securely and reliable interconnect VoIP networks. SBCs also provide a very important role in helping with interoperability of Session Initiation Protocol (SIP) signaling between the SIP products and networks of different vendors.

As Andy Abramson writes, the fascinating aspect of this acquisition is this:

This is an interesting grab by one of the tech world's true giants because it sqaurly puts Oracle into a game where they begin to compete with the giants of telecom, many of whom run Oracle software to drive things including SBC's, media gateways and firewall technology that's sold.

This acquisition does put Oracle VERY firmly into the telecom sector at a carrier / large enterprise level, as Acme Packet's products are widely used within that tier of companies. As the news release notes:

"The company's solutions are deployed by more than 1,900 service providers and enterprises globally, including 89 of world's top 100 communications companies."

Acme Packet has also long been recognized as a leader by analyst firms such as Gartner. People from Acme Packet, in particular Hadriel Kaplan, have also been extremely involved with industry efforts such as the SIP Forum and standards activity in the IETF.

As far as integration, Oracle already has a wide array of "communications" products, including several unified communications (UC) products that could potentially interact with Acme Packet products extremely well. Beyond all of that, though, this acquisition will have Oracle being a strong player in providing telecom infrastructure as we continue to collectively move to basing all our communications on top of IP.

Congratulations to my friends at Acme Packet and Oracle... and I wish them the best as they proceed down the path to completing this acquisition.

More information here:

• Oracle news release
• Oracle page about the acquisition
• A letter from Acme Packet CEO

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Are you a vendor of SIP-based products and services? Do you have software or hardware (or cloud-based products) that use SIP? If so, are you planning to attend the next SIPit test event planned for February 18-22, 2013, in Raleigh, North Carolina?

The SIPit events are an outstanding place to test your SIP implementations. Where else will you have so many other vendors also testing their equipment? It's a great place to go, test... and iterate your code even while you are there so that you can test again.

The registration deadline is Feb 4, 2013 for SIPit 30, so you need to act soon if you want to attend.

Olle Johansson posted a great set of slides about why you should go to SIPit:

And reaching back to 2009, here's a video interview I did with Robert Sparks about the SIPit test events:

If you are a vendor of SIP products or services, I would strongly encourage you to consider attending the next SIPit. It's a great way to make sure your SIP works as best as it can.

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Got a great idea for a talk to give to an excellent gathering of SIP/VoIP network operators? Have a new way of handling security? Have a case study you'd like to present for how you solved an operational issue?

The SIP Network Operators Conference (SIPNOC) is an outstanding event happening in Herndon, Virginia, USA, from April 22-25. It brings together network operators working with SIP / VoIP networks for several days of talks, networking (of the human kind) and education. I've gone the past two years, speaking about IPv6, and they are truly excellent conferences. Not too big, not too small... and with an extremely high quality of people both attending and speaking.

If you think you'd like to present, TODAY, January 25, 2013, is the end of the call for presentations for SIPNOC 2013. They are seeking presentations on topics such as (see the CFP for more detail):

• Peering
• SIP Trunking
• Congestion Control
• Applications/content Development
• Interoperability
• Call Routing
• Security
• Monitoring/Troubleshoooting and Operational Issues
• Testing Considerations and Tools
• Availability/Disaster-Recovery
• WebRTC and SIP
• SIP-Network Operations Center Best Practices
• Standardization Issues and Progress
• FoIP/T.38 Deployment
• User-Agent Configuration
• IPv6 Deployment Challenges
• Emergency Services
• Scaling and Capacity Issues
• HD-Voice Deployment Challenges
• Video Interop Issues

They are seeking individual talks, panel sessions, research sessions and BOFs.

Even if you just have an idea for a session, I'd encourage you to submit a proposal so that the SIPNOC 2013 Program Committee will know of your interest and can reach out to you for more details. More info about the process can be found on the CFP page.

If you aren't interested in speaking, but are now intrigued by SIPNOC and would like to be learning from all the excellent sessions, you can go to the SIPNOC 2013 main page and find out information about how to register and attend.

If you work at or for a telecom/network operator who is involved with SIP and VoIP, I highly recommend SIPNOC as a conference you should attend - you'll learn a huge amount and make great connections.

P.S. I have no affiliation with SIPNOC other than being a speaker there in the past. SIPNOC is a production of the SIP Forum, a great group of people focused on advancing the deployment and interoperability of communications products and services based on SIP.

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Facebook today rolled out it's free voice calling in the US via its Messenger app for iOS (iPhone/iPad). The Verge was the first I saw with the news and a great number of sites are now following.

Voice calling through Facebook has the potential to be hugely disruptive... rather than calling on your phone over your regular phone connection - or even rather than using Skype, you can just call from directly within Facebook. This is the kind of "Over-The-Top (OTT)" app that gives telco operators a fit... goodbye, telco voice minutes!

Plus, it's using some HD voice codec so the sound quality is outstanding.

And since the folks at Facebook want you to live your life inside of their very pretty walls, this just provides yet one more reason for you to stay within those walls.

BUT... there's a big huge caveat that I'll get to in a moment.
 

A Quick Walkthrough

First, though, let's look at how it works. When you go into the Messenger app and open a chat with a friend (in this case, Jim Courtney), all you have to do is click the "i" button in the upper right:

After you do that you will get a window that I showed at the beginning of an article where you have a "Free Call" button.

When you press that, you begin a call experience very similar to any other call on your iPhone. First you are connecting to the other person and then you are in the actual call:

There is apparently the standard accept and decline buttons. (I neglected to have Jim call me back to get a screenshot.) While you are in the call you have a button to hang up, a speakerphone button and a microphone mute button. The last button is very nice in that it lets you remain in the call while using other features of your iPhone. In these two screenshots you can see that I could access our Messenger chat and also go back to my main iPhone screen to launch other applications. I can always tap the bar at the top to return to Messenger and the controls to our voice conversation:

The voice quality during the conversation was outstanding. It was crystal clear and rich enough that we knew it was some kind of HD voice codec being used.

All in all it was an excellent experience.

The Big, Huge Caveat

So what's the problem? Well... the reality is that right now trying to find someone to call is a struggle!

Going down through my contacts in the Messenger app was an exercise in futility. Person after person after person had the "Free Call" button greyed out:

Here's the fundamental problem:

You must be running the MESSENGER app on your iPhone!

It doesn't matter if you are running the Facebook application on your iPhone... you must be running Messenger.

And bizarrely there is no linkage between the two applications. If I am over in the Facebook application and go into a chat with Jim Courtney, notice that I have only the ability to "View Timeline":

And of course you must have an iPhone or iPad. If you have an Android device or some other device you are out of luck right now.

So the only people you can use this with are other people running Messenger on iOS.

Presumably Facebook is assuming people will just keep Messenger running... but I know that I, for one, try to limit the number of apps I keep running on my iPhone for battery life reasons.

More fundamentally, I never have used the Messenger app for chatting with other friends in Facebook. The Facebook app already provides the ability to chat... so why would I use the Messenger app? (And I know Facebook focuses on the speed that you can get to sending messages... but that's not critical for me.)

Potential For Disruption?

Now if Facebook gets their act together and makes this more intuitive and ubiquitous, the potential is there for more serious disruption. If it can be integrated into the main Facebook app... and can work for Android as well as iOS... and can work for people outside the US and Canada... THEN we might see more people shifting voice calls over into Facebook's voice service.

The potential is certainly huge, given Facebook's massive size.

Until then... it's an interesting option to have available... but I just don't see many people using it.

What About The Technology Behind It?

My other natural question was to wonder what they are using for the technology behind their voice service. As The Verge pointed out, Facebook and Skype have had a partnership to deliver video calling within Facebook's website. Could this be another component of that partnership? Is it a partnership with another VoIP provider? Is it something homegrown?

For now, I haven't seen any details that help explain that, but I'll certainly be watching to see what we can learn.

UPDATE: A tweet from Aswath Rao pointed me to a TechCrunch article from earlier this month when Facebook rolled out free voice calling in Canada that indicates that the technology is NOT from Skype. Separately I asked a Skype representative if Skype was involved in today's rollout and received the simple answer of "no".

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Want to connect with others interested in the bleeding edge of IP communications and VoIP? Want to exchange links or engage in discussions with people interested in these topics? If you are a Google+ user (as I am), there is now the new "Communities" feature and Randy Resnick of VUC fame has set up a new Google+ community on "IP Communications & VoIP" at:

https://plus.google.com/communities/114149566116254233716

Given that Randy is very active on Google+, this community is also very active, both through Randy's posts as well as the comments and posts of others. I've already learned a good bit from a couple of the discussions that have occurred there.

There are other Google+ communities that you might find interesting, too, such as those related to DNSSEC and IPv6, but Randy's is a great one for VoIP / IP communications / UC topics. Check it out and join in the conversations....

Plus, if you haven't checked out the VUC calls that occur each Friday at noon US Eastern, they, too, are definitely worth listening to and participating in.

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