Disruptive Telephony

Four reasons I am choosing NOT to cut the landline cord

Twelve days ago I asked the question, "Do I cut the landline cord and move my new home phone number into the cloud?", and the responses have been great to read. Today, I can write the answer...

No, I will NOT cut the cord.

Around noon today my landline in Keene should be installed by Fairpoint Communications (who recently bought all of Verizon's landline business in Maine, NH and Vermont).

Why did I finally give in and get a landline installed? Four reasons:

1. FAX - Unbelievably to me, perhaps the primary reason for keeping a landline is an old archaic technology that I absolutely can't stand... fax. This was brought home to me during the process of closing on the purchase of our Keene home and the sale of our Burlington home. As much as we may hate it, there are still some transactions that require fax. There were documents that had to be faxed to the bank. Documents that had to be faxed to lawyers. Documents that had to be faxed to real estate agents. To contractors.

   To a techie like me, it was unbelievably annoying not to be able to simply use email. But in many cases, it came down to this:

   Documents required our signatures.

   Because we still haven't come up with an agreed upon "digital signature", we as a society rely on good old hand-written signatures.

   Now in some cases I was able to scan in those documents and email them off. But not everyone would accept those documents by email. Some of the folks I had to interact with needed them by fax. There were also times when fax was admittedly faster than scanning in the doc and attaching it to an email message (and perhaps I need a better scanning solution). Just put the pages in the document feeder, punch in the number and hit send.

   Now I know there are solutions like eFax (which I use for inbound faxes) but I haven't yet found one that works in the way I need it. I've also seen that fax over VoIP lines doesn't always work well. So for the few times a year when I need fax, I seem to need a landline. (And the problem is that typically when I need to fax something, I really need to fax it for some critical reason.)

2. 911 - As was mentioned in the comments to my original post, "guaranteed" access to 911 is certainly a consideration. Not as much for my wife and I as for our daughter or visitors/guests. My wife and I can pick up our cell phones and dial 911. But if something were ever to happen to one of us, I want our daughter, or anyone else visiting us, to be able to simply pick up a phone and dial 911 and have the emergency services come.

3. DSL - My choices for Internet access in Keene basically come down to Time Warner Cable or DSL. Since I've been using them since the early 1990's back in the dialup / uucp ages, I'm going to be going back to using local ISP MV Communications (who is even now still handling all my personal email) for DSL access. The thing is that getting DSL is easier with a landline. The MV folks said they can do a "standalone" install without an actual phone line that I'm paying for (as I understand it, they would basically have the link they need installed) and if the reasons above didn't enter the picture I'd probably pursue it.

4. The Cloud isn't quite ready - After writing my last post, I spent a good chunk of time trying to figure out how I could get this to work. How could I build my "abstraction layer"? Unfortunately, as I mentioned in my last post, the only service I could find today that gets you most of the way there is GrandCentral, but it still has problems. For instance, I have this perhaps archaic desire to have an area code 603 phone number and GC doesn't have any. I also don't want to have to press "1" to accept a call on a given phone. I just want to answer.

   So it seems like I would have to build my own. Now the pieces are certainly there. I can get phone numbers from any number of SIP providers (although perhaps not my desired 603). I can get call-in numbers for services like Skype or Yahoo (or AIM or MSN or Gizmo). Heck, I can build much of the abstraction layer using Voxeo's app platform (and I probably will as an experiment). Write some CCXML scripts and away we go.

   But the question is - in the midst of everything else I am trying to do - do I really want to be building and *maintaining* a phone number abstraction layer for my home phone? (And the equally important corollary: do I really want to be responsible for it when it inevitably breaks when I'm off on a business trip and suddenly my wife can't get calls at home?)

   No, I don't.

   Now maybe there are other services out there that I don't know about (feel free to pitch me in the comments if you offer one), but for the moment I think I'll let the cloud evolve a bit more. We'll see... maybe in six months or a year there will be better options out there.

So that's the scoop. For the moment, I've got a landline. We're paying the extremely basic rate plan (where if I make any long distance calls on it they are at 12 cents a minute!) and we'll see how it goes.

Fun, fun, fun...

Technorati Tags: phones, phone numbers, voip, sip, telephony, telecommunications, grandcentral, google

Do I cut the landline cord and move my new home phone number into the cloud?

UPDATE - May 21: Today I posted my answer to the question...

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In our new home, do I get a land line?

Or do I move our home phone number into "the cloud"?

We're closing on our home in Keene, NH, next Thursday and as we get set with the utilities that is one of the key questions on my mind. Do I actually "cut the cord" and NOT sign up for a land line with Verizon/Fairpoint?[1]

On one level, we don't need it. My wife and I both have our cell phones. Our daughter is six and isn't yet at the age to make phone calls. I work in the world of voice-over-IP and can certainly get a solution there.

Why should we get a land line?

ADVANTAGES OF A LAND LINE

In thinking about this, it seems to me there are the following reasons to get a land line:

1. 911 - UPDATE: As PhoneBoy reminded me in a comment, the overarching reason for having a landline is 911! A landline is the only guaranteed way to dial 911 and have emergency services arrive at your house. Precisely because it is tied to your geographical location it does indeed provide this critical function!

2. SECURITY/AVAILABILITY - Even when the power is out, your land line still works. VoIP solutions are tied to your Internet connection - which requires power. Okay... but how often does your power really go out? Or... how often does the power go out and your cell phones aren't available?
   
   
3. NO BATTERIES/CHARGING - Similar to the above, landlines (at least, the wired landline phones) don't need batteries and are always available. Cell phones need recharging.
   
   
4. LOCAL DIRECTORY LISTING - Your landline is listed in your local phone book. It's in the "411" directory. If people in your community want to find your number, they can dial 411 or look in the local phone book. With cell phones or VoIP, you aren't in those directories.
   
   
5. LOCAL PHONE NUMBER - Your landline is tied directly into your local exchange and so you have a phone number with an prefix that is "known" in your local community. Neighbors will "expect" that your phone number has one of the local prefixes. With cell phones or VoIP, your number may be somewhere else (and even have a different area code). However, in this era of "10-digit dialing" here in the US, and even more so with people simply programming numbers into their cell phone directories, does having a local number even matter?
   
   
6. LOCAL DIALING CHARGES - One point of having a local phone number is that others in your neighborhood (maybe only a few) may still have dialing plans from Verizon/Fairpoint that cost more if you dial outside your "local" area and so it may cost more to call you. However, in this era of "unlimited calling plans" these local charges are pretty much nonexistent for the vast majority of users.
   
   
7. MULTIPLE HANDSETS - How many times in your household in the past did multiple people get on the same call through using different handsets? (Or in how many households did someone eavesdrop on a call by silently picking up another handset?) A landline gives you this option where a cell phone does not (easily, at least). Still, how many times do you actually do this these days? And with the increasing quality of speakerphones, even on cell phones, is there really the need for multiple handsets? (And yes, you could do this with a VoIP solution.)
   
   
8. COMMON IDENTITY - Until recent years, it's been the norm that a family has had one number that was their identity. "Oh, yes, you can reach the Yorks at 660-9675." There was one number that would reach your household. Today in the era of ubiquitous cell phones, this concept is going away. It's not "you can reach the Yorks" but rather "you can reach Dan York at.... " and "you can reach Lori York at..." It's not "our phone is ringing"... it is "your phone is ringing".

DISADVANTAGES OF A LANDLINE

It is also easy to highlight the reasons not to get a landline:

1. EXTRA COST - FOR WHAT? - Why should I pay my local carrier for a phone I almost never use? I simply don't call as many people any more, even for business, and very often make those calls on my cell phone. Who do I call on my home phone right now?
   • local vendors/contractors when I need to get something fixed
   • local stores to find out their hours or if they have something
   • delivery of pizza or Chinese food
   • family members usually once a week (but we've moved much of this to cell phones because of the "unlimited" plans)
   • very occasionally friends (but we've moved more to cell phones, IM and email)

   Who calls us?

   • family (but they'd call whatever number we gave them)
   • friends (but they'd call whatever number we gave them)
   • political campaigns and charites
   • people responding to Craigslist postings (but they'd call whatever number we gave them)
   • other parents of kids at our daughter's school (from the number in the school directory or from the phone book)

   That's about it... so outside of the people in our community (like the parents) who might look up our number, most other folks get our number from us. So they would use whatever number we have.

   I'm already paying for my cell phone - why should I pay for another phone that I seldom use?

2. LACK OF MOBILITY - The landline is by its nature locked to our house. If a call comes in and I'm outside, I have to run to get the phone - or carry a wireless handset. But if I already have my cell phone with me, am I then carrying two handsets? And if I'm traveling, I can't get the phone calls to my home number (unless I've forwarded it).

Those are really the key factors. My cell phone is almost always with me. Now currently at home I leave my cell phone in my office at night, so if a call came in while I was sleeping I wouldn't easily hear it. But with one change of habit I could simply bring it into the bedroom and have it there to receive calls.

So do we need a home landline?

BUILDING AN ABSTRACTION LAYER IN THE CLOUD

Perhaps of all the advantages I outlined above, the one of most interest to me is the "common identity". I like having a single number that family and friends can call and reach either my wife or I (or, we know will soon be the case, our daughter). If I cut the cord and drop the landline, can I maintain that identity?

The reality is that in this era of VoIP it is possible that I can maintain that common identity through a very simple action:

Push the phone number up into "the cloud".

Move the phone number that you give to everyone up into the VoIP cloud. Think about it... with a service like GrandCentral (now Google-owned) or similar services, I can give everyone one number that rings:

• my cell phone
• my wife's cell phone
• my SkypeIn number
• any VoIP handsets I have in our house (if I actually get around to installing Asterisk or any of the other IP-PBX systems (or my employer's Prophecy app server))
• any other phone numbers I want to have it connect to

In fact, I can phase this in and add/remove numbers as I evolve services. Start out with my our current cell phones. Change that as we get new cell phones. Add the VoIP handsets when I set something up. Remove them if I change the system around.

I have incredible flexibility if I move the number up into the cloud.

CHALLENGES WITH THE CLOUD

There, are though, some challenges with this approach:

1. AVAILABILITY - The PSTN has been around 100 years now and the folks who run it have a pretty good clue about how to keep it running. Even as our landlines pass from the age-old world of Bell (now in Verizon) to a new company, Fairpoint, it's still all in the world of telco solidity. On the other hand, the cloud has a certain amount of inherent fragility. Networks break. Computers fail. Routers get clogged up. Packets get dropped. DO I TRUST THE NEW "2.0" COMPANIES TO GET MY PHONE CALLS TO ME?

2. BUSINESS STABILITY - For that matter, do I trust the new companies to be around? The telcos that run the PSTN and provide landlines aren't going anywhere. Due to regulations, legislation, emergency services, etc.... as well as certain (usually older) parts of the population that will never part with their landline... due to all of that the telcos will be here probably as long as we have phones. (Perhaps smaller, or amalgamated... but still here.) Can the same be said of the "2.0" companies? It sure looks like Google will be around for a while, but will they keep their "beta" GrandCentral service around? Who wants to bet on the long-term viability of Vonage (another option)? Look what happened to all of the SunRocket customers...

3. TRUST - Do I trust these new companies with my data? The telcos have all sorts of legislation regulating what they can do with my data... both my identity (address, phone numbers, etc.) data and also my call detail records. The new companies really have no such limitations, do they?

4. LOCAL NUMBERS - While the whole notion of "area codes" here in the US is fading into irrelevancy with the rise of "unlimited" calling plans, I still have this perhaps quaint and archaic desire to have a "603" number if I'm living in New Hampshire. To those of us who have grown up with area codes, there is still a geographic connection that is of interest. Some of the "2.0" services can get phone numbers in your area code... others can't. (For instance, Grand Central doesn't have 603 numbers right now.)

5. LOCAL NUMBER PORTABILITY - With the PSTN and the telcos, I do have a degree of portability of my phone number. I can move my telco-assigned phone number to another service (but not always back). But it's not at all clear to me that I have that with the 2.0 companies. If I have a number with GrandCentral, can I later move it Vonage (or vice versa) or to SkypeIn or to Gizmo or somewhere else? From what I've seen, that's not likely to be an option (you currently can't move a GC number). I don't like lock-in. I want to be able to move to another provider if I don't like the current one. I want to be able to take my number with me! It's part of my "identity". I want to control it.

6. EASE OF USE - For all its faults, the PSTN has one thing going for it - it's insanely easy to use. Pick up the phone. Talk. No buttons to press (with typical wired phones). Cell phones have certainly added complexity, but we do seem to be doing okay with that. Adding a cloud-based abstraction layer has the potential to add more complexity.

   For instance, GrandCentral rings the range of devices you have indicated and requires you to press 1 to accept a call on that device. So when my cell phone rings, I have to:

   • Take my phone out of my belt holster (or potentially find the phone if it's not on me).
   • Press the green "talk" button to accept the incoming call.
   • Listen to know if this is someone calling me directly or a call coming in from GrandCentral.
   • If a GC call, press "1" to accept the call
   Now, GrandCentral requires this "press 1" stage presumably so that they can hold on to the call and ultimately route it to messaging, but it's an annoying step and one that has caused numerous GC calls to go to voicemail by the time I find the phone, figure out it is a GC call and then press 1.

   Now in fairness maybe there's a way in GrandCentral to configure it differently, but I couldn't find it. I just want to accept the call on the end device of my choice and as soon as I "accept" the call on that device - whether it's picking up a handset or pressing the green button - I want to start talking to the caller.

7. MESSAGING - So if you can ring a whole bunch of phones on different services, where do your voice messages go if you don't answer the phone? Today we have a home "answering machine" where we get all our messages. We walk into our kitchen, look at the machine, see how many messages there are... and start listening. When we first moved to VT in 2005 we tried the Verizon voice messaging service that effectively moves messaging into their cloud. You picked up your landline and if you heard a quick set of tones you knew you had messages. It was nice, in a way, that if you were on the phone and someone else called they would automagically go to voicemail. The thing is... we usually forgot to check for messages. Especially after we were on a call with someone. It is not intuitive to hang up a call and then immediately pick up the phone again to see if you have messages.

   I think you do need some kind of message waiting indicator. That's largely why we dropped the Verizon central voicemail and went back to a home answering machine. When we get a message there's a blinking light (in fact, it blinks on all our wireless handsets).

   So how does this translate into the "2.0" world? Going back to GrandCentral, because they retain control of the call, they can route it to your voicemail box there at GC and then send you an email saying you have voicemail with a link back to the message. With my Blackberry 8830, this works out rather well because I just click the link in the email message, confirm that I want to "Open" the link and... ta da... the audio file is downloaded and played on the 8830's speakerphone. (Paying Verizon for the data download, naturally.) It works out well because I get the message in several places (it actually goes to a Gmail account that is then pulled down to my MacBook and also sent to my Blackberry, so I can read it wherever).

   Is an email enough of a message waiting indicator? I don't know. Can I configure it to send the email to both my wife and I? (Sure, if not directly through GC then through setting up an alias somewhere.) If I set up some voip phones at home could I somehow configure them with an MWI? I don't know.

8. TRANSFERS - In the world of the landline, a "transfer" to someone else involves handing someone a phone or having them pick up another handset. With an abstraction layer, you are answering on different devices on different systems. Obviously you can still physically pass the phone to someone. I remember hearing of a service that let you press a number and essentially park the number and pick it up on one of the other devices in your account... but I can't remember what that service was (GrandCentral does not seem to list this as a feature if it was them). I don't know that I'd realistically ever need this feature, but it's interesting to think about.

9. LOCAL DIRECTORY and 411 - One detail with a number in the cloud is that it won't be listed in the "yellow pages" directories that are passed out by the local telcos. Nor will it be in the 411 directory or probably in any of the online phone number directories. Now maybe this is fine. From a privacy perspective maybe it's good to not be in the directories. And anyone entering my name into Google can quickly find a page of mine with a phone number on it. (This happens to work for me because of my prolific writing and public life. Someone less prolific/public would have a harder time being found.)

10. COST - In end, what's a cloud-based solution going to cost? Today, the cost can be free (GrandCentral), $35-ish per year (Skype or Gizmo call-in number, which could be redirected to other phones), or $25/month (Vonage and friends). Plus, naturally, the cost to accept and make calls on the different devices. But we'll already have cell phones with essentially unlimited calling (at least, for the amount of calling we do). Will the cost stay this low? I don't know.

CONCLUSIONS (such that they are)

Are these challenges surmountable? Can I truly "cut the cord", not install a landline and push my home phone number out into the cloud?

I don't know yet. It seems like an interesting experiment to at least try (I can always get a landline installed later). I like the idea of building an "abstraction layer" in the cloud that lets me control the devices associated with my phone number.

I still get concerned about the challenges #1 and #2 that I outlined. Can I trust companies like GrandCentral to always get my phone calls to me? Can I trust that they will be around for some time?

My next step I think is to dig into a bit more what options there truly are out there for pushing my number into the cloud and building an abstraction layer. GrandCentral is obviously one option. I could build my own using some of the VoIP application platforms out there (including that of my employer). Are there other services that compete with GrandCentral? I need to investigate a bit more. (Suggestions are welcome)

What do you think? Should I do it? Or should I get a landline? Or should I just stick with multiple cell phones and forget about the "common number" concept?

In the end, I look forward to the day when we're done building the IP interconnect and we can purchase phone numbers just as we can domain names today. Why shouldn't I be able to do so? I can go to any of a zillion registrars and spend $10/year for a domain name that can point anywhere. I can move it between registrars. I can change where it points. I remain in control of that domain name.

Why shouldn't I be able to that for a phone number?

(And maybe I will be able to for some other SIP identifier that is not a "phone number", per se, but can be reached by other phones... but that's a subject for another blog post some other day... (and one that I saw *some* other VoIP blogger writing about but I can't for the life of me find that post!))

[1] Fairpoint Communications bought Verizon's land line business up here in northern New England.

P.S. And yes, I could do all of this running Asterisk or something like that on a server in my home network - but I don't want to do system administration! I don't want to set up a server. I don't want to maintain and upgrade a server. I don't want to deal with security issues on a server. I don't want to have to deal with connectivity issues to that server. I just don't want to deal with servers, period! I'll pay, if I need to, to make those sysadmin/security/reliability/availability problems someone else's problems!

Technorati Tags: phones, phone numbers, voip, sip, telephony, telecommunications, local number portability, grandcentral, google

AOL launches OpenView API and gives us half a phone connection...

Does accepting SIP connections at your SIP proxy constitute an "API"? Does providing SIP termination services to the PSTN constitute an "API"?

Those were the questions I found myself asking after AOL announced yesterday their "Open Voice API" (also see CNET article). Since I work with voice application platforms, I'm always interested in new voice APIs and naturally had to check it out.

WHAT IT IS

I have to admit it took some time to figure out what the "Open Voice Program" really is, even after reading the program page and the accompanying blog post. Largely I think the issue was that I was looking for something more.

So here's what is going on. As part of their "AOL Voice Services", AOL has a service called "AIM Call Out". This allows a user of AOL Instant Messenger (AIM) to make outbound calls from their AIM client to the regular phone numbers on the PSTN for competitive rates (under 2 cents a minute here in the US).

From a network topology point-of-view, what happens with the call is that the call goes from your AIM client to the SIP gateway on the edge of AOL's network across some SIP trunks provided by some SIP Service Provider (one of the carriers probably) out to the PSTN. Picture it something like this:

Now what the "Open Voice Program" does is allow you to use an external SIP client to connect to AOL's SIP proxy and make outbound calls. So for example, if you had one of the following:

• a SIP softphone like X-Lite on your laptop or PC
• a SIP "hardphone" on your desk or any of the many WiFi SIP handsets
• a SIP softphone on your dual-mode handset (such as those from Nokia) that would let you make connections over WiFi

you can use the service to make outbound calls using your AIM Call Out connections and minutes. The picture would now look like this:

The SIP phone connects to AOL's SIP gateway, logs in with your username and SIP password and makes the outbound call. You now do not have to be at your PC and can make calls from another device that may be more convenient (like your dual-mode phone or a WiFi SIP handset). In fact, you don't need to be logged into AIM or have the AIM client running anywhere. The SIP device makes calls completely independently.

CONFIGURATION

The process of configuring your SIP phone to work with AIM Call Out is relatively straightforward. (Assuming you have an AOL/AIM "screenname". If you don't, you need one of those first.)

1. Go to the AIM Call Out page and click the Sign Up Now page.

2. Once in the "Dashboard", buy some Call Out credit (I bought $5 for the sake of testing) through a process that is not exactly intuitive but involves:

1. Choosing Billing->Payment Method and setting up a credit card.
2. (the non-intuitive part to me) Clicking the "Add Credit" link in the upper right corner of your screen and then going through that process.

3. Still in the Dashboard, assign a "SIP device password" on the Settings->SIP Clients page:

4. Switch to your SIP device/phone and configure it with this information (along with the possibly the STUN server found on the config page).

5. Start making calls.

Like I said, it's a relatively straightforward process.

You naturally have to agree to their Terms of Service whose main point (made repeatedly) can be summarized as "This is not a telephone replacement for emergency purposes."

INITIAL USAGE

After configuration it worked fine (once I remembered how to configure X-Lite to use the correct microphone and speaker devices) and I made several calls with no problems. I actually wound up calling into today's Squawk Box podcast (where we discussed this AOL Open Voice Program at some length) using the service. It all worked well.

AOL is to be commended for their support of open industry standards like SIP!

While my initial experience was positive from a user point-of-view, there are to me a few problems.

NO INBOUND OR PC-TO-PC CALLS

First, the service is outbound-only. You can make calls from the SIP device, but you cannot receive them. While AOL is very clear about this on the Open Voice Program page, it still is a disappointment. Now, AOL does not appear to have a "Call In" program like Skype's SkypeIn or Yahoo's PhoneIn that ties a PSTN number to your AIM account, so you can't get inbound calls from the PSTN. (I thought they did at one point but I can find no sign of it on AOL's Voice Services pages.)

But you also can't receive inbound calls from other AIM users! It seems to me that if the idea is to make it more convenient to use AIM's voice services, you ought to be able to receive calls from other AIM users on your SIP device. Perhaps this is a future development. (So from a technical point-of-view, they are currently not operating a SIP registrar. Your SIP device does not register with AOL's server.)

NO SIP-TO-SIP CONNECTIONS

My second disappointment was that it did not appear to support direct "dialling" of SIP addresses. I tried both Blue Box comment lines, "sip:bluebox@voipuser.org" and "sip:9992002622@sip.voxeo.net" (with and without "sip:") and the result in both cases was a message "Sorry, that number cannot be dialled":

Now, granted, maybe 0.01% of the public out there actually has an interest in direct SIP-to-SIP dialling, but that is the world in which I move... both with Voxeo's platform and also the work I do with the IETF. If we are to ultimately build the massive interconnect that let's us have an IP-only network, we need to have services and devices that let us do direct SIP connections.

(NOTE: I fully admit that I may not have used the X-Lite client correctly in calling these SIP URIs. I believe I did, but if someone else can get this to work, I certainly am open to my failure being operator error.)

SECURITY?

Given my background, you can't NOT expect me to say something about security, eh?

The good news is that they do have you use a "SIP device password" that can be different from your regular AIM password (assuming you set it to be different). Smart move. Well done. (if people use different passwords)

Per the the Open Voice Program page, AOL supports RFC2617 Basic and Digest Authentication and I confirmed with a packet trace that they are, in fact, using Digest authentication. While that isn't incredibly secure (it basically involves a MD5 hash of a password and a server-supplied nonce), it is pretty much what the industry is using right now.

Also like most of the industry, they are not encrypting the SIP with TLS and they are not encrypting the voice with SRTP. Just plain old unencrypted SIP and unencrypted RTP. Given that almost no one else in the industry is doing this, I can't exactly fault them.

SO WHO IS GOING TO USE THIS?

In the end, though, I still do have to wonder who will use this. (And that was part of the Squawk Box discussion today.)

If you already use AIM as your primary IM service and are an existing AIM Call Out customer - and are a techie enough to own and configure a SIP device - then this may be a great way for you to make cheap calls using another more convenient device. (Although some have pointed out there are cheaper services, but again it's the convenience of already living inside of AIM.)

I do like the fact that you can use any SIP device. This is a stark contrast from, say, Skype, which requires either that the device be connected to an always-running PC, have Skype embedded inside of it, or use another software program like iSkoot that essentially proxies your Skype connection. AOL's program lets you use any device you want - and you don't need to be logged in to AIM. So for folks more comfortable with a "hard phone", they can use any of the SIP hard phones out there.

The detail here, however, is:

How many AIM users are actually using AIM Call Out?

Especially when there are competing services from Skype, Yahoo and Microsoft that offer similar rates as well as inbound calling. How many users will actually be able to make use of this?

Perhaps AOL is hoping that this will attract users...

On a different note, it is not entirely clear to me how developers might use this. The AOL blog post from Mark Blomsma in the AOL Developer Network talks about how developers can use this new "API", but I'm still missing... for what?

Essentially it seems to me that all you could really do is create an application that uses AIM Call Out for SIP termination to the PSTN - with the AIM account providing the authentication and also billing/charging. Perhaps that will be useful. I'm not sure.

FINAL THOUGHTS

I am, though, intrigued by one line of the AOL blog post:

Part of the Open Voice program is AIM Call Out.

So are we just seeing the beginning of this program? If so, that's not at all clear - from what I read - anywhere on the pages.

Overall, it's great to see AOL using SIP and it's great to see them opening up their infrastructure in some small way. I'm not sure I'd call this an "API", exactly, but let's hope that this is just the beginning and that they will do more in the months ahead. We'll see.

Meanwhile, I've still got $3.84 of calling time to use...

What do you think about this service? What's in it for developers? Is there value in having access to the AIM credentials for authentication? Or is this ultimately just yet another SIP termination plan?

P.S. For those curious, my 55-minute call into Squawk Box today cost $0.94 or about 1.7 cents a minute.

Technorati Tags: aol, aim, aim call out, aol open voice, open voice, voip, sip

Creating voice applications that interconnect with Skype and SIP

Given that I write on Voxeo's blog site, I tend not to write much here about things we are doing at Voxeo.

But I thought I'd mention here one specific post I put up recently called "Skype-ifying your voice applications" which talks about the intriguing ways in which you can use our hosted platform to make voice applications accessible through a number of different mechanisms.

As shown in the diagram to the left, an application that you write and is hosted on our platform can be called into over the PSTN, over a direct SIP connection or via Skype or FWD. Likewise calls can go out to PSTN numbers or to SIP endpoints.

This flexibility is one of the many things that intrigues me about the platform (of which I knew nothing about prior to joining the company in October).

Anyway, more information is in the full blog post. I just thought I'd mention it here. (By the way, if you'd like to try it out yourself, developer accounts are free.)

Technorati Tags: voxeo, skype, sip

Nortel's fascinating move into open source telephony... but NOT with Asterisk

Nortel and "open source telephony"? Huh?

That was admittedly my thought when I received the list of who was going to be on the panel I moderated last week at VoiceCon on open source telephony. The other two panelists were obvious choices: Bill Miller was from Digium (makers of Asterisk) and Raza was from 3Com who have recently announced that they would be reselling a version of Digium's Asterisk Business Edition. Both Bill and Raza made sense to me. But Tony Pereira of Nortel? Nortel does not leap out at me as a company working with open source telephony - what in the world are they doing with it, I wondered?

It turns out that the answer is... "quite a bit!"

As Tony Pereira outlined in our panel as well as in conversations afterwards, Nortel is in the process of launching their "Software Communications Server 500" (SCS 500) targeted at small businesses and built using open source telephony software!

Interestingly, though, it does NOT use Asterisk.

Instead Nortel is using the "other" major player in open source telephony, the "sipXecs" product from SIPfoundry.org. (Previously called "sipX" but renamed "sipXecs" about a year ago.) I've not written all that much about sipX here but it certainly has been a product I've known of over the years. It started out as a PBX product from Pingtel which they then released as an open source version ("sipX" and now "sipXecs") and also had a commercial version called "SIPxchange". sipX garnered perhaps its most attention back in October 2006 when it was announced that Amazon.com would be using it for their internal phone systems (see the links on the SIPFoundry.org site). At a fundamental level, sipX provides similar functionality to Asterisk but where Asterisk is focused on being a "platform" for telephony that can work with a wide range of protocols, sipX is focused exclusively on SIP and also provides an extensive GUI management tool. (Pingtel provides a (obviously biased) comparison of sipXecs vs Asterisk on their wiki.)

From what I learned at our panel, Nortel is essentially creating their own supported version of "sipXecs" that they will sell as the "SCS 500". It will have full commercial support from Nortel. Target market will apparently be "small" businesses. No info really available on Nortel's site yet, although glimpses are visible through support documents (such as here and here(although this appears to be about an earlier 1.0 version last year)).

All in all it's to me a fascinating move by the folks at Nortel and I look forward to learning more about the SCS500 product over the next weeks and months as they launch it. It's a rather nice boost for the whole world of "open source telephony", too, to have Nortel making this move as well.

Technorati Tags: nortel, open source, asterisk, sipx, sipxecs, pingtel, voip, sip

As I wrote over on Voxeo's "Speaking of Standards" weblog, one of the ironies of the language we use in this space is that we all have been talking about "SIP trunks" for a few years now, but nowhere has there actually been a formal definition of what exactly a SIP trunk really is!

Jonathan Rosenberg has now offered a definition in a new Internet-Draft titled "What is a Session Initiation Protocol (SIP) Trunk Anyway?" Here is the abstract:

The term "Session Initiation Protocol (SIP) Trunk" has become almost commonplace amongst vendors and SIP providers. Even though the notion of a 'trunk' has a well defined meaning in circuit switched systems, it has never been defined for SIP. This document provides a formal definition for a SIP trunk, discusses its scope and applications, and establishes best practices for identification and security of SIP trunks.

The document makes for good reading even if you are not overly familiar with the concepts behind SIP trunks. Jonathan is looking for feedback and there will I'm sure be continued discussion on this topic.

Technorati Tags: sip, sip trunks, ietf, jonathan rosenberg

The EComm 2008 Interview with Skype's Jonathan Christensen should be required reading...

As we enter into the final month before eComm 2008, I would suggest that the interview with Jonathan Christensen, Skype's general manager of audio and video, should be required reading for anyone seriously interested in this space. Why? Well, in part because Jonathan Christensen does provide some good information about what Skype has done and is doing but also because it provides some good insight into what one of the people driving Skype's agenda is thinking about this space. Take one of the final paragraphs where he answered Lee Dryburgh's question about what he saw as the the future of communications (bold emphasis added by me):

Well, a big question I guess and, having worked on the space for quite a while, I think that it's only going to get more interesting over the coming years since, well, like this open spectrum for example. You know, I just have to reiterate, I think that anybody who has not figured out that the Internet is the platform and that there isn't any such thing as walled gardens that will survive, or sub-networks [such as AOL tried] that are going to survive, those people are doomed. The intersection of these worlds is going to be chaotic. It's going to be violent. It's going to be messy for a while but it is going to happen, and the Internet will survive as the one open platform. You are going to see a trend towards extreme innovation at the edges - on the devices, in the PC platform, in software, all around the edge of the Internet.

I think that you are only going to see further disruption of the telecom industry and the emergence of totally new businesses that we can't imagine today. I think that [the] net result, that drives me every day, is that we're going to have this very rich, open, cheap and accessible communications. This is going to be not just a game changer for the telecom industry, but will be a change agent for all of humanity. So, a platform that allows us all to see each other and hear each other more clearly maybe makes us a little bit less crazy, less polarized and more open as a world society.

Good stuff... and the whole interview is worth a read. Given my recent criticism of Skype, I'm particularly pleased to read the comments I emphasized in bold. Jonathan Christensen will be giving one of the keynotes at eComm 2008, March 12-14 in Silicon Valley and if you haven't considered going, I would encourage you to do so. It should be a great event!

P.S. I also wrote about this interview in relation to SIP over on Voxeo's "Speaking of Standards" blog.

Technorati Tags: skype, ecomm, ecomm2008, telecom, jonathan christensen

I'll be speaking at Ingate's SIP Trunking Seminars at IT Expo in Miami next week

If any of you will be in Miami next week for Internet Telephony Expo, I will be speaking on VOIPSA's behalf at Ingate's SIP Trunking Seminar Series held in conjunction with IT Expo. Predictably, my session from 8:30-9:45am on Thursday, January 24th is titled "Seminar/myth 1: VoIP is not secure".

If you are going to be down at IT Expo, do check out the full schedule for Ingate's SIP Trunking Seminar Series. They have a good range of speakers and the seminars are free.

If any of you are attending either IT Expo or the SIP Trunking Seminar Series, please do drop a note as I'm always interested in meeting readers.

Technorati Tags: sip, sip trunking, ingate, itexpo, conferences

A SIP phone for the iPod Touch! (Just add microphone...

Fascinating development on the Apple frontier... in late December some developers posted information about a SIP phone for the iPod Touch! They included this helpful demonstration video:

The team has obviously received a lot of questions and has therefore released a lengthy FAQ list. If you have an iPod Touch, you can download the software. Of course, you really need a microphone to use it... which the Touchmods folks are building.

All in all an interesting development. I look forward to seeing how it moves along!

Technorati Tags: apple, ipod touch, sip

SIP phone soon to be available on iPhone and iPod Touch?

Will there soon be a native SIP client on the iPhone and iPod Touch? Dameon Welch-Abernathy writes on his VoIP weblog that some developers have gotten a basic SIP stack working on the iPhone and iPod Touch. The limited details available are over on The Unofficial Apple Weblog:

iPhone hacker eok writes to let me know that he and Samuel have gotten SIP registration and signalization working. They took a few mobile terminal shots, but the real work is being done via ssh. Samuel is working on connecting the audio in/out to the pjSIP. If you have iPhone or iPod touch coding skills and want to get involved in the project, connect to #touchmods on irc.undernet.org. It looks like most of the work will be done on European time.

As you can see in the screenshots, this is still very early in the development. Still, it's great to see this kind of development taking place.

Technorati Tags: sip, voip, apple, iphone, ipod touch

Introducing "Speaking of Standards", a new Voxeo blog about industry standards, IETF, W3C, SIP Forum, etc.

A large part of why I have NOT been writing here all that much in the past few weeks is that I've been busy in my new role with Voxeo working on a corporate blog portal. I've been covering a bit of that odyssey over on my Disruptive Conversations blog as well as in my weekly reports into the For Immediate Release podcast. It's been a great amount of work but also a lot of fun - I've been very lucky to have a colleague who does amazing things with CSS and graphics, and so the sites look a whole lot better than they would if I were left to my own devices.

I'm very pleased to say, now, that we've reached the point where I'm willing to link to our work and talk a bit about what we are doing. The main blog portal is the predictable "blogs.voxeo.com" but the weblog that we're really starting to use and could be of interest to readers of this blog is our "Speaking of Standards" blog found at:

http://blogs.voxeo.com/speakingofstandards/

I've obviously been very occasionally writing here about standards and some of that may continue, but I expect most of my writing on the subject will now occur over on this new Voxeo weblog - and I'll naturally be writing more on the subject. We'll be writing about the IETF and SIP standards, but also the W3C and standards such as VoiceXML and CCXML that I've never covered at all here. We'll be linking to events and tutorials we find and generally providing whatever information we can about standards affecting our industry, as well as Voxeo's views and implementations of those standards.

Why would Voxeo sponsor a weblog about standards? Primarily because the company and our products are all about open standards - which was one of the things that attracted me to the company after they first approached me. I've since learned that they've been leading the IVR industry in adopting open standards. As the products page says in the "Fast Facts" sidebar:

• 100% Standards based IVR
• Supports W3C VoiceXML 2.0
• Supports W3C CCXML 1.0
• Supports W3C SRGS 1.0
• Supports W3C SSML 1.0
• Supports CallXML 3.0
• First platform with XML call control
• First platform with XML conferencing
• First shipping CCXML implementation
• First SIP/VOIP IVR platform

Not bad, eh? Add to that the fact that our CTO (my manager), RJ Auburn, chairs the W3C's Working Group on CCXML and we've hired other folks involved with standards efforts... all of that is why we added a weblog on standards.

So if you would like to see our view on industry standards, find tutorials about various standards or learn about standards-related events we may be attending, I would invite you to come on over and check out "Speaking with Standards" - or subscribe to the RSS feed. While I (and others) will still be working on improving the site, it's mostly done and I'm delighted to be able to return to writing more. Let us know what you think!

Technorati Tags: ccxml, ietf, sip, standards, voicexml, voip, voxeo, w3c

New Facebook voice app: VoxCall lets you do free calls between SIP phones/numbers

By way of my Facebook NewsFeed this morning, I learned that several friends had installed a new Facebook app called "VoxCall" (must be logged into the walled garden of Facebook to see the link). A quick Technorati search brought me to Alex Saunders' blog post on the subject which clued me in to the fact that this was from the folks at Voxalot, some of whom I'd met down at Fall VON in Boston.

[Side Rant: This also shows the inherent weakness and stupidity of Facebook's current implementation of "groups". The Voxalot folks had posted info about this app in both the news and wall of their Facebook group, but of course I would never see it unless I just randomly happened to go there. Had they sent a message to all group users, I would have seen it in my Facebook Inbox, but it would be nice if instead Facebook had some way to notify you that you had new info in the groups to which you subscribe.]

The VoxCall app is basically a "click-to-call" app that makes use of Facebook's directory. You simply click on the name of someone else who has the app installed and, like many click-to-call apps, you are called first and then the other party is called and the connection is made.

An interesting aspect is that VoxCall works with SIP URIs (addresses). When you install the app you have to enter your SIP URI at which point you then receive a call on that URI where you are asked to enter the PIN displayed on the screen:

It's actually a pretty nice way of authenticating the endpoint. Given that Voxeo's a VoIP application platform company, we naturally all have SIP URIs for our extensions (sip:dyork@corpsip.voxeo.com for me) so it was easy for me to sign up. Users of Gizmo would likewise have a SIP address, as would users of many other VoIP services. If you don't have a SIP URI, Voxalot has a suggested path to get one on their VoxCall FAQ. (One thing I don't completely understand is why you would need to do their step #2, Register for a VoxPremium account, if you already get a SIP URI from the Voice Service Provider you signed up with in step #1. But maybe the point is that some of those VSPs won't give you SIP URIs... ?)

Once registered, the process is quite simple. You have a "Call Friends" tab that is shown below (complete with some advertisement being blocked by the local proxy server that I run that blocks ads from typical ad-serving sites):

You simply click on the person's icon and the call process starts. First it calls you, then it calls the other party. No charges incurred by anyone outside of whatever inbound connection fees we would normally pay (in my case, none). I called Alec and so my page changed to show his picture and the fact that I was calling him:

Alec and I had a good chat with surprisingly good audio quality given the convoluted path our call was taking. I was on a Polycom IP phone connected across the Internet to Voxeo's SIP servers in Florida. The call went across some network cloud to Alec's TruPhone number (which has a SIP URI) which wound up ringing his mobile as he was driving along the 401 somewhere in southern Canada. Audio quality was quite good and didn't seem to have any real issues in the 5 or 10 minutes we chatted.

The VoxCall app also has an Echo Test number you can call to hear the latency and has some conference rooms that I have not yet tried.

Overall, it's an interesting app, although I guess my basic question is simply this: will I use it? As I wrote earlier, the phone is no longer as critical of a communication tool for many people, myself included. When I think of Facebook, I think of it as a place for email-ish communication. If I need to reach someone urgently, I have used Facebook as a place to get a phone number from in the past. Will I think to use to it place a call in the future? I don't know.

There are a couple of barriers to that, really. First, the app only works with people who have it installed. Second, to install it you need a SIP URI and the whole concept of SIP addresses is only really now starting to come to people's attention (outside the early adopter crowd). Third, initiating the call requires going into the VoxCall application page inside Facebook to click on the person's icon to call. It would be nice if it could be done simply from the list of friends that you have. (Having said that, it's actually easier to simply go into the app page than it is to search through Facebook's friend list and then go into their profile to then click on a link below their picture.)

The nice thing about the app, though is that it does use the Facebook directory. As Alec puts it:

Perhaps the biggest differentiator for Voxcall is simply that it hooks into a directory that a lot of people know and use.

As Facebook continues its climb in popularity and moves onward toward the goal of being your definitive "portal" to the Internet, this VoxCall app (and others like Alec's own Free Conference Call app) help connect in voice to the communications mix (for those who still want/need to use it).

In any event, kudos to Voxalot to bringing out another voice app on top of Facebook. It's good to see the platform being used for voice. As a advocate for SIP and open standards, I applaud apps that promote the use of all things SIP. Give it a try. What do you think of it? (Feel free to give me a call if you are a Facebook friend of mine.)

Technorati Tags: applications, facebook, facebookplatform, voxalot, voip, voxcall

I'll be out in Vancouver Dec 2-7 for the 70th meeting of the IETF.

Just confirmed travel plans today - I will be heading out to the 70th meeting of the Internet Engineering Task Force (IETF) in Vancouver, British Columbia, Canada, from December 2-7. If any readers will be out there (either for the IETF or in Vancouver in general), please do drop a note and let me know. This will be my first meeting in my new role with Voxeo and I'm very much looking forward to renewing old acquaintances and also getting more directly involved with the work of the IETF.

Technorati Tags: ietf, sip, standards, voxeo

Did you know RFC 4733 had replaced/obsoleted RFC 2833 for DTMF signaling in SIP?

Did you know that RFC 4733 replaced/obsoleted RFC 2833? I just learned this myself through a SIP Forum mailing list exchange the other day. For those not aware, RFC 2833 and now 4733 define methods of carrying DTMF signals (and other similar signaling) in RTP streams separate from the main audio component of the RTP stream. A typical example of use might be where you were using a highly-compressed audio codec for audio between two SIP endpoints where the high degree of compression might make it challenging for the DTMF tones to be correctly interpreted on the receiving end. Using "RFC 2833 compliant" signaling, the sending SIP endpoint would send those DTMF tones as separate packets within the RTP stream.

My key takeaway from learning about RFC 4733 is that we should really be talking about "RFC 4733 compliant" signaling... but given that the industry is really only now starting to really talk about "RFC 2822 compliant" signaling, I'm not sure I expect to see that happening anytime soon.

Anyway, here's the abstract from RFC 4733 - you can naturally read the rest of the document to understand more:

This memo describes how to carry dual-tone multifrequency (DTMF) signalling, other tone signals, and telephony events in RTP packets. It obsoletes RFC 2833.

This memo captures and expands upon the basic framework defined in RFC 2833, but retains only the most basic event codes. It sets up an IANA registry to which other event code assignments may be added. Companion documents add event codes to this registry relating to modem, fax, text telephony, and channel-associated signalling events. The remainder of the event codes defined in RFC 2833 are conditionally reserved in case other documents revive their use.

This document provides a number of clarifications to the original document. However, it specifically differs from RFC 2833 by removing the requirement that all compliant implementations support the DTMF events. Instead, compliant implementations taking part in out-of-band negotiations of media stream content indicate what events they support. This memo adds three new procedures to the RFC 2833 framework: subdivision of long events into segments, reporting of multiple events in a single packet, and the concept and reporting of state events.

Technorati Tags: DTMF, ietf, sip, standards

Skype and secure SIP? (Why would I see this message?)

Whenever I'm using Skype, I have the "Display technical call info" setting enabled so that I see technical stats about the calls I am on. Those windows tend to stay around after a call... and I noticed this one still around with an identity of "securesip". (click on the image for a larger version) I've tried to replicate this with calls that I've recently made to see if I could get the window again, but can't seem to do so. Anyone know why I might be seeing this?

I'm curious...

Technorati Tags: sip, skype, voip security

Heading to New York today for Interop... speaking tomorrow on VoIP Security

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 audacity of Asterisk - why the 3Com/Digium partnership fundamentally changes the game in SMB telephony

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

Digium buys SwitchVox and gets presence, Web 2.0 interface, mashups to Google Maps, Salesforce.com, SugarCRM...

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

Zoiper - a free SIP / IAX softphone for Windows, Linux or Mac

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

FWD rolls out a "Voicemail" Facebook app... with the promise of calls to the *external* FWD client going to FB voicemail soon (i.e. FB becomes voicemail for SIP connections)

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:

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