Disruptive Telephony

In a few hours I'll be boarding a plane back to New York where I'll be attending Interop New York this afternoon and tomorrow. If any of you reading this will be there, please do drop an email. Tomorrow, I'll be on a panel at 2:45pm with Jonathan Rosenberg about "Voice-oriented Attacks". (Side note to Interop: Please make it so that we can link to individual sessions instead of having to link to the entire list of "security"-related sessions!) If you aren't aware of who Jonathan Rosenberg is, he works for Cisco and is a huge contributor to IETF efforts related to SIP and in fact was one of the co-authors of RFC 3261 which is the primary RFC defining SIP. He's also the author of "The Hitchhiker's Guide to SIP" which aims to help guide people through the maze of the many, many documents that now are part of "SIP". More relevant to tomorrow's session, he's also the author of a series of NAT traversal protocols for SIP, namely STUN, TURN and now ICE. Eric Krapf, the moderator of the session, is aiming to make it a more interactive and discussion-focused session (i.e. no slideware-to-death)... we'll see if we can make it fun as well. I've also asked Interop for permission to record it and run it as a Blue Box podcast - we'll see if they give me permission.

Note that if you are a CISSP, the ISC2 is holding a member reception today (Wednesday October 24, 2007) starting at 5:30 PM in Jacob Javits Center Room 1EO2 - LEVEL 1. Assuming that everything works with my flights today, I'll be there.

I'll even have some new business cards to give out... ;-)

Technorati Tags: cisco, conferences, danyork, interop, jonathanrosenberg, voip, voip security, voipsa, voxeo

The SMB VoIP game is changing. Fundamentally. And in a pattern we've seen before in other industries. In the news release out today, Digium and 3Com announced that:

Under the terms of the agreement, 3Com will offer Digium’s award-winning Asterisk Appliance™ to small businesses that need a reliable, easy-to-deploy voice solution based on open standards. 3Com Asterisk will be available through the company’s proven channel of partners worldwide.

Let's think about that for a minute. 3Com will make Digium's Asterisk appliance available through "the company's proven channel of partner's worldwide", which some reports are putting at around 60,000 resellers. Digium just wound up with a large global sales channel. Yet to be seen is whether there will be any channel conflict with existing Digium Partners/VARs, but regardless, Digium just wound up with a way to deploy Asterisk-based solutions globally. It does, however, get one step better (my emphasis added):

“3Com is focused on delivering products and solutions for converged secure networks, in which voice is an application that can be readily integrated with many others,” said Bob Dechant, senior vice president and general manager for 3Com Corporation. “We’ve announced a complete voice strategy and new product offerings for small businesses, including the 3Com Asterisk Appliance. We also offer innovative enterprise-caliber 3Com Global Services for customers who purchase the 3Com Asterisk. We chose to partner with Digium because of the company’s position as the Asterisk leader, its commitment to open standards and the ease-of-use of the appliance.”

Yes, indeed, Digium winds up with a global support organization behind Asterisk. Powerful announcement. Global sales and support - for an open source PBX... According to information from Digium, the "3Com Asterisk", priced at $1,595, will include a 3Com-co-branded interface and easy configuration/provisioning of 3Com SIP phones (as can be done today with Polycom phones). Given last weeks' announcement of the SwitchVox acquisition, I would think that rolling some of that GUI/functionality into the offering would be another logical step longer-term. The implications of this announcement, though, go far beyond the commercial relationship between Digium and 3Com. Those of us who remember Linux in the late 1990s and early 2000s remember that Linux took a trajectory like this:

1. Techies/geeks/early-adopters started to install Linux into their businesses to solve specific needs. Often it was installed without corporate permission as a DNS server, web, server, etc.
2. A range of small, specialized vendors started to ship servers with Linux pre-installed. Very often these companies were founded by people within the Linux community (ex. VA Linux, Penguin Computing)
3. Larger, more mainstream but still lower-tier manufacturers started to ship servers with Linux. (I forget the first one I saw doing this.)
4. Tier 1 manufacturers (ex. IBM, HP, Dell) started to ship servers with Linux.

Asterisk just moved to step #3 (after already moving through #1 and #2). While 3Com does not have the same market status as Cisco, Avaya and Nortel (or Mitel in SMB), 3Com definitely has a presence out there and to me their endorsement of Asterisk certainly brings a level of credibility to Asterisk-based software and hardware. It's good for Asterisk. It's good for Asterisk-based products and services (including those of Digium's competitors). It's good for open source. Ultimately, in my opinion, it's good for all of us.

Yet to be seen is how good it is for 3Com's own SMB offerings and that will be interesting to see. Right now it seems that they are all about "offering customers choice" between 3Com's own product and the Asterisk-based appliance. Will that last? Will 3Com continue to maintain its own SMB products long-term? Or will it cede that lower-end market to Asterisk and focus on apps that interoperate with Asterisk and/or phones for Asterisk (and 3Com's higher-end offerings)? Interesting questions to consider, particularly in light of 3Com's announcement on Friday that it is being acquired by Bain Capital and Chinese giant Huawei as well as their announcement today of new VCX systems targeted at the SMB market.

Nor is it clear to me how much of a short-term impact there will be on the SMB market. 3Com has been less of a presence in that space in recent years although its clear from their various announcements today that they are intent on playing a larger role in the space. Will Mitel, Avaya, Cisco, etc. lose any sales today as a result of 3Com selling Asterisk? Maybe. Maybe not.

Longer-term, though, I personally view this as a huge validation that open source telephony has a role in the business space. The cracks in the wall of proprietary telephony just got a whole lot larger today. Congrats to Digium and 3Com... and now the question is - who's the next vendor to get on board?

What do you think? Is this a validation of Asterisk? Or a flash in the pan? (Or as one more cynical person put it to me, "a desperate move by 3Com to stay relevant?") What do you see as the short-term and long-term impacts to the SMB market? Should the existing vendors be scared? Or just ignore it?

More coverage:

• Alec Saunders: Digium Scores with 3Com
• Greg Galtizine: 3Com to Offer Customers Asterisk Appliance
• VoIP-News: Digium Targets Mass Market Again With 3Com Deal
• ZD Net (Russell Shaw): 3Com fills SMB nices with new IP Telephony products
• eWeek Channel Insider: 3Com Preps VOIP Switches for SMBs

Technorati Tags: opensource, asterisk, digium, voip, 3Com, telephony

Imagine you are a customer service rep (CSR) at a small/medium company and a phone call comes in from a customer. As your phone rings, up on your screen pops all the information about that customer, pulled from your CRM database in Salesforce.com or SugarCRM, plus other information from other databases and finally a nice Google Map showing you where that customer is located and potentially other information like the locations of your nearest offices. During the call, the CSR needs to bring in a subject matter expert so the CSR consults their web panel and looks at the presence information displayed for each of the other people in the business. The CSR can then contact someone showing as available and potentially bring them into the call.

Now imagine that all that is running on top of open source telephony... specifically Asterisk.

You can now stop imagining, because Digium just bought the company that does precisely that. There will undoubtedly be much attention today (at the very least in the VoIP blogosphere) about Digium's announcement here at AstriCon today that they have acquired SwitchVox. I am going to bet that much of the reporting today will focus on angles like these:

• Digium now has very competitive offerings (SwitchVox SOHO and SwitchVox SMB) for going after the small / medium business market.
• Digium bought themselves a very sophisticated/simple/easy GUI/management interface that moves them forward dramatically in making Asterisk easy to use, deploy and manage.
• Digium just got 1400 paying customers with over 65,000 endpoints.
• Digium bought themselves parity (or more) in their ongoing competitive feud with the folks at Fonality/Trixbox.

All of that is true. The SwitchVox products offer a very seriously competitive list of features (you have to go through and expand the subsections to see all the features). The GUI is very well done and simple. The price is quite compelling for the servers and also the support. I mean, for $1200 ($995 server plus $199 support) an SMB gets an IP-PBX with a very broad range of features and an unlimited number of users! Yes, the business still has to pay for IP phones, but they can buy any of a wide range of phones at varying price points to suit their needs. Considering that almost all the mainstream IP-PBX vendors charge on a per-user basis for licenses, the unlimited user model is certainly disruptive in its own right. (Digium has also been doing this with their Asterisk Business Edition.) And yes, Digium now has an answer to the growing competitive threat of Trixbox and it's management interfaces, support, hybrid model, etc.

All that is true - but it's not the really interesting story.

To me, what is far more compelling is that Digium just bought themselves a whole group of people who "get" the world of "unified communications", business process integration, Web 2.0 mashups, etc.

Digium has had no story at all around "presence" within its core offerings. Now it does. While Asterisk has always been a platform play where you have the ability to integrate Asterisk with other apps, doing so has not exactly been for the faint-of-heart. Hire yourself some programmers and you can do pretty much anything with Asterisk... but that's not something that many businesses want to get into. SwitchVox now gives Digium a way to do easy integration with databases and web sites. The integrations to Salesforce.com and SugarCRM are slick. The Google Maps popup is a seriously cool mashup! (And where is that on the roadmap of the mainstream vendors?)

Throw in a "click to call" add-in for Firefox to let you dial any number you see on any web page, plus a plug-in for Outlook, and you've got a very compelling offering. For a very nice price. My only knock (other than the fact that I can't find a picture of their Google Maps mashup anywhere on their website) is that it doesn't seem like their presence capability is yet integrated with existing instant messaging services. Given Asterisk's XMPP (Jabber) capabilities, this seems an obvious path that could get them connected to Jabber and GoogleTalk presence information. If they don't have that yet, I hope they add it soon, as we really do NOT need yet another place to change/update our presence info.

Regardless, this integration capability is, to me, the real story. Phones are being commoditized. I have to believe call servers/IP-PBXs are on their way to being commoditized. (Folks like Microsoft are going to help in pushing those prices down.) The money will ultimately go away from those areas.

The future of "unified communications" is about platforms. About mashups. About web services. About exposing APIs. About making it easy to combine different sources of data into interfaces that make people more productive. Microsoft gets that. Some of the traditional IP-PBX vendors get that. Digium has always known that, but this acquisition gives them a far better ability to make it happen.

Congrats to the folks at both Digium and SwitchVox for making this happen... I very much look forward to seeing where it evolves! (And in the meantime, I'm going to have to go down to the AstriCon exhibit hall and get some video of the Google Maps mashup to show how very cool it is...)

Read more:

• coverage in the blogosphere
• Digium FAQ about the acquisition
• Digium blog post by Bill Miller (more posts will apparently follow today on their blog)

Technorati Tags: asterisk, astricon, digium, mashups, opensource, presence, switchvox, telephony, voip

In watching Jay Phillips do his great presentation here today at AstriCon about Ruby and his Adhearsion package, I found myself wondering what the interesting little softphone was that he was using. It turned out to be "Zoiper", an IAX or SIP softphone that was previously called "Idefisk". (I can understand perhaps why they changed the name... "Idefisk" does not exactly roll off your tongue.) There turn out to be two versions (comparison chart here): a free version and a "Zoiper Biz" version which includes more functionality and starts around 30 euros.

Clearly built for Asterisk, it was interesting to note that it supports both SIP and also Asterisk's own IAX protocol. Anyway, I just thought I'd share that this softphone is out there if you were not aware of it.

Technorati Tags: asterisk, iax, softphone, video, zoiper

Another new "voice" application for Facebook come out today, this one from the 12-year-old FWD (the service formerly known as "Free World Dialup" and backed by Jeff Pulver, who recently teamed with Daniel Berninger to relaunch FWD - read Daniel's perspective here and also Jeff's post about FWD's beta of a tunneling service )

This first Facebook app, called simply "Voicemail", was announced to members of the FWD group inside of Facebook with a message from Daniel Berninger providing the URL and stating this:

We are particularly interested in novel uses enabled by the several differences with traditional telephone voicemail.
1) CD quality audio
2) Messages public or private
3) Ability to re-record message without sending
4) Sent messages remain accessible
A direct integration with FWD will be available shortly allow you to pickup and leave Facebook voicemail via FWD.

My initial response was admittedly a bit of a yawn.  Back in June, I had written about the existence of several Facebook apps that allowed FB users to leave each other voicemail messages.  The last sentence, though, was enough to intrigue me:

A direct integration with FWD will be available shortly allow you to pickup and leave Facebook voicemail via FWD.

I don't think I've really ever written much about FWD in any of my blogs, but it was one of the earliest VoIP systems (some history here). It uses SIP and interconnects with a range of other IM systems. (See the feature list for more info.)  I have had a FWD number, but haven't really used it that much in a long time.  It will be interesting to see where this relaunch takes it.

Trying It Out

In any event, I was intrigued enough by the tease that SIP-connected endpoints might be able to leave a voicemail inside of Facebook to try the Voicemail application out.  The installation was as painless as any other Facebook app.  Once installed, you get a screen like this:

I logged in and next had an inbox-type of screen (click on image for larger version):

 

I naturally had to click on the "Friends with VoiceMail" link to see what it did and, sure enough, it showed me all my Facebook friends who had the VoiceMail app installed and gave me the chance to leave them a message. Of course I had to try it with Jeff, so I clicked on his name and my system went off and started spinning for a few seconds.  I noticed the Java icon appeared in the Windows systray and soon I wound up with this confirmation box:

Once I clicked on Run, the resulting box gave me a very simple interface to use:

At this point I just thought I should click the big button in the center, not realizing that it had the arrow for "Play" in the middle. Clicking the button gave me a status message that clued me in to that fact and so I clicked the first button which did record and let me see my audio levels as well as the amount of time of the recording:

When I was done, I clicked the third button and stopped the recording.  I could then go and play the recording.  Since it wasn't that great, I decided to re-record.  I clicked the button and was told to confirm:

It's interesting that it is effectively telling me where FWD's server is via the IP address.  I confirmed, re-recorded and then hit the Send button to fire the message off.  There was a brief status message as the voicemail was uploaded, and then I was back to my "Friends with VoiceMail" screen with the typical Facebook-style "success" message at the top of the screen.

Clicking on "My Messages", I returned to my "inbox" and clicked on "Sent Messages", where I saw the message listed:

with the options to delete it or listen to it. 

Conclusion

All in all a pretty straightforward app to use.  I'd note that the button visible on the pages simply takes you to the "Friends with VoiceMail" page where you can then send a message to one of your friends.  There's also an "Invite Friends" page which lets you very simply invite some of your friends to check out the app.  (Feedback for Daniel/Jeff: You are told on that page that you can only invite a max of 10 friends per day but all of your friends are selected and there doesn't seem to be a way to "Select None".  I would therefore conceivably have to go through and de-select all of my friends in order to only select 10. Needs to be a better way to do this.)

The one aspect I was curious to try was this:  "2) Messages public or private"  However, I didn't have any messages waiting for me to try it on and there seemed to be no settings for the Sent message.  So if someone reading this can try out the app and send me a message, that would be great.  Of course, you need to be a friend of mine, eh?

The External Connection

But what about the external connection to FWD clients?  How could we have a call wind up in Facebook voicemail?  Well, inside the Facebook forums, Daniel Berninger left us this tease about the system they are beta testing:

FWD-FB Integration
A) FWD User Leaves FB Voice Message
* FWD user A picks up the handset and dials an FB enabled FWD user (FWD user B)
* User B doesn't answer the call, and the call is diverted to the FB voicemail bridge via SIP or IAX. The call is forwarded using a special number format, indicating the FB voicemail server and the receiver of the voicemail.
* The voicemail application on the FV-VM bridge is activated, and the user records the users.
* Once the user hangs up, the bridge records the voicemail into the database, activating a conversion script to convert the WAV format to an MP3 format - and updates the database accordingly.
B) FWD User Picks UP FB Voice Message - via the phone
* FWD user calls his voicemail service, via a special FB-VM access code.
* FWD identifies itself on the VM system.
* FWD performs normal user interaction with the voicemail system (requires some Asterisk core modifications).
* FWD user hangs up when complete.
C) FWD User Picks UP FB Voice Message - via FB
* FB user listen to messages via the web interface, in an identical fashion to what's available now.

So if I parse through this, it sounds like the FWD team wrote a custom script for Asterisk to do this conversion and is perhaps using Asterisk for the rest of the functionality as well.  Now I'd be curious to wonder if the "FB voicemail bridge" will accept any SIP connections or just those from authenticated users. 

Regardless, I find it an interesting app for two reasons.  First, with the external connection, Facebook turns into a voicemail server.  Now, it may only be for calls between FWD users, but still, it's an interesting place to store the voicemail messages.  If you buy into Facebook as a "portal" for communication, this provides a nice integration of your voicemail along with your Facebook email messages, wall posts, News Feed, etc. 

It gets even more interesting if you can attach a PSTN number to your FWD account.  I don't see a way to do that right now.  I know for a while in the past there was going to be a "FWDin" service, but I don't recall seeing that launch and can't see any sign of it on the FWD web site right now.  Given, though, that you can connect a FWD client to multiple SIP accounts, there's probably some way to go and do it...  but in any event, think about how that then would work.  You could give someone a phone number and if you weren't there, the voicemail message would ultimately wind up inside of Facebook.  Reinforcing the value of Facebook as a communications portal.

[Side note - since your voicemail is now inside of Facebook, does it fall under the terms of Facebook's TOS (which I wrote about here and here) where basically Facebook now owns all your content?  And you grant them a non-exclusive right forever to do whatever they want with your content?  What it if is someone calling with confidential information?  FB now has that.... Or do they NOT have the voicemail messages because they are actually residing on an FWD server?  Hmmm.]

The second reason I find it interesting is because the "FB voicemail bridge" is a SIP device (and IAX, so I am led to assume it's an Asterisk box).  If it's a SIP device it can have connections from other SIP devices... and so now we have a SIP connection going into Facebook in a manner of speaking.

Facebook and SIP.  Interesting.  Walled garden meets open standard.  (although only to leave messages)

Anyway, this is all really mere speculation because the connection from the FWD service is in private beta testing right now. Still, it's intriguing to me to see as an app.  What do you think?

Over on Blue Box, I uploaded on Friday what I consider one of the best overviews about SIP security that we've done: Blue Box Special Edition #20.  I recorded the interview out at VoiceCon San Francisco in August and it's with Cullen Jennings who is a Distinguished Engineer at Cisco Systems, but more relevant to SIP is one of the Area Directors for the Real-time Applications and Infrastructure (RAI) area within the IETF.  Basically all of the proposals for RFCs relating to SIP roll up under the RAI area.  Cullen's also quite interested in and knowledgeable about security and in fact several of the security-related RFCs related to SIP include Cullen as one of the authors (as do a number of the current proposed Internet-Drafts). 

So he knows his stuff... and being a frequent presenter, he's also good at distilling complex things down into more simple descriptions, so it was an enjoyable interview that I think you will also find quite educational.  If you're working with SIP, or considering it, I'd highly recommend you listen to the show.

FYI, for those of you attending the Internet Telephony Conference & Expo in Los Angeles on September 10-12, I'll be participating in a panel session that is part of Ingate's SIP Trunking Seminar Series.  I expect it will surprise no one to learn that I'll be on the panel about "Enterprise Security and VoIP" wearing my VOIP Security Alliance hat.  My particular session is Tuesday, September 11, 2007, from 9:30-11:00 am.  (And yes, I guess it is appropriate in a way to be talking about security on 9/11!)   More details and the schedule are available online.

The sessions are free and open to anyone to attend.  Simply fill out the pre-registration form.

News breaking out today is that Google has acquired GrandCentral for something around $50 million. GrandCentral is a service that gives you one phone number that can ring multiple numbers, provide one common voicemail - and all sorts of the other features (see "howitworks" for a list of features). As  the GrandCentral blog entry says:

We started GrandCentral because we wanted to create a service that puts users in control of their voice communications and not the other way around. As you have discovered, with GrandCentral you get all of your phone calls through just one number that never changes and you can link and ring up to six phones to ring when somebody calls you. But that’s just the start. You can set different rules for each caller (some ring all your phones, other can go straight to voicemail), create personal voicemail greetings for each of your callers, and even check your voicemail on the web with all of your messages in just one inbox. We’ll even save your messages for as long as you want.

I first learned of GrandCentral quite some time ago from Andy's blog and subsequently heard GrandCentral CEO Craig Walker talk out at O'Reilly's Emerging Telephony conference at the beginning of this year.  It seemed to be an interesting service, although unfortunately I didn't sign up for the service at the time. (Now you have to wait to be invited if you want to try it out.)

As to Google's motivation, they discuss it in the Google blog entry:

GrandCentral is an innovative service that lets users integrate all of their existing phone numbers and voice mailboxes into one account, which can be accessed from the web. We think GrandCentral's technology fits well into Google's efforts to provide services that enhance the collaborative exchange of information between our users.

GrandCentral offers many features that complement the phone services you already use. If you have multiple phone numbers (e.g., home, work, cell), you get one phone number that you can set to ring all, some, or none of your phones, based on who's calling. This way, your phone number is tied to you, and not your location or job. The service also gives you one central voice mailbox. You can listen to your voicemails online or from any phone, forward them to anybody, add the caller to your address book, block a caller as spam, and a lot more. You can even listen in on voicemail messages from your phone while they are being recorded, or switch a call from your cell phone to your desk phone and back again. All in all, you'll have a lot more control over your phones.

So will we ultimately see voicemail inside of Gmail?  One would assume that we will eventually see integration with GoogleTalk, which would give that service its first direct PSTN connectivity.  With a GrandCentral integration, GoogleTalk essentially winds up with a "SkypeIn" kind of service that can route calls to you on GoogleTalk.  The "WebCall Button" and "Click2Call" services also fit in with other Google efforts to expand further into "click to call" (as you can do now in Google Maps).

All very interesting to see... congrats to the GrandCentral team and it will be very interesting to see what emerges from the integration.

Although I could not earlier find many Facebook apps that I considered true "VoIP apps" that linked out to the PSTN or larger VoIP world, we are seeing more apps emerging.  Two notable new entrants:

1. VoIP User Presence by Dean Elwood

Dean's a friend who runs VoIPUser.org (and provides the SIP voicemail box for Blue Box comments) and he's been experimenting with the Facebook API.  This is the first of what he says will be a series of Facebook apps.  Dean says this one is "SIP meets Presence meets Google Maps" and provides this on the Facebook apps page for a description:

"This application shows within your profile if you have a SIP device currently logged into the VoIP User server and are available to take calls. The application page also shows a Google Maps mashup page showing your current location."

There is also a thread going on in the VoIPUser.org forums about the application with more info, screenshots, etc.  It will definitely be interesting to see what else Dean cooks up.

More feedback on this one once I have a chance to actually use it.

2. Skype Me by Nabil Naghdy

This one does the rather obvious and lets you see the Skype presence of your other Facebook friends (provided they have installed the application). With a tagline of "skype meet facebook.  facebook meet skype." the developer says on the FAQ:

"SkypeMe is a Facebook application that links with your Skype account and lets you make calls right from inside Facebook. You can see which friends are online, make calls, and even buy SkypeOut credit from Facebook."

It appears to work primarily by getting the web presence of each of your friends and making that available to you, which means, naturally, that you need to enable Skype web presence inside the options of your Skype client.

There's actually two parts to it.  On your public profile page visible to everyone else, there is a box like the image shown on the left that lets another Skype user initiate a call to you.  On the internal page inside of my Facebook account for the applications I have installed, there is a page with a screen like the graphic on the right that shows all the Facebook friends I have who have also installed the SkypeMe Facebook application.

This Facebook app was announced in the Skype Mashup public chat.  Actually, I think it was really the first I saw announced (but then again, I wasn't really watching the chat last week while I was off).

Speaking of the Skype mashup contest, it was officially announced last Friday.  It will be very interesting to see what developers come up with!

Yesterday, Bandwidth.com announced that their SIP trunking service would be powering Mitel solution centers across the US.  From the news release:

Bandwidth.com, a leading nationwide provider of complete business communications solutions, today announced that it will be powering all Mitel(R) Solution Centers across the country enabling customers and VARS to preview innovative solutions, including SIP Trunking technology in a live environment. Mitel operates solution centers in five locations; Chicago, Costa Mesa, Atlanta, New York and Herndon (Virginia), all of which will be equipped with Bandwidth.com's SIP Trunking VoIP solution by the end of June.

There's been a relationship between Bandwidth.com and Mitel since last September. This announcement yesterday is a logical evolution of that relationship.

There's a lot to write about the incredibly disruptive power of SIP trunking... I don't think we yet fully understand how the power to obtain SIP trunks from anywhere in the world is going to so severely disrupt the global telecommunications infrastructure.  With IP, geography no longer matters... and there are all sorts of local carriers - and tax authorities! - who I don't think fully understand how much this messes up their business models.  I really need to write that up........

Two notes for Bandwidth.com:

1. On the positive side, they have to get credit for one of the coolest graphics I've yet seen for SIP trunking!  I'm talking about the image above that is also on their bandwidth.com/mitel page.  I'm going to have to see about getting permission to use that graphic in some presentations... I just really like it from the design side!

2. On the less positive side, it continues to astound me the number of companies that do not immediately post their news releases on their web site "news" area!  This news release went out yesterday (June 11) but yet it's still not on Bandwidth.com's news page!  It's too bad, because they are missing out on a good potential for inbound links to their site.  Instead, we're left to link to either TMCNet or PR Newswire, both of whom I'm sure don't mind the traffic.  Our (Mitel) PR team have moved to getting the news releases posted on the site right away... I don't know the stats on what kind of traffic we get, but I do know that it lets bloggers like me link directly to the site if we want to.

(See also Ken Camp's commentary about the rising importance of SIP trunking in SMB.)