Disruptive Telephony

About 1:30pm today, I lost internet connectivity.  It was quite comical, really, how I noticed.  For some reason, I did something I almost never do and hit the "Music On Hold" button on my teleworker sets hanging off of Mitel's switch up in Ottawa.  So there, in a wonderful use of bandwidth, radio station CHEZ-106 out of Ottawa was streaming into my home office down here in Vermont.  (That's what the trial guys are using as the MOH music source for the trial switch to which I am connected.) 

All of a sudden, the phone started playing the same audio packet again and again and again... I felt like I  was transported back about 25 years to the era of skipping records!  I wondered what was up but then I noticed a browser window on my computer not being able to find a link I had just opened to a popular web site.  I quickly looked at my other teleworker phones and they, too, were going into a resiliency mode attempting to failover to their secondary IP-PBX.  A glimpse at my laptop showed that Skype, MSN, Jabber were all starting their contortions of trying to reconnect.

Uh-oh.

Being a network geek, I did the usual response of ssh'ing into my home network gateway, looking at the routing table and trying to ping the next hop router.  Nothing.  Then I did the usual "consumer-hooked-to-a-cable-network" response of power-cycling the cable modem and then re-trying the ping.  Nothing.

A quick call to a neighbor who is also a home office worker got me this: "Oh, yeah, they've cut the Internet access as part of the re-wiring they are doing to the neighborhood power lines. Didn't you get the letter?  There are all sorts of Burlington Electric, Comcast and Verizon trucks up there right now. "

Oops.

We did get "the letter", but given that this is school vacation week and we'd been planning to go away, I'd actually forgotten all about the potential problems.  Since we decided to stay around, the issue hadn't re-entered my consciousness...  perhaps we'll rethink those plans!

I kind of feel like I'm back teaching the "Intro to Networking" classes I used to teach where I'd always say, "Probably 90% of networking issues are Physical Layer - always check your cables first."   If the power company cuts your Net access... there's not really much you can do!

(2:20-ish - Net back up... (or else I'd be hard-pressed to post this note))

Here's something for a Friday afternoon... yes, indeed, out at Emerging Telephony 2007 back in February, someone (David Troy) did indeed have Asterisk running on a Roomba.  And yes, it was "Press one to start sucking. Press two to stop sucking."

Pictures are now available on Flickr.

More precisely, Asterisk is running on a hacked Linksys WRT54G access point (which is Linux-based) and the controller is using a Nokia WiFi/GSM phone to connect to the Asterisk install.  You could also control the direction of the Roomba using the other keys on the phone keypad.

Why would anyone do this?  Well... why not?

As I mentioned in a recent post, the beautiful thing about VoIP is that it now enables people to "play" with telephony... and do wacky things like hook it up to a Roomba!  :-)

Enjoy the weekend!  Perhaps next week I'll actually get some time to upload the rest of the pictures I took out at ETel. (Hey, it was only 6 weeks ago... )

In the usual (and ongoing) flurry of IETF announcements, there was one notice that caught my attention.  It announces that an Internet Draft document about "dialstrings" has been approved to become a standards-track RFC.  So what, you say?  Well here's a bit more info:

This document provides a way of incorporating a dial string into the SIP or SIPS URI scheme. A dial string is a cousin of a telephone number, but rather than taking the form of a fully-qualified E.164 or national-specific telephone number, it is a description of a literal set of dialed digits that would be delivered over a POTS line. As such, it may include pauses, omit prefixes like area codes, and its applicability is necessarily restricted to a particular context (an enterprise, a LATA, etc). Support for dialstrings was formerly a feature of the tel: URI scheme specification (back in RFC2806); since that functionality did not make it into the revision (RFC3966), it is provided here specifically for the SIP and SIPS case.

Think of it as extra digits you have to type when making a call... or extra keys you have to press to start a service.  The challenge is that SIP proxies and other services need to know that it is a string of numbers that should be handled in a special manner, rather than just thought of as part of a SIP address or something like that.  I mention it here only because it's one of those really low-level things that you can do on the PSTN but until now haven't had a (standard) way to do in SIP. It's also one that ultimately anyone implementing SIP will need to implement.  No RFC number yet, but that will come soon.  Note the nice security warning:

Dialstrings exposed to the Internet may reveal information about internal network details or service invocations that could allow attackers to use the PSTN or the Internet to attack such internal systems. Dialstrings normally SHOULD NOT be sent beyond the domain of the UAC. If they are sent across the Internet, they SHOULD be protected against eavesdropping with TLS per the procedures in [RFC3261].

Yep... as we've been saying over at VOIPSA and Blue Box, you definitely need to think about encrypting SIP if you are sending it across the Internet.  If not, bad things will happen eventually.

As I travel around giving presentations about the technologies that are disrupting telephony, one of the themes I discuss is that one of the most severe disruptions brought about by VoIP is that people now have the ability to "play" with telephony in ways that were never possible before.  Pre-VoIP, you needed special (and typically costly) equipment.  Yes, there have been any range of CTI cards that let you play to a degree, but buying the real equipment was just not possible for most folks who might want to "hack" in the original meaning of the word.  Enter VoIP.  Now all you need is an old PC and some open source software and... ta da... you're playing with telephony. 

What I also see out there is that this ability to hack on telephony is happening at the same time that hacking on networks or operating systems seems to be getting less exciting and interesting.  Oh, don't get me wrong, there's still amazing things happening out there... but for people who want to "play" with technology, those areas aren't as exciting or novel as they once were.  So many of those early adopters have moved on to hack on other things... primarily, it seems to me, on "Web 2.0" apps/services/mashups... or telephony.  (And you'll note the already happening collision of both.) 

Because I like giving presentations with very minimalist slides (unless forced to bulletize), I often summarize this latter point as:

The cool kids now hack telephony.

Whether you agree or disagree with my point, I don't think anyone can deny the continued growth in the number and capability of open source telephony projects.  By way of voiploser's blog (also worth checking out), I learned of VoIP Now's list of 74 Open Source VoIP Apps and Resources.  It's a great list, which really serves to illustrate the amount of open source activity happening with regard to telephony.   Some of the projects on there have been around for quite a long time, while there were certainly some there that were quite new (and I'd not heard of them).

My only quibbles with the article would be these:

• There appears to be no way to leave comments, which is too bad, because you would undoubtedly get all sorts of other developers coming out of the woodwork and leaving comments saying "Hey, what about my project?"
• No matter how you structure a list, people will always say it's wrong.  So naturally, I question why you would start with "H.323 Clients" given that all the major work these days is on SIP.
• Given my past interaction with FreeSWITCH (read the comment left by the lead developer), I somehow doubt that they would want to only be classified under "H.323 Clients".  In fact, the inclusion there really makes absolutely no sense to me given that FreeSWITCH is decidedly not a client, but rather more a telephony platform.  It should probably have gone down near the PBXs or in a separate "platforms" category.  Ditto with YATE.  Part of me wonders if the author just wanted to list FreeSWITCH as #1...
• Under "SIP Test Tools", they list some of the more prominent ones, but the VOIPSA "VoIP Security Tools" list has far more, most all of which are also open source and are used to "test" your VoIP system.

Quibbles aside, the list is definitely a good one, and kudos to VoIP Now for putting it together.

If you are viewing this site in Microsoft Internet Explorer 6 (IE6)
there are problems with my new design that cause it not to render
correctly.  Until I can get time to fix it, your only choices are unfortunately to
either: 1) upgrade to IE7; 2) use Firefox; or 3) read the RSS feed.

My apologies... when I was testing the site design and navigation bar on top,
I tested the site with IE7, Firefox 1.5 and 2, and MacOSX Safari., but neglected
to test with IE6.

P.S. And if there are any CSS experts out there who know why my design
is messing up IE6, any such advice would be greatly appreciated.

(Continuing my effort to flush my "queue of things I want to blog about"...)

Last week, per Rich Lafferty, I learned that there are now "unofficial" versions of native Blackberry clients available for AIM, MSN/WLM and ICQ.  I used WebMessenger a bit in the past and found it useful, but stopped using it for some reason I can't exactly recall...  in any event, I'll be curious to try out native Blackberry versions.   Of course, I can't right now.  I naturally tried to download the MSN/WLM client and was told that "this messaging service is not supported by your service provider" (Verizon).  Ah, well, I'll just have to wait a while.  I don't really need IM on my blackberry, but every once in a great while there's a time when I'm travelling and IM would be great to have.

(Continuing my effort to flush my "queue of things I want to blog about"...)

Some time back, I stumbled upon this post "Top Ten: Most underrated VoIP Contributors".  Yeah, okay, so I liked the list because it had my employer on it, and mentioned the work Mitel has done with VOIPSA, which is really the activities that I do. So, yes, it's nice to be recognized and nice to hear the kind words about Mitel's contributions.  Some of the other people on the list were also interesting as well, so it was good to be introduced to others whom I haven't read.

If you are a Blackberry user (I am), you probably discovered sometime early this morning that you were not receiving email messages... and then you no doubt learned that pretty much all Blackberries in the entire Western hemisphere were offline since last night. For email, that is... they still worked as a phone, but I mean, you don't really get a Blackberry for the phone aspect. At this point, basically every major news outlet is covering the story, and I'm sure we can expect the stories to continue for quite some time.   The service seems to be back up now (mine is, anyway), but I'm sure it will take a bit for it to be restored everywhere.

Working in my home office today, I actually didn't notice the outage until I did one of my very occasional scans of Twitter and saw Chris Brogan complaining being stuck on a train without email access.   Knowing Chris, I figured I'd give him a quick call and was rewarded with this great quote about his trip around New York:

"Yeah, I was just in Penn Station and there were all these dudes in suits looking down at their hands and getting increasingly frantic!"  (Chris Brogan)

Indeed!  Given how much the financial industry (as well as the US government!) relies on Blackberries, I'm sure there were a heck of a lot of frantic calls being made all morning.

As a network technology geek, I'll be curious to see what information comes out later about the cause.  ComputerWorld is running speculation that it may have had to do with issues with one of RIM's Network Operations Centers (NOCs), but that is, at the moment, purely speculation and may be a red herring.  (Although it does raise another issue of why RIM has two NOCs that are both located in Canada.  With a global service such that they have, I would have thought that they would have gone for greater geographic distribution!)

In any event, something like this will definitely serve to remind people of how addicted they are to "push e-mail" and will undoubtedly cause larger customers to ask RIM serious questions about network availability (and perhaps to consider other alternatives).  Having some friends working at RIM (with whom I have not touched base), I can only hope they get it all sorted out rather soon.

Those who know me well are aware that one of my hot buttons is my belief that one of the greatest disruptive potentials of VoIP is to fundamentally change the sound of telephony. With VoIP, we are no longer constrained to the 3.5kHz frequency range of the PSTN... I'll save my wideband rant for another day, but tonight I'll just point you over to Rich Tehrani's post "DiamondWare in HD", which talks about the power of DiamondWare's "3-D" stereo technology.  As Rich describes in his blog entry:

Once on the call I was able to easily position the three callers all around me. One could be directly ahead of me and one on either side. The computer can automatically position participants as well if you so choose.

When everyone was in place, Keith had one coworker start counting from 1 and another reciting the alphabet from letter “A.” While these two participants spoke, Keith proceeded to speak with me and the strangest thing happened. I could focus on anyone I wanted and was able to absorb what all three participants were uttering.

It was an amazing experience and the sound quality was beyond compare. I could hear everything in a conversation and more. A sniffle. Lips moving, etc. I swear I could almost hear eyes blinking.

I've not yet personally tried DiamondWare's stuff (although I've met CEO Keith Weiner several times at various shows and he's offered to take me through it) but I am aware that it's also being used in the SecondLife voice beta trial (also another topic for another day).  To me I just think this kind of thing (and publicity by folks like Rich) is great to see... with VoIP (or IP Telephony or whatever we want to call it today) we have the very real capability to have far better audio communication than we've ever had with "traditional" telephony.  It's here now already (just ask a Skype user, but that's yet another topic)... but companies like DiamondWare are showing us just how much farther the boundaries can be pushed.  Cool stuff!

In my incredibly long queue of things I've wanted to write about for the past few weeks, one item was the Mitel news release about making a direct SIP connection to Microsoft Exchange Server 2007 Unified Messaging. The cool part is that you can just use our basic 3300 ICP communications platform (or IP-PBX, or whatever you want to call it) and connect it directly into a Microsoft Exchange Server to use the Exchange Server for a unified inbox (email, voicemail, fax, etc.).  No other boxes or gateways necessary.  Just a nice, standard SIP trunk.  As a long-time proponent of open standards and general "standards geek", it really can't get much better.  It's great to see.