Disruptive Telephony

Now that we see some incredibly powerful peer-to-peer (P2P) technology models emerging in the telephony/communication space, will we see that innovation being challenged or delayed by patent lawsuits?

The New Hampshire Business Review reported this week that Skype has settled a patent lawsuit with Mangosoft for $2.3 million over a patent apparently related to "dynamic directory service". Now per the NHBR article, it would appear that Mangosoft is fading away as a company and indeed while the website appears on initial view to be there, the management team is simply the one CEO and the newest "news" on the web site dates from early 2007. Their news release about the settlement with eBay is very brief and refers now to "MangoSoft Intellectual Property, Inc." Phil Wolff over at Skype Journal notes that MangoSoft's SEC filing is also brief (but discloses the amount). Looking back at MangoSoft's 2007 annual report, they are themselves very clear on what they are doing:

BUSINESS STRATEGY

We no longer develop new software products or services. We continue to market, sell and support our software services. Our strategy also includes seeking strategic business partnerships and distribution channels to leverage our patented technology. All of our business operations are overseen by our sole officer and director, who utilizes third party contractors, as required, to implement the Company’s business strategy.

Though I had not heard of Mangosoft until this article (even though I was living in southern NH during their height), I will say that their technology sounds interesting and indeed in reading Mangosoft's patent 6,647,393 on "Dynamic Directory Service" (either at the US Patents and Trademark Office or over on Google Patents) their invention filed back in 1997 does appear to be essentially what we would call today a peer-to-peer distributed directory service, where "directory" is used in the truly generic form as referencing a list of objects of any form (ex. file descriptors, user info, any pieces of information). [Obvious HUGE caveat - I am NOT a patent lawyer, nor do I play one on TV or the Internet or anywhere else.] From what I know of Skype's architecture, it would seem that they do use a distributed directory service and so it is perhaps no surprise that they eventually settled.

The question is really - is this just the beginning of more lawsuits in the P2P space? MangoSoft's annual report for 2007 shows a debt of $89 million as of December 2006 and the NHBR articles notes that the trend in operating losses has continued with a $680,000 loss in 2008 year-to-date. There is obviously an incentive for them to continue on to try to recoup the ~$90 million that investors have sunk into the company. Beyond this patent, Mangosoft holds several other patents that are related to distributed architectures. It could very well be that this $2.3 million from Skype will be invested now in future lawsuits against other players in the space. Or perhaps not... perhaps it will simply be distributed to some of the existing investors as the operation fades away. I guess that will largely depend upon how much of a solid case to proceed MangoSoft's investors and sole employee believe they have.

While I am definitely sympathetic to inventors who pursued a new technology but were perhaps too far ahead of their time, I must say that I'm not personally excited to see more lawsuits hitting the industry as we see more and more companies (startups, typically) exploring new ways to build communications technologies based on P2P networks. We're in a fascinating time from a network technology point-of-view, as massively distributed networks are now possible and through systems like Skype and BitTorrent we've seen that they are very possible to create. I'd like to hope that this innovation will continue unimpeded by legal battles... although I realize that that's probably an idealistic dream. Even if MangoSoft does not pursue others, over time other larger players will challenge the startups in court should they become more of a competitive threat.

Ah, well, we shall have to see. In the meantime, I guess the good news for Skype is that with their one-time licensing of MangoSoft's patents, they will at least be protected from any further issues in court on these particular patents.

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Technorati Tags: skype, voip, mangosoft, p2p, patents, peer-to-peer, telephony, communications

I was rather humbled to find myself included among Sheryl Breuker's list of "10 people you should follow on Twitter" related to VoIP/telecom/communications. I appreciate that she and others find value in what I post in my Twitter stream... or at least... they find enough value to outweigh the other random posts I put out in my Twitter stream. :-)

Seriously, Sheryl's list is a good one and if you are interested in the VoIP / telecom / communications space, I'd definitely encourage you to follow the others on Shery's list (it's probably not a surprise that I have been). If I were doing my own list, there's probably a few more I'd add... and maybe I'll have to do that sometime...

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In about 2.5 hours, at 11am US Pacific / 2pm US Eastern, I (Dan York) will be participating in a "Hosted Speech Solutions" webinar sponsored by Speech Tech Magazine. I'll be joining colleagues from Microsoft (TellMe), Angel.com and Convergys. We'll be talking about each of our hosted offerings and then answering a series of questions before then throwing it open to questions from the audience.

If you would like to join in and learn about our solutions (and those of our competitors), you can register for free.

Technorati Tags: voxeo, dan york, danyork, speech, hosted, cloud, cloudcomputing, ivr, voice, voip

If you are in the San Francisco area (perhaps for VoiceCon?) and are interested in "emerging communications" or "emerging technology", would you like to join a group of similar folks at a dinner Wednesday night (Nov 12, 2008)?

Lee Dryburgh, the organizer of the eComm Emerging Communications conference, is hosting a private dinner in conjunction with Thomas Howe at the San Francisco Airport Marriott (Burlingame). There are currently some 50+ folks attending and some seats left and if you are tracking or pushing things forwards in the communications space you may like to try and reserve a seat (75.00 USD) by emailing Lee.

I'll be there, naturally, along with Thomas Howe, Eric Burger, Ken Camp, Sheryl Breuker and many others who are involved in the space. If you do want to join us, please email Lee very soon.

Technorati Tags: ecomm, ecomm2009, voip, san francisco

Over on Blue Box, I've now uploaded two recent episodes:

• Blue Box #83: SIP and Asterisk vulnerabilities, voice biometrics, P2PSIP, Aircell blocking Skype, VoIP security news and more…
  
  
• Blue Box #84: New Cisco, Avaya, Nortel VoIP security vulnerabilities from VoIPShield, Skype in China, UCSniff and other new tools, news and more

With that I am almost caught up with our main shows... and I still have a bunch of Special Editions to finish producing and post. I'm hoping to finish post-production on #85 tonight so that I can post it tomorrow. We'll see...

Technorati Tags: bluebox, blue box, voip, voip security, security, voice, skype, sip, dan york, jonathan zar, voipsa

Is the role of "voice" diminished or enhanced by the availability of web services? How does voice fit into the "cloud"? Where do service providers fit into the picture?

Out at ITEXPO last month in Los Angeles, industry analyst Jon Arnold asked me (Dan York) to participate in a series of video interviews he was recording for his IPConvergence.TV site. In the interview, which is now available for viewing, we talked about "voice-enabling" business processes, web services, "cloud computing", the challenges to service providers and customers and much, much more. Jon also asked me to talk a bit about what I see ahead of us in the next few years. It was a fun interview to do and I appreciated the opportunity.

NOTE: There is no way to currently embed the video, so you'll need to watch it over on TMC's site. I'll also note that on my Mac I couldn't watch the video in Firefox but instead had to use Safari. Now this may be due to some local configuration issue on my system, but I thought I'd mention it.

Technorati Tags: voxeo, dan york, jon arnold, itexpo, commdev

While I write here mostly about technology, my interest has always been in the social impacts of all this communication technology. Here on the Today show was a fascinating story of someone who is blind using VoIP (and specifically Skype) to allow him to "see" courtesy of a remote person seeing his view and telling him what is there.

Fascinating...

P.S. Thanks to Chaim Haas for passing this along...

Technorati Tags: skype, voip, video

If you read this blog, the Voxeo blogs or follow my Twitter stream, you know that I very frequently travel and speak at conferences related to VoIP, telecommunications, etc. Often to the tune of 1-2 weeks per month. I enjoy the presenting immensely in that I get to teach people and tell the story about the disruption going on within telephony and communications in general... and also get to learn from some of the amazing people out there about what they see happening all around us.

Yet while I greatly enjoy and find so much value in the large VoiceCon and ITEXPO shows and so many of the others that I attend, one show stands out each year among them all:

eComm

.... eComm 2008 as well as the O'Reilly ETel shows that proceeded it in 2006 and 2007.

Why? Mostly because eComm and the earlier ETel shows were NOT the big conferences. There is no massive trade show. There aren't a huge number of different tracks. It's small... it's focused... and it has some amazing speakers who are truly out on the bleeding edge of how the ways in which we communicate are changing. It's a place to learn from those who are out there leading the disruption... it's a place to forge partnerships... it's a place where many startups have found interested investors... it's a place where you can sit and randomly meet those who are creating the "Post-Telecom Era".

Now if you would like to be one of those amazing speakers at the upcoming eComm 2009 to be held on March 3-5, 2009, at the Burlingame Marriott right near the SFO airport, the Call For Speakers is now open. The deadline for proposals is November 17th, but I expect the speaking slots to fill up before then. I'm on the eComm Advisory Board and we're already starting to see proposals come in by way.

If you've got an idea for a talk you'd like to share, please do submit a speaking proposal - please do note that eComm is very definitely NOT a place for sales presentations and such talks get quickly rejected. As the site says:

This isn't a traditional telecom conference. The eComm audience has very high expectations of speakers. They are both seizing opportunities of the post- telecom era (or re-inventing traditional products and services) and can engage the audience. Rules include a ban on "brochure speak" from stage (overt marketing pitches) and a strict enforcement of the clock.

Plenary presentations lasts just 15 minutes including 2-3 minutes of Q+A. We've found this format of carefully prepared presentations keeps the atmosphere charged. This format also helps us to fit in more than 70 speakers over 3 days. Thanks to the intimate feel of the venue and the energy and attention of the audience, many speakers find it to be a great public speaking experience.

Some of the topics we're looking for include:

• Democratization of communications innovation; anything from VoIP community, XMPP enabled social networking to DIY 12-volt telephony
• Convergence of the media industry with personal communications
• Theme "telecoms is becoming software"
• Telecom restructuring, threats, or new business models
• Telecom trends, particularly Asian
• iPhone or Android applications
• The new old - traditional carriers or vendors who are changing the game
• Mobile Social Software (MoSoSo) applications on any platform; any socio-centric devices or applications will be considered whether mobile or not
• Social Computing
• Mobile leveraging of Cloud Computing or Telco in the Cloud
• Future of Social Networking Applications
• Network Equipment Providers plans for next 1-3 years
• Facilitating business processes with voice
• 4G Technologies
• P2P modes of production or networking

We're looking for all those topics and more... basically any topic related to how what we have known as "telecom" is fundamentally changing all around us... if you have an idea, please do submit a speaking proposal as soon as you can.

And if you don't want to speak - but do want to stay up on where the bleeding edge of telephony is - please do mark March 3-5, 2009, down to join us in San Francisco! (and follow the eComm 2009 blog to know when registration opens).

P.S. If you'd like to help promote eComm 2009, web site badges are available.

Technorati Tags: ecomm, ecomm2009, conferences, voip

Earlier this month out at ITEXPO in Los Angeles, I participated in the Ingate SIP Trunking seminars as I have been doing for the last year or so. My talk was "SIP Trunking and Security in an Enterprise Network". The slides are available for viewing or download from my SlideShare account and I'll also embed them here in this post.

I did record the presentation in both audio and video and hope to be making that available as a Blue Box podcast some time soon. I'll then sync the slides to the audio. Meanwhile... enjoy the slides!

Technorati Tags: voip, voip security, sip, sip security, voice, security, dan york, itexpo, commdev, voipsa

After my post yesterday about "Skype for Asterisk" (and the update post) and the potential it allows for SIP interoperability via Asterisk, I've received a few comments that seemed to interpret what I wrote as somehow indicating that the Skype announcement somehow meant that there was new "Skype to SIP" functionality in the "Skype for Asterisk" announcement.

Just to be clear, there isn't any new "Skype to SIP" functionality in the "Skype for Asterisk" piece announced yesterday by Digium and Skype. None.

It is purely a commercially-licensed software module (which most of us speculate will be a binary software module, i.e. we won't be able to actually see the code) that provides two-way connectivity from Asterisk to and from the Skype cloud. Skype users can call into an Asterisk system. Users connected to an Asterisk system can call out to Skype users. Users on the Asterisk system can also call to the PSTN (via what was called "SkypeOut") and receive calls from the PSTN (via what was called "SkypeIn").

That's it. That was the announcement yesterday. Period. End-of-story.

However, the point I was making in my post yesterday was this announcement has the potential to turn Asterisk into a two-way "Skype-to-SIP" gateway. Asterisk - with the "Skype For Asterisk" module installed - could be deployed into a network where it could provide interconnection between Skype users and SIP users.

Let me explain...

ASTERISK INTERCONNECTION EXPLAINED

Asterisk has a large number of "channel drivers" that allow phones and systems to be connected to the Asterisk system via various protocols. These systems include:

• SIP phones, systems and clouds
• Cisco phones and systems (via the SCCP channel driver)
• H.323 phones and systems
• MGCP phones
• other Asterisk systems (via the IAX channel driver)
• the PSTN and legacy TDM systems (using the various hardware channel drivers)

There is also an unsupported UNISTIM channel drivers to go into Nortel systems and various other channel drivers out there. (I know of someone who is using a radio channel-driver to interconnect two-way radios to Asterisk.)

The beautiful part about Asterisk is that you can simply and easily interconnect all these systems because the individual endpoints simply become extensions in the "extensions.conf" file inside of Asterisk. So extensions that use the SIP channel can call in and connect to an extension that uses H.323. H.323 endpoints can call Cisco IP phones. Cisco IP phones can call an IAX softphone (there are some out there). Any of those different types of endpoints can call the PSTN through either hardware cards or through SIP or IAX trunks. Graphically, the pictures looks something like this:

Although perhaps to better depict common scenarios, here's how it could look if you include various options for PSTN connectivity:

DIVING A BIT DEEPER

To dive a bit deeper into Asterisk configuration, when you decide to use one of the various "channel drivers" you essentially perform two steps:

1. Modify the channel driver configuration file
2. Add appropriate extensions or trunk settings to the 'extensions.conf' file.

Now you might be doing this by editing the configuration files directly using your favorite text editor - or more likely these days you are probably using one of the many different graphical user interfaces to do the actual configuration modification. In the end, the config files are being modified in some manner.

For the SIP channel driver, the config file is the aptly named sip.conf and in the file you enter information such as:

• Connection information to a SIP Service Provider if you are doing a "SIP trunk" for PSTN connectivity (username, password, connection details, etc.)
• Connection information to an IP-PBX or application server to which you are connecting via SIP
• Information about the different SIP phones connected to your Asterisk server

Note that you could use the SIP channel driver to connect individual SIP phones to Asterisk; you could connect your Asterisk server to the PSTN via a SIP trunk; or you could do have both SIP phones and a SIP trunk. In fact, you can have multiple SIP trunks. You could connect over to an existing IP-PBX or to an application server that supports SIP. Asterisk is incredibly flexible in the way that you can configure it.

The second step is to configure the extensions.conf file to have use this channel driver. For instance, to make it so that all calls starting with the digit "9" go out a SIP trunk, you might do something like this:

exten => _9.,1,Dial(SIP/${EXTEN:1}@mysipprovider-out,30,r)

(Taken from the voip-info page on sip.conf)

To configure actual phones as extensions, you would enter something like this in extensions.conf (taken from the sample file on an Asterisk install I have around):

exten => 6245,1,Dial(SIP/Grandstream1,20,rt)   ; permit transfer
exten => 6245,1,Dial(SIP/Grandstream1&SIP/Xlite1,20,rtT)
exten => 6361,1,Dial(IAX2/JaneDoe,,rm)         ; ring without time limit
exten => 6389,1,Dial(MGCP/aaln/[email protected])

If someone dials one of these extensions, it would then ring the associated phone. Note that for extension 6245 it will actually ring two different phones. Note also that the phones will have to be configured themselves to register with Asterisk, etc.

Now, Asterisk dialplan creation is an enormous subject in and of itself... but this is the general idea. You configure the channel drivers to support connections to either (or both) service providers or endpoints (phones) using the given protocol. You then configure actual extensions or trunk connections in the extensions.conf file.

SO HOW *MIGHT* THIS WORK WITH 'SKYPE FOR ASTERISK'?

Now we don't have any information yet about how this new channel driver would work... but most all of the channel drivers work in a similar fashion. I would expect that there will be something like a skype.conf file that will let you establish what Skype user names are associated with the Skype For Asterisk module. You will probably need to include the login credentials, any restrictions on access (by Asterisk users), etc.

Separately, in the extensions.conf file, you will wind up probably putting in something like this to enable outbound connections:

exten => _9.,1,Dial(SKYPE/${EXTEN:1}@t,30,r)

Or something like that. Now all calls that start with 9 will go out the "Skype trunk".

If you wanted to associate a user ID on the Asterisk system with a Skype ID, you would possibly add something like this:

exten => 6400,1,Dial(SKYPE/danyork,,rm)

Now any calls to ext 6400 on that Asterisk system would go to my Skype ID. You could imagine getting a bit fancier with something like this:

exten => 6400,1,Dial(SKYPE/danyork,,rm)
exten => 6400,1,Dial(SIP/1234,,rm)

which would have the effect of calls to ext 6400 going to both my Skype ID and a SIP phone.

So Skype can just be another way for inbound or outbound calls to enter the Asterisk server - and Skype users can simply be added as extensions on an existing Asterisk server.

Returning to my graphic above (gotta love quick graphics via Skitch!), the picture now looks like this:

The "Skype For Asterisk" module allows two way connectivity into the Skype cloud and also the use of Skype as another mechanism for PSTN connectivity.

SO WHAT ABOUT "SKYPE-TO-SIP"?

The point of my post yesterday was now that two-way Skype connectivity becomes just another channel driver for Asterisk, you have all sorts of interconnection possibilities. As a standalone system, you could connect SIP phones on an Asterisk server out to the Skype cloud.

You can also deploy Asterisk as a "SIP-to-Skype" gateway. You could connect an Asterisk box via SIP to an existing IP-PBX and enable connectivity from that IP-PBX to Skype users. You could even be incredibly stupid (given existing security issues) and connect your Asterisk box directly to the Internet and provide a SIP-to-Skype gateway. (If you aren't aware of the SIP security issues, listen to any of my Blue Box podcast episodes or read the VOIPSA blog.)

If the Skype For Asterisk module delivers the functionality it sounds like it will, there are a whole range of possibilities now available for interconnection between the Skype cloud and other VoIP systems simply by putting an Asterisk box in the middle.

We'll see. All of this is mere speculation until we can actually use the Skype For Asterisk module.

Technorati Tags: skype, asterisk, skypeforasterisk, voip, sip, interoperability