Disruptive Telephony

Microsoft accomplished something today they haven't done for a while (at least in my memory) - they dominated the main page of Techmeme and had a great amount of the tech media talking about them.

The news, of course, is of Microsoft's purchase of Nokia's Devices and Services business and licensing of Nokia's patents and mapping services.

Is anyone truly surprised by this?

Consider:

• Microsoft is being beaten in the market by Apple and Google as everything moves to mobile. Their only hope was Nokia, who provided a hardware platform that would run Windows Phone.
• Nokia is being beaten in the market by Apple and Google as everything moves to smartphones. Their only hope was Microsoft, who provided a different mobile operating system for their devices that gave them a competitive angle.

Given those conditions, the marriage makes a certain amount of sense.

But... you only have to scroll down that Techmeme page (captured at 1:30pm US ET today) to realize how desperate a situation this is for both companies.

First, news is out that Apple is holding an event one week from today on September 10 where they are widely expected to announce new iPhones, including potentially a lower cost iPhone 5C. They are also expected to announce a release date for iOS 7 ... and who knows what else is in store.

Second, Google announced the next version 4.4 of the Android operating system, named "KitKat", along with a branding deal with Nestle, makers of the KitKat candy. The first link also points to a Google+ post from Google's Sundar Pichai where he states that over 1 billion Android devices have been activated.

Third, Amazon announced the 6th generation of their Kindle, and while it is not a phone, per se, it is a massively used mobile device. Amazon continues to learn and evolve their devices and has been rumored for years to be contemplating entering the smartphone space. Jeff Bezos thinks in the long term and so could easily be biding his time.

Meanwhile, Nokia sold a whopping 7 million Windows phones last quarter (per IDC).

Microsoft and Nokia need each other, if for no other reason then they don't really have a choice. They bet on each other... and it doesn't seem to be working out so well. Their only hope is really the "synergy" or whatever other marketing buzzwords you want to apply to the merged entity.

I agree with much of what Om Malik wrote today, "Why I think the $7.2 billion Microsoft-Nokia deal is a terrible idea", largely for the reasons I wrote earlier... while Microsoft and Nokia work to make this deal happen - and then the actual integration - Apple, Google, Amazon and others will be rolling out the next versions of their massively successful mobile devices.

Microsoft's "Strategic Rationale" document lays out a glowing plan... let's see if they can execute on it - and whether it turns out to be too little, too late. I wouldn't completely count Microsoft out, as they do have great resources and capacity, but they are definitely far behind.

As a consumer, I definitely would like to have a third major ecosystem for mobile devices. The question is whether Microsoft/Nokia can emerge as that third ecosystem...

What do you think? Smart move? A yawn? Or the proverbial rearranging deck chairs on the Titanic?

P.S. The most entertaining take on today's news definitely has to be the "Dear MR NOKIA!" post written in the style of the emails probably all of us have received. :-)

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Ten years ago today, on August 29, 2003, a group of entrepreneurs and developers from Denmark, Sweden and Estonia unleashed a small software program that would fundamentally and irrevocably disrupt telecommunications and just communications in general. Everything would change. Skype has a 10th anniversary blog post out today that highlights some of those changes that have been brought about by Skype, although I personally find their 9th anniversary infographic a bit more interesting because it traced back the history of Skype.

Massive Disruption

There is a GREAT amount for Skype to celebrate on it's 10th birthday. The disruption that has occurred within telecom is truly massive:

• The cost per-minute of international phone calls has been commoditized to near zero. (Indeed, how many people actually make real "phone calls" internationally anymore?)
  
  
• Telcos - and governments! - who depended upon those per-minute fees have seen almost that entire revenue stream evaporate, or at least show that it is rapidly fading away. Economic disruption on a massive scale!
  
  
• Skype came to be a prime example of how "over-the-top (OTT)" apps could exist on top of the existing telecom networks - and take both marketshare and revenue from those networks.
  
  
• Skype introduced the masses to high quality audio and helped people understand that the "phone quality" they were used to was actually really poor quality and that they could do so much better... that they could have the experience of "being there" with someone else.
  
  
• Video calls, while they had been around for quite some time in many other apps and devices, were made available to everyone for free using the easy Skype user interface (and were helped by the rise of ubiquitous webcams embedded in laptops and mobile devices). An entire industry around video-conferencing was disrupted through the simple combination of Skype and webcams.
  
  
• Long-distance audio and video collaboration became extremely routine. Think of the thousands of podcasts that use Skype between contributors? (Such as my own Blue Box or the FIR podcast to which I contribute.) Think of the number of video news reports you have seen coming in over Skype... or the guests coming in to talk shows.
  
  
• Skype demonstrated that you could have secure, encrypted phone calls and IM chats, at least with the pre-Microsoft peer-to-peer architecture. They enabled those of us advocating for more secure phone connections to be able to go to other vendors and say "Really? You can't do secure calls... but Skype can?
  
  
• Speaking of that p2p architecture, it, too, was something new and fascinating... perhaps one of the most innovative things to hit telecom in ages... that showed that you could think differently about how to connect endpoints.
  
  
• Curiously, Skype also demonstrated the incredible power of persistent group chats in creating a system that enabled people to continually participate in conversations, even as they came and went from the network. Skype chats still to this day are better that most every other system out there.
  
  
• Skype showed the power - at least in their earlier versions - of focusing on creating an extremely simple user interface and focusing on the user experience. The simplicity of using Skype was a large part of why so many people started to use it. That and the fact that Skype "just works" from behind most firewalls and in most network environments.
  
  
• Along the way, Skype built up a massive directory of users... estimated at 300 million now. Most people I interact with do have a "Skype ID" and those names are exchanged at conferences, printed on business cards, listed on websites and generally made available.
  
  
• Skype became a verb. It's routine now to say: "Let me skype you.", "Can you skype me?", "Let's skype", etc. We don't "call", we "skype".

At a fundamental level, Skype rocked the world of telecom and enabled so much more communication to happen all over the world. As a frequent global traveler, Skype has been such an incredible means by which I can keep in daily touch with my family back at home.

Skype has indeed MUCH to celebrate on it's 10th birthday.

And Yet...

And yet as Skype turns ten, I find myself wondering what the next 10 years will be like... and whether Skype will remain relevant.

You see, that list of disruptions I wrote above is pretty much the same list I wrote about two years ago on Skype's 8th anniversary, just with updated numbers.

What happened in the last two years?

Last year on the 9th anniversary I was asking the "what comes next?" question and Jim Courtney was similarly saying "whither Skype?" Phil Wolff was asking "is Skype boring?", a theme I picked up on for my own post.

Fast-forward a year and the questions are still relevant. Skype is no longer the "bright shiny object" that so many of us were so incredibly passionate about. Indeed, for so many years Skype was the single biggest topic I wrote about here on Disruptive Telephony. There was a reason that my phone number became associated with Skype and I was getting all sorts of calls for Skype's corporate office.

And yet... how many posts did I write here on this blog about Skype in the last year since the 9th anniversary?

One.

Just one post... and at that a short and simple post about a new Skype version being available for the iPhone/iPad.

That's it.

Now, there's a larger issue that I'm simply not writing as much here on Disruptive Telephony as I used to, given that my energy these days is focused so much more on the worlds of IPv6, DNSSEC and Internet routing. But still... had something struck me as exciting or useful about Skype, I probably would have written about it.

I still use Skype each and every day - or at least every work day - and it is a critical part of my day when I'm traveling. But the reality is now for me:

Skype is just a tool.

That's it. A tool to be used. A tool to be expected to be there.

In one way, that's a massive success for Skype, in that millions of people now just expect Skype to be there and to be able to help them communicate.

But it's no longer anything to get excited about. It's a tool. Nothing more. Nothing less.

In a chat earlier today about this feeling shared by a number of us, Phil Wolff, long-time editor of the Skype Journal, said this (reprinted here with his permission):

Skype is boring, like electricity. The BBC interview that came out today where Skype said they'd done proof-of-concept 3D video chat in the lab says it all.

They are busy working on customer acquisition (the next billion users), usage (more conversation per user per month),  and more Microsoft integration (% of MSFT products with Skype inside, a la Outlook.com).

They are busy getting more than 2000 employees to work together, nearly half on the job less than a year.

They are figuring out how to make money when the price of minutes - even international PSTN minutes - are falling faster than Skype can pick up share.

They are learning how to stay relevant in a universe where talk is a feature anyone can add to any app for free/cheap.

Bigger scale usually means innovation on plumbing moves faster than innovation on user experience. Skype hasn't offered up new experiences as shiny as "now with video!" in a long time.

Phil's second-to-last comment is particularly relevant as we think about WebRTC and how much it has opened up the world of voice, video and chat to so many more developers.

What Could Have Been...

Stuart Henshall, an original founder of the Skype Journal and someone promoting Skype for pretty much all of its 10 years, said something similar in a post today, "Skype’s First Decade – A Wasted Opportunity. He sums up rather well how I think some of us who were early adopters of Skype now feel:

Today Skype is a feature, part of Microsoft. While it may generate substantial dollars it isn’t the company it could have been. Skype was one of those once in a lifetime products that today could have been revolutionizing how the world evolves. It was once secure, outside the reach of the NSA. It had the network and the membership so it could have been a Facebook, or a Twitter. It had strong developer support in the early days and it’s own store. Most of us still use Skype some of the time today. It is still the most universal free calling solution. It works across platforms including the PSTN, PC, Mobile. And yet Skype today is a brand without a “soul”. That’s what you get when you sell-out one too many times and lose a passion for changing the world.

Ten years in, Skype went from being the scrappy, interesting, exciting underdog challenging the telecom infrastructure... to in fact becoming that telecom infrastructure to the point where they can't innovate as much as they once did because they do have such an enormous installed base.

Ten years ago, Skype was the disruptor - now the question is if Skype will be disrupted by all the new entrants. Maybe Skype still has some innovation in store and may surprise us... but I'm doubtful at this point.

There is a lot to celebrate in 10 years of Skype, but the question is really whether Skype is today coasting on the innovation of those earlier years and now the increased integration with Microsoft products.

As Stuart wrote:

In Internet years like dog years Skype has had a good run. Still it’s aged some and I know it’s no longer my primary communication method. If I had one wish I’d love to see another Skype P2P like system take root although this time on mobile resetting the rules for the telecom stack. That’s still something I could promote.

I, too, would like to see some system that was truly innovative and brought back many of those innovations of how Skype used to be - and did once again truly disrupt telephony as we know it.

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Can we create a "secure Caller ID" for IP-based communications, a.k.a. voice-over-IP (VoIP)? And specifically for VoIP based on the Session Initiation Protocol (SIP)? Can we create a way to securely identify the origin of a call that can be used to combat robocalling, phishing and telephony denial-of-service (TDOS) attacks?

That is the challenge to be undertaken by the "Secure Telephone Identity Revisited (STIR)" group meeting tomorrow morning, July 30, 2013, at 9:00 am in Berlin, Germany, as part of the 87th meeting of the Internet Engineering Task Force (IETF). The meeting tomorrow is a "Birds Of a Feather (BOF)", which in IETF language is a meeting to determine whether there is sufficient interest to create a formal "working group" to take on a new body of work within the IETF. The proposed "charter" for this new work begins:

Over the last decade, a growing set of problems have resulted from the lack of security mechanisms for attesting the origins of real-time communications. As with email, the claimed source identity of a SIP request is not verified, and this permits unauthorized use of source identities as part of deceptive and coercive activities, such as robocalling (bulk unsolicited commercial communications), vishing (voicemail hacking, and impersonating banks) and swatting (impersonating callers to emergency services to stimulate unwarranted large scale law enforcement deployments). This working group will define a deployable mechanism that verifies the authorization of the calling party to use a particular telephone number.

The agenda for tomorrow's STIR meeting begins with a presentation by Henning Schulzrinne, now CTO of the US Federal Communications Commission (FCC) but also a long-time IETF participant and one of the co-authors of the original RFC 3261 specification for SIP. Henning will be laying out the problem statement and there will be a discussion of the proposed scope of the IETF work. He'll be followed by presentations of potential solutions by Jon Peterson, Eric Rescorla and Hadriel Kaplan and then a discussion of the proposed charter and the work to be done. Given the intense debate that has occurred on the STIR mailing list over the past weeks I expect tomorrow's session to be one where some points will receive a great amount of passionate debate and discussion. (If you are interested in listening in or participating remotely in tomorrow's STIR meeting, see the information later in this article.)

Revisiting Previous SIP Identity Work

As some background, the Internet Architecture Board (IAB) laid out some of the challenges to "secure origin identification" in IP-based communication last November and took a very high-level look at the overall issue. Next, in preparation for what became this STIR effort, Jon Peterson, Henning Schulzrinne and Hannes Tschofenig authored a draft problem statement and requirements document.

The "Revisited" part of the group name is a nod to the fact that this whole issue of asserting "identity" has been explored within the SIP community in the past. Way back in 2006, RFC 4474 defined what has been called "SIP Identity" and provided a method for cryptographically signing certain SIP headers to identify the origin of a call. Unfortunately, RFC 4474 turned out not to work well with the way SIP was actually deployed and so usage has been virtually non-existent. An effort to update that document, what is called "RFC4474bis", has also been proposed and some of those ideas may be incorporated into the new proposed work for the STIR group.

There have also been other efforts such as the "P-Asserted-Identity (P-A-I)" defined in RFC 3325. The challenge here, though is that theoretically P-A-I is supposed to be limited to usage within a trusted network, although in practice it may be seen by other networks. There have also been several efforts to define or document identifiers for billing purposes (including my own P-Charge-Info) although these efforts are trying to solve a slightly different problem.

The point here really is that the STIR effort is drawing upon a rich body of "SIP identity" work that dates all the way back to some early drafts in 2002. Much thought has been given to this issue and many of the people involved with STIR have also been involved with earlier efforts and understand well some of the challenges faced by that past work.

An Important Difference

One important difference between STIR and earlier "SIP identity" efforts is that initially the STIR effort is only focused on telephone numbers. The draft charter explicitly states this:

As its first work item, the working group will specify a SIP header-based authorization mechanism to verify the originator of a SIP session is authorized to use the claimed source telephone number, where the session is established with SIP end to end. This is called an in-band mechanism. The mechanism will use a canonical telephone number representation specified by the working group, including any mappings that might be needed between the SIP header fields and the canonical telephone number representation.

and later:

Expansion of the authorization mechanism to identities using the user@domain form deferred since the main focus of the working group is to develop a solution for telephone numbers.

Previous "identity" work was also undertaken to include a "SIP URI" or "SIP address" and while the ultimate STIR mechanism (or a variant thereof) might also work for SIP URIs, the focus in this initial work is all around securing the origin identification of telephone numbers.

This initial focus makes a great amount of sense given that so much of the SIP traffic today is a result of telecom service providers moving their regular calls to telephone numbers off of the legacy PSTN networks and over to IP networks where they use SIP. Additionally, a great amount of the "problem" traffic seen in VoIP today can be created by attackers who use simple VoIP software to generate their calls to regular telephone numbers.

Remotely Participating In Tomorrow's STIR BOF

If you are interested in participating in the meeting (or at least listening in) on Tuesday, July 30, the meeting will go from 9:00 - 11:30 local time in Berlin, Germany. Berlin is in Central European Summer Time (CEST) which is UTC+2 (and 3:00 am US EDT / midnight US PDT for my friends back in the USA).

You can hear the audio stream at:

• http://ietf87streaming.dnsalias.net/ietf/ietf874.m3u

You can also join the Jabber chat room at:

• xmpp:[email protected]

The slides and other meeting materials can be found at (and note that materials may not be uploaded until shortly before the session and so you may need to refresh your browser):

• http://datatracker.ietf.org/meeting/87/materials.html#stir

Alternatively you can use the "MeetEcho" conferencing system that integrates the audio, the slides and the Jabber chat room at:

• http://www.meetecho.com/ietf87/stir

More information about participately remotely can be found on the IETF 87 Remote Participation page.

To get the most out of the meeting, you'll also want to read these three Internet Drafts that will be part of the solutions being discussed:

• draft-peterson-secure-origin-ps
• draft-rescorla-stir-fallback
• draft-kaplan-stir-cider

.... and be prepared for what should be a LIVELY discussion!

If you are unable to participate remotely, the session will be recorded and you will be able to listen to the archived audio stream, view the Jabber chat logs and also playback the MeetEcho recording.

Getting More Involved

Beyond listening to tomorrow's BOF session, the best way to get involved - either to actively participate or to at least monitor the effort - is to join the STIR mailing list at:

https://www.ietf.org/mailman/listinfo/stir

The list is open to anyone to join. There are no membership or corporate requirements or fees - anyone with an email address may participate.

WARNING! - As can be seen in the list archive, there is currently a large volume of discussion and it will probably continue for some time. If you do join the mailing list you may want to consider setting up rules to sort the STIR email into a folder - or just prepare for the volume to be added to your inbox.

The other way to be involved is to monitor and read the documents that are created for the STIR effort. Newer documents are being created with "stir" in the document name and so they can be easily found at:

http://datatracker.ietf.org/doc/search/?name=stir&activedrafts=on

Other documents that are useful to understand this effort are linked to earlier in this article and can also be found in the text of the proposed STIR charter. After tomorrow's STIR BOF session there will be more information about how the effort will proceed within the IETF. The meeting tomorrow should result, I expect, in the recommendation to go ahead with formally creating a working group and undertaking this work, but we'll see what outcome occurs.

Can a method of secure origin identification for SIP-based VoIP calls be created? Given that basically all telecom traffic is in the process of moving to be based on IP, the need for a secure origin identifier is very clearly here - and many of us do believe we can develop a system that will work in today's environment.

What do you think? Are you ready to join in and help?

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Update: Added the additional charter text about "Expansion of the authorization mechanism to identities..."

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Want to learn more about the Opus codec and why it is so important? As I mentioned at the end of my last post about why Opus matters, there will be a special presentation about Opus as part of the IETF 87 Technical Plenary happening in about 2 hours starting at around 17:45-18:00 in Berlin, Germany (Central European Summer Time, UTC+2, 6 hours off of US Eastern time).

There are three options for watching and participating live:

• using a WebRTC-capable browser (latest editions of Chrome and Firefox) and connecting to: http://www.meetecho.com/ietf87/tech_plenary
• listening to the audio stream at for either Potsdam 1 or Potsdam 3 (the plenary is in the combined room and I don't know which stream will be used)
• watching a video live stream at: https://new.livestream.com/internetsociety/ietf87techplenary

The technical plenary begins at 17:40 but there are some other reports before the Opus section. The agenda can be found online and includes:

1. IAB Chair Report
2. IRTF Chair Report
3. RSE and RSOC Chair Report
4. Technical Topic: Opus Codec

a. Introduction
b. Overview of Opus
c. Testing
d. History of Opus in the IETF
e. Opus Deployment Panel
f. Future Work

5. Open Mic

I suspect that the Opus session will begin closer to 18:00 local time, but you can tune in around 17:40 to see the start of the session.

It should be quite an interesting session!

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What makes the Opus codec so interesting? Why is there such a buzz about Opus right now? If you are not in telecom or doing anything with audio, why should you even remotely care about Opus?

In a word...

Innovation!

And because Opus has the potential to let us communicate with each other across the Internet with a richer and more natural sound. You will be able to hear people or music or presenters with much more clarity and more like you are right there with them.

Opus can help build a better user experience across the Internet.

You see, the reality is that today "real-time communication" using voice and video is increasingly being based on top of the Internet Protocol (IP), whether that communication is happening across the actual Internet or whether it is happening within private networks. If you've used Skype, Google+ Hangouts, any voice-over-IP (VoIP) softphones, any of the new WebRTC apps or any of the mobile smartphone apps that do voice or video, you've already been using IP-based real-time communication.

Dropping The Shackles Of The Legacy PSTN

Part of the beauty of the move to IP is that we no longer have to worry about the constraints imposed upon telecom by the legacy Public Switched Telephone Network (PSTN). Chief among those constraints is the requirement to use only part of the sound frequencies we can hear. You all know the "sound" of the telephone - and you hear it in any movie or TV show when someone is using the phone. It's that certain "sound" that we are all used to... that's what the "phone" sounds like.

In technical terms, we call this "narrowband" audio and it has a frequency range of only 300-3400 Hz.

There are historical reasons for this limitation in telecom, but moving to IP-based communications removes those limits. With VoIP we can use what is called "wideband" audio to have a full rich sound to our voice or video call.

Have you had a really good Skype connection with someone where it sounded like they were almost right there in the room with you?

That is wideband audio.

The Codec Problem

Now, for voice or video over IP to work, you need to use something called a "codec" to translate the sound of your voice to digital bits and carry them across the network (and to do the opposite for whomever you are speaking with). There are MANY audio codecs out there and they come in all sorts of flavors and with all different kinds of capabilities. The problem has been that there hasn't been a codec that:

1. is optimized for interactive Internet applications;
2. is published by a recognized standards organization; and
3. can be widely implemented and easily distributed at little or no cost.

In particular that last point about the cost of licensing, especially for wideband codecs, often caused developers to shy away from giving us the rich voice quality that we can now have with IP. Or, in the case of companies like Skype or Google, they went out and bought companies who created wideband codecs so that they could use those codecs in their products. (See my story from 2010 about Google buying GIPS.)

Now there are free codecs out there that developers can use. For narrowband, there has been the ubiquitous G.711 which provides an IP version of "PSTN audio". There have been many others, including notably Speex.

But the struggle has been that there hasn't been a widely accepted "G.711 for wideband" equivalent that developers can just bake into their products and start using. Instead there have been a number of different, incompatible codecs used in different products.

Enter Opus...

So to address these points, back in 2010, engineers within the IETF got together and formed the CODEC Working Group to come up with a codec that could meet these requirements and become the ubiquitous wideband codec used across the Internet. Skype was involved early on through contributing their SILK codec. The folks at Xiph.org contributed their CELT codec. People from many other companies got involved and there were huge technical discussions on the mailing lists and at IETF meetings.

And it worked... the Opus codec was standardized in RFC 6716 in September 2012.

You can read all about the codec at:

http://www.opus-codec.org/

The key points are at the beginning:

Opus is a totally open, royalty-free, highly versatile audio codec. Opus is unmatched for interactive speech and music transmission over the Internet, but is also intended for storage and streaming applications.

Open, highly-versatile... and royalty-free.

At that site there is some great information, including:

• examples of Opus compared to other codecs
• a comparison of Opus to other audio codecs with a variety of charts and graphics
• a presentation by Jean-Marc Valin that dives way down into the technical details
• documentation and sample libraries

There is also a FAQ and many other great pieces of information.

So Why Does Opus Matter?

Opus matters because it lets developers focus on creating a high quality user experience and not having to worry about codec incompatibilities and licensing issues.

Opus matters because it lets developers easily create applications with high quality audio. They can just start using available libraries and communicating with other applications and devices using a common wideband codec.

Opus matters because it can work in very low-bandwidth environments enabling real-time communications across Internet connections that might not previously have supported such communications. As we start to get more Internet connectivity out to the 5 billion people not yet on the Internet, the ability to work over different kinds of connections is critical.

Opus matters because it can help foster innovation in applications and the user experience. Opus is the default audio codec for WebRTC, and so all the zillion new WebRTC-based apps and startups are already beginning with a far superior audio experience than we've had before.

Opus matters because it will enable even more ways that we can connect with family members or friends and have the experience of being "right there". It can help musicians collaborate better across the Internet. It can help podcasters and journalists deliver higher quality interviews across the Internet. It can, in the best conditions, give us that rich audio experience we get when we are right with someone - even though we may be thousands of miles away.

Opus can help us deliver on the potential of the Internet to create more powerful user experiences and to help us better communicate.

THAT is why Opus matters.

Learn More At Monday's IETF 87 Technical Plenary

To understand more about the current status of Opus, who is using it and where it is going, the IETF 87 Technical Plenary on this coming Monday evening in Berlin, Germany, will have a special segment focused on Opus that will include a number of people involved with the Opus work. The agenda for the session can be found at:

http://trac.tools.ietf.org/group/iab/trac/wiki/IETF-87

It is happening from 17:40-19:40 Berlin time, which is Central European Summer Time, which is currently UTC+2 and 6 hours ahead of where I live in US Eastern time. If you can't be there in person, there are several remote options:

• using a WebRTC-capable browser (latest editions of Chrome and Firefox) and connecting to: http://www.meetecho.com/ietf87/tech_plenary
• listening to the audio stream at for either Potsdam 1 or Potsdam 3 (the plenary is in the combined room and I don't know which stream will be used)
• watching a video live stream at: http://new.livestream.com/internetsociety (Look for the "Technical Plenary" event)

If you are unable to watch the meeting in real time it will be archived for later viewing.

The first option above to listen to the session using the Opus codec (and WebRTC!) is a very cool one. The panel also includes people who have actually implemented Opus including people from Google and also Emil Ivov from the Jitsi softphone. Their insight into what they did will be great to hear.

What's Next?

So if Opus is so great, how do you get it?

Well, if you are using any of the WebRTC apps popping up all across the Internet, you are already using Opus. As I noted above, the Jitsi softphone supports Opus. In an interesting bit of synchronicity, I noticed that Michael Graves wrote today about the Blink softphone now supporting Opus. More and more communications apps are starting to implement Opus.

If you are a developer of communications apps or services (or a product manager), you can look at how to incorporate Opus into your application or service. There is documentation and software available to help with the process, and many people are out there who can help.

If you are a user of IP-based communications apps or services, ask the company or vendor behind those services when they will support Opus. See if you can get it on their radar as something to implement.

And regardless of what you do with audio, let people know that this new way of communicating exists - help spread the word about Opus - let people know that audio across the Internet can be even better than it has been to date.

As you can tell, I'm excited about the potential - and very much looking forward to seeing what happens as Opus gets more widely deployed.

What do you think? If you are a telecom developer, or a vendor of such services, have you implemented Opus already? Are you thinking about it? (and if not, why not?)

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Seeking to understand the basics of WebRTC and why there is so much interest in it? There is a new July 2013 issue of the IETF Journal out this week that includes an article I wrote titled "WebRTC: Moving Real-Time Communication into the Web Browser" that looks at WebRTC from a high-level user perspective.

My aim with this IETF Journal article links was to summarize some of the links on my my WebRTC/RTCWEB page and is admittedly similar in style to my 2012 post, "How WebRTC Will Fundamentally Disrupt Telecom (And Change The Internet)", although this newer article focuses on the work happening within the IETF and provides links to get more involved.

On that note, the RTCWEB working group within the IETF will be meeting next week in Berlin (twice, actually) and has an agenda for IETF87 focused primarily on security questions and looking at the "data channel" aspect of WebRTC/RTCWEB. It should, as always, be an interesting session to listen in to.

If you can't get to Berlin, there are audio streams you can listen to remotely and a Jabber chat room where you can raise questions. Links to both can be found on the top of the agenda page. Do keep in mind that the times listed are local to Berlin, Germany.

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Want to understand more about WebRTC and where it is going? Want to see some demos of new WebRTC apps? At the recent INET event in Bangkok, Thailand, Dr. Cullen Jennings, one of the co-chairs of the IETF's RTCWEB Working Group, gave an excellent presentation that walks through the basics of WebRTC and provided some demos as well:

The presentation is about an hour and is followed by a question period. Well worth watching if you want to understand the current state of WebRTC and how it may impact telecommunications today.

Note, you can also view the video directly on YouTube to better see it in a larger size or on a mobile device.

P.S. For more information about WebRTC, see the links off of my WebRTC/RTCWEB page.

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Do "smart" parking meters really need phone numbers? Does every "smart meter" installed by electric utilities need a telephone number? Does every new car with a built-in navigation system need a phone number? Does every Amazon Kindle (and similar e-readers) really need its own phone number?

In the absence of an alternative identifier, the answer seems to be a resounding "yes" to all of the above.

At the recent SIPNOC 2013 event, U.S. Federal Communications Commission CTO Henning Schulzrinne gave a presentation (slides available) about "Transitioning the PSTN to IP" where he made a point about the changes around telephone numbers and their uses (starting on slide 14) and specifically spoke about this use of phone numbers for devices (slide 20). While his perspective is obviously oriented to North America and country code +1, the trends he identifies point to a common problem:

What do we use as an application-layer identifier for Internet-connected devices?

In a subsequent conversation, Henning indicated that one of the area codes seeing the largest amount of requests for new phone numbers is one in Detroit - because of the automakers need to provision new cars with navigation systems such as OnStar that need an identifier.

Why Not IPv6 Addresses?

Naturally, doing the work I do promoting IPv6 deployment, my first reaction was of course:

"Can't we just give all those devices IPv6 addresses and be done with it?"

The answer turns out to be a bit more complex. Yes, we can give all those devices IPv6 addresses (and almost certainly will as we are simply running out of IPv4 addresses), but:

1. Vendors Don't Want To Be Locked In To Infrastructure - Say you are a utility and you deploy 1,000 smart meters in homes in a city that all connect back to a central server to provide their information. They can connect over the Internet using mobile 3G/4G networks and in this case they could use an IPv6 address or any other identifier. They don't need to use a telephone number when they squirt their data back to the server. However, the use of IP addresses as identifiers then ties the devices to a specific Internet Service Provider. Should the utility wish to change to a different provider of mobile Internet connectivity, they would now have to reconfigure all their systems with the new IPv6 addresses of the devices. Yes, they could obtain their own block of "Provider Independent (PI)" IPv6 addresses, but now they add the issue of having to have their ISP route their PI address block across that provider's network.

2. Some Areas Don't Have Internet Connectivity - In some places where smart meters are being deployed, or where cars travel, there simply isn't any 3G/4G Internet connectivity and so the devices have to connect back to their servers using traditional "2G" telephone connections. They need a phone number because they literally have to "phone home".

While we might argue that #2 is a transitory condition while Internet access continues to expand, the first issue of separating the device/application identifier from the underlying infrastructure is admittedly a solid concern.

Telephone Numbers Work Well

The challenge for any new identifier is that telephone numbers work rather well. They are:

• easily understood - people in general are very comfortable with and used to phone numbers (assuming they have access to phone networks)
• ubiquitous - phone numbers are everywhere and are available globally
• well defined - they have a fixed format that is well known and standardized
• easy to provision - they can be entered and configured very easily, including via keypads, speech recognition and more

For all these reasons, it is understandable that device vendors have chosen phone numbers as identifiers.

The Billing / Provisioning Conundrum

The last bullet above points to a larger issue that will be a challenge for any new identifier. Utilities, telcos and other industries have billing and provisioning systems that in some cases are decades old. They may have been initially written 20 or 30 (or more) years ago and then simply added on to in the subsequent years. These systems work with telephone numbers because that's what they know.

Changing them to use new identifiers may be difficult or in some cases near impossible.

So Why Change?

So if telephone numbers work so well and legacy systems are so tied to those numbers, why consider changing?

Several reasons come to mind:

1. Security - There really is none with telephone numbers. As Henning noted in his presentation and I've written about on the VOIPSA blog in the past, "Caller ID" is easily spoofable. In fact, there are many services you can find through a simple search that will let you easily do this for a small fee. If you operate your own IP-PBX you can easily configure your "Caller ID" to be whatever you want and some VoIP service providers may let you send that Caller ID on through to the recipient.

2. OTT mobile apps moving to desktop (and vice versa) - Many of the "over the top (OTT)" apps that have sprung up in the iOS and Android devices for voice, video or chat communication started out using the mobile devices phone number as an identifier. It's a simple and easy solution as the device has the number already. We're seeing some of those apps, though, such as Viber, now move from the mobile space to the desktop. Does the phone number really make sense there? Similarly, Skype made the jump from desktop to mobile several years ago and used its own "Skype ID" identifier - no need for a phone number there.

3. WebRTC - As I've written before, I see WebRTC as a fundamental disruption to telecommunications on so many different levels. It is incredibly powerful to have browser-based communication via voice, video or chat... in any web browser... on any platform including ultimately mobile devices. But for WebRTC to work, you do need to have some way to identify the person you are calling. "Identity" is a key component here - and right now many of the WebRTC systems being developed are all individual silos of communication (which in many cases may in fact be fine for their specific use case). WebRTC doesn't need phone numbers - but some kind of widely-accepted application-layer identifier could be helpful.

4. Global applications - Similarly, this rise of WebRTC and OTT apps has no connection to geography. I can use any of these apps in any country where I can get Internet connectivity (and yes, am not being blocked by the local government). I can also physically move from country to country either temporarily or permanently. Yet if I do so I can't necessarily take my phone number with me. If I move to the US from the UK, I'll probably want to get a new mobile device - or at least a new SIM card - and will wind up with a new phone number. Now I have to go back into the apps to change the identifier used by the app to be that of my new phone number.

5. Internet of Things / M2M - As noted in the intro to this post, we're connecting more and more devices to the Internet. We've got "connected homes" where every light switch and electrical circuit is getting a sensor and all appliances are wired into centralized systems. Devices are communicating with other devices and applications. We talk about this as the "Internet of Things (IoT)" or "machine-to-machine (M2M)" communication. And yes, these devices all need IP addresses - and realistically will need to have IPv6 addresses. In some cases that may be all that is needed for provisioning and operation. In other cases a higher-level identifier may be needed.

6. Challenges in obtaining phone numbers - We can't, yet, just go obtain telephone numbers from a service like we can for domain names. Obtaining phone numbers is a more involved process that, for instance, may be beyond many WebRTC startups (although they can use services that will get them phone numbers). One of the points Henning made in this SIPNOC presentation was the FCC is actually asking for feedback on this topic. Should they open up phone numbers within the US to be more easily obtainable? But even if this were done within the US, how would it work globally?

7. Changes in user behavior - Add to all of this the fact that most of us have stopped remembering phone numbers and instead simply pull them up from contact / address books. We don't need a phone number any more... we just want to call someone, the underlying identifier is no longer critical.

All of these are reasons why a change to a new application-layer identifier would be helpful.

So What Do We Do?

What about SIP addresses that look like email addresses? What about other OpenID or other URL-based schemes? What about service-specific identifiers? What about using domain names and DNS?

Henning had a chart in his slides that compared these different options ("URL owned" is where you own the domain):

The truth is there is no easy solution.

Telephone numbers are ubiquitous, understood and easy-to-use.

A replacement identifier needs to be all of that... plus secure and portable and able to adapt to new innovations and uses.

Oh... and it has to actually be deployable within our lifetime.

Will there be only one identifier as we have with telephone numbers?

Probably not... but in the absence of one common identifier we'll see what we are already seeing - many different islands of identity for initiating real-time communications calls:

• Skype has its own proprietary identity system for calls
• Apple has its own proprietary identity system for FaceTime calls
• Google has its own proprietary identity system for Hangouts
• Facebook has its own proprietary identity system used by some RTC apps
• Every WebRTC startup seems to be using its own proprietary identity system.
• A smaller community of people who care about open identifiers are actually using SIP addresses and/or Jabber IDs (for XMPP/Jingle).

And in the meantime, Amazon is still assigning phone numbers to each of its Kindles, the utilities are assigning phone numbers to smart meters and automakers are embedding phone numbers in cars.

How can we move beyond telephone numbers as identifiers? Or are we already doing so but into proprietary walled gardens? Or are we stuck with telephone numbers until they just gradually fade away?

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RELATED NOTES: Some additional pointers are worth mentioning:

• The Internet Society (my employer) has a team focused on the broader subject of online privacy and identity (beyond simply the telephone numbers I mention here) and the links and documents there may be of interest.
  
  
• There's a new Internet Draft out, draft-peterson-secure-origin-ps, that does an excellent job on the problem statement around "secure origin identification" as it relates to VoIP based on the SIP protocol and why there are security issues with what we think of as "Caller ID".
  
  
• Chris Kranky recently argued that telcos are missing the opportunity of leveraging telephone numbers as identifiers in the data world.

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My post this week about Google Voice ringing into Google+ Hangouts generated a good bit of commentary, not only on the original post but also out on Hacker News, Reddit, Google+ and other areas. Given the range of responses, I thought I'd reply to a couple of points and also expand on some further related topics. So here goes...

"DUH! This is nothing new/disruptive. You could do it forever with GTalk/Gmail!"

A common response was to point out to me that Google Voice had been integrated with GoogleTalk / GMail for quite some time and so this integration was really nothing new.

Okay, fair enough. Point taken.

I'll admit that I never keep GMail open in a web window and so while I do recall that this integration was there in the past, I never personally used it.

Similarly, in Google+, I've taken to logging out of the GoogleTalk/chat sidebar because I found it was sucking up CPU cycles on my Mac. For whatever reason, the new Hangouts sidebar doesn't seem to consume as much CPU cycles and so I've left it running there.

So yes, the integration may have been there in the past and now it is there in Hangouts - and people like me are actually now noticing it. :-)

Ringing G+ Hangouts BEFORE Ringing Other Devices

There were a couple of comments that it seemed like calls to a Google Voice number rang the Google+ Hangouts first and then rang the other devices connected to the GV number. In my own testing there does seem to be about a 3-second delay between when the call starts ringing in Google+ Hangouts and when it starts ringing on my cell phone and Skype. Now, this may be a fact of Google giving priority to their own application - or it may just be an architectural fact that when they fork the call out to the different numbers it is faster to connect to their own service while the calls to my cell and my Skype numbers have to go through various PSTN gateways. Either way, there does seem to be a degree of delay before all devices ring.

Delay In Answering

A couple of people noted that there was a delay from the time you hit "Answer" to when the call was actually established. I've noticed this, too, although not consistently. I think part of it may be with starting up the Hangouts component inside of your browser - particularly with getting the video going, since that seems to be required for the Hangouts component. It may also be just the paths through whatever systems Google is using. It's certainly something to monitor.

Google Voice Call Does Not Ring The Hangouts App on iOS

In my own testing, I found a curious omission. When I call in on my Google Voice number, it does not ring on my Hangouts app running on my iPad. It rings Hangouts on my web browser... but nothing happens in the mobile app. Now, my iPhone rings - but that is because it is also connected to the Google Voice account. I didn't try removing that number from Google Voice and then seeing if the Hangouts app on the iPhone would ring. At least for the iPad, nothing happens. It would be great if this did work so that I could receive the calls on that mobile device.

XMPP...

Multiple people pointed out that my final remark about maybe some day getting SIP support was probably unrealistic given Google "dropping" XMPP support. I was admittedly away on vacation and at a conference last week and so I missed this point in all the announcement about Hangouts coming out of Google I/O. I wrote about this yesterday, though: Did Google REALLY Kill Off All XMPP/Jabber Support In Google+ Hangouts? It Still Seems To Partially Work

Although, as pointed out in a comment on Google+, this "partial" XMPP support may just be a factor of the continued GoogleTalk support - and may fade away when Google finally pulls the plug on that.

This is definitely an area where it would be helpful if Google could provide a few clarifications.

That's all I have right now for a quite update and response to points. Thanks for all the great comments and I do look forward to seeing where Google is going with all of this.

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Did Google really kill off all of their support for XMPP (Jabber) in Google+ Hangouts? Or is it still there in a reduced form? Will they be bringing back more support? What is really going on here?

In my excitement yesterday about Google Voice now being integrated with Google+ Hangouts, I missed a huge negative side of the new Hangouts change that is being widely reported: the removal of support for the XMPP (Jabber) protocol and interoperability with third-party clients.

But yet a few moments ago I did have a chat from an external XMPP client (Apple's "Messages" app) with Randy Resnick who received the message in a Google+ Hangout. I opened up a Google+ window in my browser and I could see the exchange happening there as well. Here's a side-by-side shot of the exchange in both clients:

So what is going on here?

Reports Of Google Removing XMPP

This issue has been widely reported in many of the tech blogs and sites. Matt Landis covered this issue very well in his post: Hangouts Won’t Hangout With Other Messaging Vendors: Google’s New Unified Messaging Drops Open XMPP/Jabber Interop which then generated long threads on Reddit and Slashdot.

The Verge in their lengthy story about Google+ Hangoutscontains this statement from Google's Nikhyl Singhal:

Talk, for example, was built to help enterprise users communicate better, Singhal says. "The notion of creating something that’s social and that’s always available wasn’t the same charter as we set out with when we created Talk." With Hangouts, Singhal says Google had to make the difficult decision to drop the very "open" XMPP standard that it helped pioneer.

The "Google Talk for Developers" pagealso very clearly states this:

Note: We announced a new communications product, Hangouts, in May 2013. Hangouts will replace Google Talk and does not support XMPP.

A Google+ post by Nikhyl Singhal has generated a large amount of comments (not solely about XMPP) and a post from Google's Ben Eidelson about how Google Messenger will be changed by Hangouts has also received many comments.

There was also a Hacker News thread about the news out of Google AppEngine that apps hosted there would not be able to communicate users of the new Hangouts app via XMPP - and providing a couple of workarounds.

A couple of Google+ threads from Matt Mastracci and Jan Wildeboer are also worth reading as is this note from Daniel Pentecost about how he has lost interop with his clients / customers.

But Is XMPP Support Still There?

I was a bit puzzled, though, by a couple of comments from Google's Ben Eidelson down in one of the G+ threads:

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Ben Eidelson
+Thomas Heinen Thanks for your report of the issue. Hangouts supports basic interop with XMPP, so you can-for the time being-continue to use 3rd party clients. It does not work the same way as Talk, and so I believe the issue you're having with the XMPP bridge will not resolve in Hangouts.

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Jason Summerfield
+Ben Eidelson So there is still some basic XMPP functionality under the hood? Does this mean that Hangouts will still be able to communicate with federated Jabber servers/clients, at least for now?

Ben Eidelson
+Jason Summerfield Not federated support, but supports interop with XMPP clients. Meaning you can continue to use XMPP clients to log in to Google Talk and those messages will interop with folks on Hangouts.

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It was this that prompted me to call up Messages on my Mac, where I am logged in via XMPP to my GMail account, and to initiate a chat with Randy as shown above. We found we could chat perfectly fine. We couldn't initiate a callinto a Google+ Hangout from an external XMPP client - although I'll be honest and say I don't know how well that worked in the past. My own usage of external clients has entirely been for chat.

So What Is The Story?

I don't know. The statement quoted in The Verge's story seems pretty definitive that XMPP has been dropped, as does the message sent to AppEngine developers. It does seem so far that:

• "Server-to-server" XMPP, used for federation with other servers / services, has been dropped.
• "Presence" and status messages have been dropped (because the idea seems to be with Hangouts that you just send a message and people will get it either right then or whenever they are next online).
• Within the Hangouts app, you can only connect to people with Google+ accounts, i.e. contacts on external XMPP servers no longer appear.
• Google hasn't made any clear statements on what exactly is going on.

But is this partial XMPP support only temporary? Will it go away at some point whenever Hangouts fully "replaces" GoogleTalk? Or is this a communication mixup? (As happened recently with Google's announcement of DNSSEC support for their Public DNS Service?)

For me the disappointment in all of this is mostly that Google has been one of the largest advocates for the open XMPP protocol and I enjoyed the fact that I could use multiple different chat clients to interact with my GoogleTalk account. I was also very intrigued by the federation that we were starting to see between GTalk and other systems out there via XMPP.

Whereas before Google+ seemed to be an interesting social/messaging backbone to which I could connect many different apps and systems, now Google+ is looking like simply yet another proprietary walled garden - and we don't need more of those!

Hopefully we'll hear something more out of Google soon.

P.S. Here's another interesting viewpoint: Google Hangouts and XMPP – is cloud harming the Internet?

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UPDATE: In a comment over on Google+, Daniel Pentecost states that Randy and I were not actually using Hangouts:

Dan, you weren't actually chatting through Hangouts. You were chatting through Google Talk which itself has a bridge into Hangouts. It only works b/c Randy is a Gmail user and still has access to Google Talk in Gmail.

Perhaps that is the case, which again then begs the question of whether this is only a temporary capability until GoogleTalk is shut down.

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