January 2006 Archives

With Super Bowl Sunday fast approaching, it’s not surprising to hear about how sports fans are driving HD television sales. According to a new survey "Inside the Mind of the HD Sports Fan" conducted by the Consumer Electronics Association (CEA) and the Sports Video Group (SVG), these same fans more readily adopt new technologies that quickly deliver sports content than non-sports fans.

The survey was unveiled at the first TV Sports Summit held at the 2006 International Consumer Electronics Show (CES) earlier this month, as reported by Millimeter’s HD Focus newsletter today. Highlights of the survey include:

• Nearly 60% of high-definition television owners consider themselves sports fans
• Nearly 50% of HDTV owners cited HD sports programming as the primary force behind their HDTV purchase
• Sports fans said their favorite sports to watch in HD include the Super Bowl (78%), World Series (44%) and college football bowl games (41%)
• 39% of HD sports fans are extremely disappointed when a sports event they want to watch is NOT in high-definition
• 65% of HDTV sports fans say sound is an important component of their sports viewing experience

While I've always been a fan of Adobe's PhotoShop, it just seems that I've been using Paint Shop Pro more and more to edit, enhance and repair my digital photos.

With Paint Shop Pro 9 (a newer version -- X -- is out now), you can cover up elements -- even backgrounds -- you don't want in your photos (perfect when you don't have time to make sure you see the full frame when you need to get a fast shot off), eliminate the colored glow that can appear in high contrast areas, correct dark shadows and bright backlight as well as add cool effects, like vignettes or halftones, to photos. This has all been pretty easy with 9, but apparently X makes it even easier.

Corel is now offering a special offer with orders for Paint Shop Pro X -- a free copy of Paint Shop Xtras — Creative Editions. This package extends the creative power of Paint Shop Pro with more than 600 artistic resources to choose from you, including 250 Picture Tubes and Frames. These ready-made picture frame art elements can be added to photos and graphics with a single mouse click.

Price: $99 (regularly $129) boxed or download; existing users can for upgrade for $59 (boxed) or $54 (download).

I met with Digium's founder, Mark Spencer yesterday to discuss what was happening with Asterisk. Mark told me that they are working very hard on Jingle support, an open set of extensions to the IETF's Extensible Messaging and Presence Protocol (XMPP) for use in VoIP, video, and other peer-to-peer multimedia
sessions. This will enable federation with Google Talk and other Jingle supporters. Mark said that Jingle support will be coming very soon. If Jingle is successful, this is yet another reason to be skeptical over Tello, which I discussed yesterday.

Mark also told me they are participating in more SIP interoperability bakeoffs and tests. He also reiterated the cool Bluetooth integration thast Asterisk has and offered to give me a technology demo at some point.

Mark noticed the Nokia N90 that I have as part of the blogger's review program and asked how I liked it. I told him it was pretty feature-rich but I still hadn't figured out how to increase the handset volume or how to switch to vibrate mode. (assuming it supports vibrate). I hate reading manuals, but looks like I may have to for the N90. Anyway, Mark noticed the built-in camera and he decided to take a photo. Actually, I found out later that Mark took a video of himself, not a photo. I was going to share the video, but I can't figure out how to email the damn thing from the Nokia phone. The N90 keeps telling me the file format isn't supported when I try to email it.

We also talked about a cool new videophone from Grandstream Networks called the GXV3000 which is launching at Internet Telephony Conference & Expo and which Mark said works with Asterisk. I'll post a new entry when I find a photo of the GXV3000. from I have a printed photo of it, but no scanner handy.

I know there was some other interesting stuff Mark told me, but I must have forgotten. Getting up at 5am to fly down to Florida will do that to your memory recall.

A new version of Performancing for Firefox has just been released. The most notable new features are the Technorati tagging support and Delicious bookmark integration. Now that I can tag with Performancing for Firefox I'll have to re-evaluate whether or not to switch from my current WYSIWYG blogging software that I use (HTMLArea). Not being able to tag was a stumbling block for me. Oh yeah, the new version also supports Trackback URLs, autodiscovery, and ping support - all critical features.

Here's a list of the new features:

• More Technorati Support
• Delicious integration
• Trackback support (including auto-discovery)
• Draft support (and default to draft option)
• Ping support
• Statusbar Icon Enhancments (drag and drop)
• More Settings Options
• Full metaWeblog API implementation, now working in Roller, blogharbor, and many more
• Lots of Bug Fixes
  • Logging in from account wizard now times out if connection fails
  • More informative Blogger.com error reporting
  • Hitting a tab twice no longer looses data.
  • Less confusing automatic technorati tag adding
  • Numerous metaWeblog API fixes
  • Edit post now conserves original dateCreated parameter
  • Issues with setting and editing categories (drupal still has issues).
  • New posts no longer created when editing a post (on WP)
  • Preview links now open in a new tab
  • Spaces are now possible in technorati tags
  • Wordpress 2.0 support
  • Many more small bugs

Also new to this release:

• Support and Documentation
• PFF Buttons and Banners

Cell phones have become indispensable -- can you imagine not having one?

But can you guess how many people around the world now use cell phones? Do you want to even try?

Well, many more cell phones than jelly beans in that jar that we have always been asked to guess at school math night -- 2 billion to be exact, according to The McKinsey Quarterly, the online journal of the business consulting firm. With the world population approaching 6.5 billion (according to world and U.S. population clocks), that 2 billion figure is truly remarkable.

Let's see how long it takes Nokia, Motorola, Samsung, LG and all the other cell phone manufacturers to get a cell phone to everyone on earth. Anyone care to guess?

Today at Internet Telephony Conference & Expo, Digium and Ranch Networks have announced that they are teaming up to make Asterisk a secure and scalable VoIP solution. I have a meeting with Mark Spencer, Mr. Asterisk himself at 4pm today to discuss this new relationship, but I wanted to share this news before it hit the news wires.

Digium, the original creator of Asterisk and pioneer of open source telephony, and Ranch Networks, the first IP telephony network appliance provider to integrate security and bandwidth control for IP-based applications, today introduced its security code for the Asterisk open source IP telephony platform. When combining Asterisk with Ranch Networks new RN series appliances, the solution provides unprecedented security and scalability to the open source telephony industry. The code is available for download at www.ranchnetworks.com and ftp.digium.com

Ranch Networks solves the problems associated with VoIP implementations through both its MIDCOM integration with the Asterisk platform, and several of its Patent-Pending technologies. The MIDCOM integration provides for dynamic per-call firewall control, bandwidth management, NAT traversal, and RTP traffic bridging - all supporting encrypted signaling streams, while the Patent-Pending technologies separate voice, video, and data traffic into multiple, secure zones without having to reconfigure IP addresses.

"Security is potentially an issue for SIP deployments since ports need to remain open at all times in order to enable voice traffic-leaving potential open doors for intruders," said Mark Spencer, president of Digium and creator of Asterisk. "In providing this technology, we are proactively addressing the security of VoIP before it becomes a major concern, while also providing quality of service and increased scalability."

Ranch Networks' technology sends instructions to the firewall to open the ports on an as needed basis. Media streams can flow from one phone to another, then close ports when the call is terminated. Instructions are also sent to the appliance determining the amount of bandwidth needed. As calls come in, data is shrunk to allow necessary bandwidth on a protocol basis.Ranch Networks guarantees that voice is given priority over data. This technology supports multiple phones behind any number of firewalls.

"We expect this will change the mindset of large enterprises and service providers that were previously skeptical of VoIP and open source technologies, to look more closely at the customization, flexibility and huge costs savings that a platform like Asterisk provides," said Ram Ayyakad, founder and CEO of Ranch Networks. "We have eliminated any security and scalability issues that could be argued by proprietary vendors."

The Ranch Networks security code has received the highest certification through ICSA.The security code is available by the download of Asterisk 1.2.2 and is available for download from www.ranchnetworks.com and ftp.digium.com

Further, Ranch Networks today introduced the RN series of appliances which are the first to integrate security and bandwidth control for VoIP applications. The company's IP telephony network appliances secure, manage and scale VoIP traffic beyond existing firewall technologies. Ranch Networks' appliances separate voice, video, and data traffic into multiple secure zones without having to reconfigure IP addresses. Several patent-pending technologies, allow service providers to deliver VoIP services to enterprises without affecting existing data networks.

Ranch Networks' technology is designed to work with leading IP PBXs and supports all sizes of enterprise and carrier deployments. Instead of requiring labor intensive access lists or protocol specific application layer gateways, Ranch Networks' IP PBX controlled appliances provide dynamic, protocol independent, per-call authenticated network access. This unique approach both simplifies and increases network security, while allowing encrypted signaling streams.

Starting with Asterisk, the open source PBX, Ranch Networks' offers the RN300, RN20, RN40 and RN41 to provide dynamic, protocol independent, per-call authenticated network access. These products will supply unprecedented VoIP security, bandwidth management, VPN, accounting and switching capabilities to small to mid-size enterprises, service providers and carriers.

"Asterisk was a logical starting point for us as it allows us to leverage a vast growing user base, a focused and enterprising reseller channel and avoid the red tape of the proprietary PBX vendors," said Ram Ayyakad, founder and CEO of Ranch Networks. "We believe now that this puts Asterisk in line with the leading proprietary IP PBX vendors."

The RN300, RN20, RN40, and RN41 are available through leading Asterisk resellers worldwide ranging in price from $750 to $20K.

Today, at ITEXPO, Signate is announcing SigPRO. SigPRO is a a hosted telephony system for telephone service providers with 5,000 to 500,000 customer extensions that Signate claims "sets a new price/performance standard".

I have heard various issues surrounding scalability issues when it comes to the Linux-based Asterisk IP-PBX solution, so I contacted one of my sources and asked about Signate's scalability claims. My source responded, "Signate is in for a shock when they really try to scale this stuff. They are using GPL'd code from Asterisk, and have not licenced the code. This means that they are likely not changing core Asterisk source code to implement scale. This means they are in for a bit of a surprise when they actually start getting hundreds, let alone hundreds of thousands concurrent users on one box."

I pointed out to my source how Signate was achieving scalability by showing him a portion of their news release: "SigPRO achieves further efficiencies because it works in concert with Signate’s Telephony Server 5000 softswitches, which provide better price/performance than PC-based or proprietary switches. In an installation, the SigPRO server provides applications, services, and feature-functionality. Call routing, trunking, translations, and most call-control/management services are provided by Signate's Telephony Server 5000 softswitches. Softswitches are load balanced and deployed in an n+1 configuration for redundancy."

He replied to this by saying, "It takes significant engineering to truly achieve scale, not just shoving it on a box with more horsepower. Under the GNU-GPL of Asterisk, any changes to the source code must be given back to the community. Therein lies the rub, and therein lies Digium's best business model yet."

He then theorized that Signate is using one Asterisk box for SIP routing, this is the "softswitch box", and another Asterisk box for their PBX feature-set, etc. Thus, they may achieving some scalability by distributing the load across multiple servers. The problem is that they would need to actually get into that source code and make significant changes, and hence make community contributions, so long as they are using the GPL'd version of Asterisk.

How Signate achieves their scalability without running afoul of the GPL is something I will be sure to investigate at ITEXPO and report back here. I'll keep you posted. In the meantime, here is the release which will go out on the newswires in a few hours...

Signate Announces SigPRO Hosted Telephony Application That Sets New Price Performance Standard

Signate, the leading provider of VoIP telephony solutions based on industry standard hardware and open source software, today announced the production release of SigPRO, its state-of-the-art hosted telephony system for telephone service providers with 5,000 to 500,000 customer extensions. Targeted at providers who are moving into the IP voice market, SigPRO enables providers to increase revenue, reduce expenses and deliver feature-rich solutions to individual consumers, small businesses, and large enterprises.

"Signate's SigPRO is enabling our market entry with a capital investment of less than $5 per potential customer," said Jason Cohen, CEO of FutureLink, a service provider based in Coram, New York. "That means we can offer very competitive prices and richer service offerings than TDM-based carriers at the same time," he said.

"Until SigPRO, hosted systems for small and medium-sized service providers have either been prohibitively expensive or limited by low calling capacities and minimal feature sets," said William Boehlke, CEO of Signate. "SigPRO enables a new generation of nimbler IP carriers to deliver services to new markets, such as business PBX replacement," he said, "with the same features as the very best PBXs but without the capital investment."

About SigPRO
SigPRO is both feature-rich and cost-effective because it sits atop a foundation of industry standards such as SIP and open source components like Linux, Apache, MySQL, php, SIP Express Router and Asterisk.

SigPRO achieves further efficiencies because it works in concert with Signate’s Telephony Server 5000 softswitches, which provide better price/performance than PC-based or proprietary switches. In an installation, the SigPRO server provides applications, services, and feature-functionality. Call routing, trunking, translations, and most call-control/management services are provided by Signate's Telephony Server 5000 softswitches. Softswitches are load balanced and deployed in an n+1 configuration for redundancy.

Up to eight Telephony Server 5000 softswitches may be deployed in a stack with a call set-up capacity of up to 350 calls per second on seven servers. The eighth server is back-up capacity should one of the active servers fail.

SigPRO includes one interface to a credit card provider and templates for popular customer premise devices such as Linksys adapters and Cisco and Polycom SIP telephones. Customer detail records at the system, reseller, enterprise and consumer levels may be interfaced to external billing systems.

Functionality is delivered through five web browser interfaces that can be branded with reseller or enterprise customer logos and colors.

SigPRO's Administration interface provides access control, DID inventory management, call rating, customer premise device management, and other administrative features.
Service provider contact centers and resellers manage customer care through SigPRO's ITSP interface. Each reseller has its own interface where the reseller services its own customers independently of other resellers, under parameters set by the service provider for that reseller.
Each business customer uses a SigPRO Enterprise panel to manage their own extensions, within the limits of the service plan they have contracted for.

From the Enterprise panel, customers can configure business features such as:

• Auto Attendant / IVR – Enterprises manage inbound calls and deliver them to the intended destination through interactions that can include recorded answers, extension dialing, and time of day pathing
• Conferencing – Conferencing lets enterprises coordinate dispersed teams
• Complex IVR – Enterprises can quickly build their own voice response systems that take DTMF tones or record responses
• The Self-Service interface lets consumers sign themselves up for service without human intervention. SigPRO automatically provisions their customer premise device and charges their credit card to begin service. Each reseller has its own unique Self-Service interface.

The SigPRO user panel gives users control over their own telephone service functions such as:

• Follow Me – Users can set the system to ring numbers such as mobile or home phones in sequence when someone calls their desk phone, so they never miss an important call
• Call blocking and Screening – Each user decides who reaches them and when
• Email Integration – Users easily integrate voicemail with their Microsoft Outlook mailbox
• Voice Messaging – Feature-rich voice messaging provides users with the flexibility to use and manage their messaging service from anywhere

Service and Support
SigPRO is offered with 24/7 technical support 365 days a year by Signate engineers in the U.K., U.S. and New Zealand. On site hardware service is provided by the global SGI support organization. Four hour and 30 minute service level agreements are optional.

Pricing and Availability
A redundant calling configuration for 1,000 simultaneous SIP calls begins at $125,000. That configuration includes a SigPRO server and two Signate Telephony Server 5000 softswitches.

An English language North American version of SigPRO is available immediately. An English version with support for currencies other than dollars will be available in the second quarter of 2006. User interfaces for Spanish and other languages will be available in the second half of 2006.

About Signate
Signate is a leading global provider of design, installation, configuration, training and management services for open source VoIP telephony systems. For more information, visit Signate at www.signate.com

VoIPSter has announced the formation of The OpenZoep Foundation and the release of OpenZoep communications engine. The OpenZoep communication protocol is based on the XMPP Extensible Messaging and Presence Protocol, an open, XML-based protocol for near-real-time, extensible instant messaging and presence information, which happens to also be supported by Google Talk. It also is the core protocol of the Jabber Instant Messaging and Presence technology used by thousands of Jabber servers.

According to VoIPSter, "OpenZoep will enable the proliferation and rapid development of standards based telephony application throughout the world as VoIP grows and continues to take hold with users and service providers." They added, "OpenZoep wants to become the de facto open source VoIP solution that is embedded in software clients, web browsers, games and other mobile or desktop applications," said founder Sjoerd Hannema, CEO of VoIPSter."With OpenZoep all applications can immediately begin offering free PC-to-PC VoIP calls, instant messaging (IM) and both outbound PSTN and SIP calls to free and premium SIP based telephony providers."

"Building VoIP clients using OpenZoep is very straightforward," said Dirk Griffioen, VoIPSter's Senior VOIP Engineer. "The OpenZoep API offers simple methods to communicate with the OpenZoep client-side VoIP engine and server-side web services."

Further, OpenZoep goal is to create "Open Directories" to help finding others easier. Users will also benefit from a shared, global telephone directory that will make finding other people easier. By supporting open standards like SIP and XMPP OpenZoep also becomes interoperable with leading presence and identity based messaging platforms, protocols and services. This previous statement aligns pretty much with what I said about Tello yesterday - that there shouldn't be a need for Tello to act as a hosted (& paid) centralized directory aggregating your presence from various IM services. Instead, there should be "open directories" that let you view the presence/status of IM services that support industry standards. Sounds like OpenZoep has the right idea...

In any event, in collaboration with leading directory providers, OpenZoep will also introduce an initiative to establish a global service provider discovery server. This server allows end-users to pick and choose a service provider that offers PSTN in, PSTN out, voicemail and other telephony services from thousands of Internet Telephony Service Providers.

"Already awkward interoperability issues are plaguing the IM industry. Those issues should be avoided at all cost in the VoIP market," said Sjoerd Hannema. "OpenZoep's founders feel it's not too late to promote open VoIP standards, even when some major players already have adopted proprietary protocols."

OpenZoep has also added so-called XMPP 'stanzas' for working with sound cards, mics and speakerphones as well as other audio input and output devices. These ‘stanzas' simplifies the handling of phone calls and provides easier access for getting configuration information from the engine by making it easier for developers to work with.

With a mission to keep VoIP open and secure, OpenZoep believes computer-based telephony should be based on open source software. Only publicly accessible software allows the open source community to scrutinize the code for safety and quality.

"Since OpenZoep uses Jabber for its presence and instant message handling, it is possible to search for users on other Jabber networks," said Griffioen. "This means user of OpenZoep based applications can benefit from a shared, global telephone directory. To accomplish this OpenZoep will pursue cooperation agreements with leading IM Federations and providers."

Features:

• Free calls, make free calls over the internet
• Zoep>Out, make cheap calls to regular land lines and mobile phones
• Zoep, receive calls from regular phones (coming soon)
• Chat (available in Firefox plugin) 
• Conferencing (coming soon) 
• SMS (coming soon)
• Voicemail (coming soon)

They even feature a Zoep Firefox plugin which adds the client to the Firefox toolbar. I wonder if this is the first VoIP Firefox plugin? In any event, I installed it and played around with it a bit. Just a warning if you do install this Firefox plugin and you use the Firefox SessionSaver plugin -- it crashed my Firefox and I lost all my saved tab windows. Well, I was able to recover them in the prefs.js file, but I had to manually copy/paste each URL one at a time back into Firefox. So forewarning - you can Bookmark your open tabs to a backup group just to be safe before installing this.

Here are some screenshots:

Error I got with my initial password I chose. Must have at least 1 capital letter and 1 lowercase letter and 1 digit? Boy, talk about strict password security!


Here's the icon added to the Firefox toolbar:


Chat window:


Main interface with dial-pad displayed


Making a call. Note the "+1" required in the dial string. I had no credits, so I couldn't test the audio quality. I'm not even sure why it attempted to dial when I had no credit. You'd think it would save "Please click here to buy credits" or something to that effect.


With OpenZoep's any service provider could provide PSTN, voicemail, SMS and other services by following the open standards.By striving for top-notch speech quality, security, and openness; OpenZoep plans to support as many platforms as possible; by offering any version on any platform that is backward compatible for at least all telephony functions. The OpenZoep engine is currently available for Windows, with versions for OS X and Linux being ported.

Using OpenZoep based applications also means user may choose the service provider of his/her liking, while each service provider remains responsible for billing and customer support of their respective customers. Additionally, all providers will use one shared central OpenZoep/SIP directory that allows users to discover and invite friends and colleagues.

The OpenZoepAPI documentation is available at http://openzoep.org/docs/api/primer/
 

Just the other day I wrote about how the small screens of mobile phones directly affects what viewers want to watch on it – short stuff!

Now, today, a bunch of companies that are household names -- including Intel, Motorola and Texas Instruments – are teaming up in an alliance designed to foster the growth of mobile digital TV and accelerate something called Digital Video Broadcasting – Handheld (DVB-H), an open standard for broadcast digital TV reception on mobile devices.  The organization is called the Mobile DTV Alliance.

As mobile video entertainment gains increased awareness and achieves greater availability, the Alliance intends to focus on promoting the best practices and open standards that deliver premium-quality broadcast television to mobile devices for the North American market. The open, industry-supported standard is expected to foster growth throughout the wireless market with more choices and allow mobile DTV handsets and services to reach us faster and at a lower cost.
 
In their scheme, mobile devices capable of decoding DVB-H signals will enable us to receive live TV programming directly on our phones and other mobile devices. In addition, we will benefit from on-demand and interactive programming that would utilize the cellular network. (Good idea as revenue generator for the operators; good idea for us?).

Apparently, there are more than 10 DVB-H network trials that have either concluded or are currently underway around the globe, including Australia, Finland, France, Germany, Italy, the UK and the U.S.

Another development to watch ... (Maybe mobile phone screens should be larger?)

The Philips VP5500 WiFi VoIP phone was announced way back in September 2005 and it was finally launched today in the Netherlands of all places. No offense Netherlands, but how come you get first dibs on this cool phone? Anyway, the sleekly styled VP-5500 is powered by Linux and lets users enjoy live video calls using its built-in VGA camera (640x480 resolution) that rotates up to 240 degrees and supports 30 FPS. Video calls are displayed on a 2.2" color LCD supporting 64k colors. The VP5500 features a video out port that lets others watch the video on a TV. Though hooking up a video wire kind defeats the purpose of using a wireless videophone, don't ya think?

You can also zoom in on captured still images stored on the phone's internal 1MB memory. It also features a built-in speakerphone and hands-free headset compatibility.

As previously mentioned, the Philips VP-5500 VoIP Videophone runs on Linux, so they've built this phone around "standards", such as Wi-Fi, WPA, and most importantly the SIP protocol standard. The VP5500 can be upgraded wirelessly and will support applications developed by service providers. No date has been set for a release outside of Holland, however Philips is looking to partner with third party operators.

Tello was launched today by Pulver with some help from Craig McCaw, telecom banker Michael Price and former Apple CEO John Sculley. What is Tello? Tello enables enterprise users to see the presence of the person they are trying to reach - whether the users is on a phone, cellphone, etc., regardless of which IM client service they use, i.e. MSN Messenger, AOL/AIM, Yahoo! Messenger, etc. (Note: only SIP-based IM services are supported as far as I can tell.)

The concept is simply to "bridge" the various IM services and be able to see the presence of the user. It features IP-PBX integration with the likes of Avaya and even the popular Asterisk open-source Linux-based IP-PBX, as well as a Blackberry client. Ironically, I just blogged about IM interoperability between AOL & IBM's Sametime and I predicted IM interoperability would finally happen in my 2006 predictions. Thanks for proving me right Mr. Pulver!

IP Democracy has some thoughts on this news worth a look, as does BusinessWeek, Om, SiliconBeat, and the WSJ. Mostly positive reviews from what I read. Also, according to Tello, "The solution is delivered as a hosted Instant Communications and Collaboration network service, Tello Connect, and is available with complementary Desktop and Mobile Client applications. It provides a Real Time Communications Presence & Connectivity Hub that allows people to instantly locate, contact, and connect with friends, colleagues, and partners using the devices and applications that they already know. Using the Tello Desktop and mobile applications, individuals can, at a glance, easily determine the availability of their contacts at any time anywhere in the world and initiate rich media multi-modal communications with the click of a button."

Here's how it works by leveraging the SIP standard. Tello's VoIP federation service is built around a directory which provides SIP routing based on dialed telephone number or SIP URI. When a call is dialed (either via a SIP UA or an IP-PBX), the dialed call is signaled using SIP to Tello Connect which checks several databases and applies a set of customer-defined policies to determine the optimal SIP route. Thus, using this solution which queries multiple IM databases, you can increase the percentage of calls that remain all IP and thereby save money.

If this sounds eerily similar to FWD, the Free World Dialup service Pulver created, a SIP-based direct SIP URI dialing directory, you would be correct. In fact, when I read this news on a few news sites, I thought to myself "Didn't Jeff already try this with FWD?" The added benefit with Tello is that it performs "hosted federation" and integration with just about any IM service that supports the SIP standard, which includes MSN Messenger, AOL, Yahoo Messenger, and several others. At first glance, this hosted solution to the IM interoperability dilemma seems like a great idea. However, it seems like a near-term solution to me. If the big IM players get their act together and decided to use the same SIP/SIMPLE IETF standard, which allows for presence sharing, you wouldn't even need Pulver's hosted solution. In addition, many of the IM players have announced federation plans already, including Google.

Honestly, I'm not convinced a hosted service provider that centralizes and consolidates all of your presence information is going to be a "killer app" that people will pay a monthly fee. The target for this service is the SMB market, not consumers, so it's possible that SMB CEOs looking to improve employee productivity will sign off on essentially "renting presence knowledge". It could especially help sales people quicky and easily reach their B2B sales prospects instead of playing voicemail tag or even email tag. Still, the verdict on IM in the enterprise being a productivity enhancer or productivity waster (chatting with your spouse or friends at work) is still out. I should mention this service won't work with Skype's proprietary chat mechanism, which is the #1 used VoIP application in the world today.

Also, I'm going to assume Tello's hosted service requires your various usernames & passwords to the various IM services in order to logon, authenticate and access your presence information. If that's the case, then Tello will also face an uphill battle against users that don't want to share their various IM client username and passwords. Considering MSN Messenger accounts use Microsoft's Passport which shares the same account as your hotmail.com email address, Hotmail users might be wary over sharing (with a third-party) their password to their email account containing personal and confidential emails. Am I wrong here?

I leave for Internet Telephony Expo tomorrow and I figured I may as well use my blog here as my checklist for stuff I need to do in preparation for the show. I am going to be crazy busy at this show with several appointments, I'm moderating a session, and I hope to blog all the cool stuff happening at the show.

Tom Keating's pre-Internet Telephony Conference & Expo checklist

• Bring Nokia N90 phone to ITEXPO and do some live video blogging (vlogs) of the show. I only have a 64MB MMC card though. Thought the Nokia N90 used SD. I was going to pick up a 1GB SD card so I can do some large vlogs. MMC cards are hard to find at retail, and don't have time to order it online.Anyone wanna lend me a large MMC card during the show?
  
• Bring 250 business cards. I'll probably need em' all too.
  
• Bring both badges. One is a press badge titled "CTO & Executive Technology Editor" and the other is titled "VoIP & Gadget Freak". Actually the other badge is a "TMC Management" badge with the same title that enables me to get into top secret places no else can.
• Bring laptop with DVD player and headphones. (make sure laptop is full charged)
• Don't forget EVDO card for ubiquitous wireless Internet access.
• Fill up Jeep with gas so don't have to fill up in NY or on way home from airport.
• Print calendar of appointments.
• Pack protein bars (come in handy when don't have time to eat a sit down meal.)
• Bring sneakers to give my feet a rest at night time due to walking the show floor in dress shoes.
• Charge and pack the iPod
• Bring Canon digital camera Use Nokia N90 2MP camera instead - one less thing to carry to show floor.
• Write review of the Nokia N90 that I have been neglecting to do.
• Bring Skype Cordless DUALphone I just received in the mail to play around with.

I think I've got everything covered, though I have this feeling that I'm forgetting something. Hmmm...

Looks like my prediction that in 2006 that IM clients will interoperate is coming true. AOL today announced that it has formed a relationship with IBM to connect their instant messaging communities. The popular enterprise IM client, IBM Lotus Sametime will be able to connect and communicate in real time with users of the AOL, AIM, ICQ and Apple iChat instant messaging services.

With today's announcement of Lotus Sametime 7.5, IBM plans to provide instant messaging customers with federated access to AOL's global instant messaging network as an embedded feature for each user. These customers will be able to exchange instant messages with the more than 70 million users of the AOL, AIM, ICQ, and Apple iChat.

Sametime users will also be able to exchange presence information by adding AOL Screen Names and ICQ numbers to their Sametime contact lists while inviting others to add Sametime identities to their AIM Buddy List, iChat and ICQ Contact List features.

According to AOL, now with its new relationship with IBM, AOL's instant messaging services will reach more than 75 percent of all enterprise instant messaging users. AOL's existing federation partners include Microsoft Corporation, Reuters, Inc., Jabber, Inc., Antepo, Inc., Omnipod, Inc., Communicator, Inc., Parlano, Inc., Thomson Financial and Pivot Solution's IMTrader service.

Update: 4:45pm
Just learned that Yahoo also announced interoperability with IBM's Sametime. More proof that IM interoperability is happening in 2006!

From the obvious facts file: A report by Starcom USA released earlier this week (and reported on MediaPost) shows that four out of five early adopters of video iPods and other mobile devices prefer to watch short programs when they are mobile; they are apparently quite happy to watch longer TV programs on (can you believe it!) a television.

Now I don't know how many of you have either watched a full TV show or seen someone watching one on a mobile video device, but I can't image staring at that small screen for 30 minutes, 60 minutes or two hours.  It's no wonder that small portable TVs (those with 3 x 3 screens) never took off big in CE. 

This reminds me of the discussions about how watching TV on your computer was going to be the next big thing.  Seems like the smaller the screen, the less likely anyone wants to watch anything on it for any length of time.

PhoneBoy came across my Gabcast post, a really cool service that lets you use VoIP or PSTN dialing to record audio and post to your weblog (Wordpress, Blogger, Movable Type) and then he posted his first Gabcast. In case you missed that blog post, Gabcast supports Asterisk trunking with 2 or 3-digit dialing (depending on your dial-plan).

Good start PhoneBoy, keep it up and we'll have to call you GabBoy!