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Slashdot was abuzz with the news that Skype was going open source. Not so fast my friend. Have you not forgotten Skype founders Niklas Zennstrom and Janus Friis who now run JoltId are in a legal fight with Skype?

Surely Skype wouldn't be so bold as to release their P2P 'secret sauce' while still in court litigation with JoltId. Who would want to thumb their nose at the all-powerful judge? In fact, the lawsuit reads: "A source code version of the GI Software (Global Index Software) is licensed by Joltid to Joost, allowing Joost to be the first company to successfully deliver television and other video content in real-time over a peer-to-peer network. An executable-only object code form of the GI Software was licensed by Joltid to Skype, a well-known Internet-based company that providers users throughout the world with free or low-cost telephone services over the Internet. Skype did not obtain a license to the GI Software source code, however, and the license it did obtain was terminated based on Skype's breaches of the license agreement."

Well, it turns out Skype doesn't have balls of steel, but rather they are making the GUI open source on the Linux platform. Whoopdeedoo. So the graphical user interface is going open source. Move along, nothing to see here.

Ribbit Mobile Launches

November 3, 2009 9:59 AM | 3 Comments
ribbit-mobile-web-admin.png Ribbit today announced the release of Ribbit Mobile, a free communication suite designed for mobile professionals who need to be able to easily manage all of their phone numbers, communication devices, and how & when they are contacted.

Ribbit Mobile is a cloud-based service that lets you manage calls, messages and phones. It's hard not to compare this new offering to Google Voice. One advantage of Ribbit Mobile is that it can make calls from within your browser using their Flash-based software client. Ribbit Mobile features include voicemail transcription, smart call routing, and web-based calling. During the introductory (beta) period, Ribbit is making their "Professional" package free.

Since it leverages Ribbit's open VoIP platform, it can even call your Skype, MSN or Google Talk account, as well as SIP devices. Additionally, when a new voicemail arrives, Ribbit can notify you via email, Google Talk, Skype, or SMS. Ribbit Mobile transcribes voicemail to text and sends it to you via SMS and email. It also provides a web portal inbox where messages can be managed and stored forever. No voicemail retention limit which is nice.

Ribbit Mobile also has something they like to call "Caller ID 2.0" which links your Ribbit contacts address book with your contact's Facebook, Twitter, LinkedIn and Flickr accounts. You can see recent updates made by your contacts to their social networks and pictures of your contacts as seen here:
ribbit-mobile-callerid-20.jpg
blabbelon-logo.jpg
Video gamers love VoIP in video games but have often been disappointed with poor voice quality - including jitter, lag, and choppiness, inability to continue to talk via VoIP once you close the game and just an overall poor user experience. Well, Blabblelon aims to change all that with the launch of their browser-based VoIP application which uses a mix of Java and Flash to VoIP-enable any PC, Mac, or Linux computer. The beauty of Blabbelon is that it works at the browser-level, so you can VoIP your friends whether you are inside the video game or not. With Blabbelon, you can blab all you want - even start your own blab-a-thon if you want.

Blabbelon is not to be confused with Babylon where the story goes that God wasn't too pleased with the Tower of Babel the people built, so he took one common language and confounded it into many so they couldn't understand each other. Well, Blabbelon may not be a Star-Trekkian universal translator-- or some anti-Tower-of-Babel gizmo helping you to understand what that French guy who just fragged you said, but it will give you crystal clear wideband HD voice.

Leveraging Skype's wideband SILK codec the audio quality is superb. In fact, as far as I know, this marks the first time anyone has embedded the SILK codec into Java. When I interviewed Blabbelon, I asked them if anyone else had successfully embedded Skype's SILK codec within Java and they confirmed they are the first. They pointed out that it took serious coding and some tricks to get the SILK codec embedded into Java. Dean Elwood, CEO of telecom provider Voxygen Limited and chief technology strategist of Blabbelon explained they are using a LAMP architecture on the back-end and the front-end is a combination of Java and Flash. Dean said, "Because Skype is only releasing binaries and not source code. The naked binaries are not Java, so we had to do a few tricks to get Java to work nicely with binaries for Mac and Windows. It's not an easy thing to do. Getting it into the browser we had to do some work."

The browser-based VoIP chat tool not only leverages Skype's SILK super wideband audio codec, but it can handle thousands of simultaneous users - up to 7,000 in fact. According to Blabbelon, "Blabbelon provides a platform for a wide variety of users such as: gamers securely competing in team-based quests; businesses running 7,000 person global conference calls; or grandparents taunting grandkids over a game of Facebook Scrabble."

I took it for a test drive and it worked pretty well with very good voice quality. I was able to hit a "hot key", by default the right Ctrl-key and then talk to my other test account. Here's a screenshot (click for larger image):
ring2skype-logo.jpgRing2Skype, a new startup, brings yet another way to have free calling by leveraging both Skype and the PSTN. Ring2Skype allows you to have a free local number (in NYC, London, Madrid, etc.), forwarded to your Skype (wherever you are) -- once again for FREE.

I should point out that Skype charges you for a PSTN telephone number while Ring2Skype offers you a free number. Also, Ring2Skype provides numbers in more than 100 destinations, while Skype only has about 20. You just need to select the city where you want the number and Ring2Skype emails you the telephone number, which you can share with family and friends. Example: 212-555-1000 x1000. The local phone number assigned is shared amongst other Ring2Skype users, so you have to enter an extension, but hey it's free, so you can't complain!
captain-kirk-kick.pngAccording to Infonetics Research, VoIP services brought in $21 billion for service providers in the first half of 2009. According to their report, the number of residential/SOHO VoIP subscribers is forecast to top 225 million by 2013. In other words, with about 310 million Americans that means 75% of all residential subscribers will be VoIP by 2013. Buh bye old analog PSTN landlines!
[RE: Image - Hey just be glad I didn't pick an image of Captain Kirk vs. Gorn]

"Demand for residential and business VoIP services continues to grow even as spending in other communication areas tightens. For the first half of 2009, the worldwide VoIP services market grew to $20.7 billion. Residential VoIP services remain healthy, comprising the majority of worldwide VoIP services revenue, and subscribers are up 14% from the end of 2008. On the business VoIP side, while managed IP PBX revenue growth has slowed in line with IP PBX shipments, we are expecting IP Centrex and hosted UC service revenue to grow 26% year-over-year in 2009," explains Diane Myers, directing analyst for service provider VoIP and IMS at Infonetics Research.

Report Highlights:
- IP connectivity services currently make up about a third of total VoIP business service revenue, growing to 40% of the total by 2013 (managed IP PBX services and hosted VoIP and UC services make up the balance)
- The current sweet spot of the North American IP Centrex market is small business (those with fewer than 100 employees)
> Roughly two-thirds of all IP Centrex seats sold in the first half of 2009 went to small businesses
- While the largest VoIP services opportunities are in North America and EMEA (Europe, Middle East, Africa), demand for VoIP services is growing fast in Asia Pacific and Central and Latin America
- For the first half of 2009, Japan's NTT, France Telecom, and Comcast in North America retain their leadership as the world's largest residential VoIP service providers, together holding nearly 20% of the world's VoIP subscribers
simplesignal-logo.jpgHosted VoIP is making great strides, but there is often one problem - with hosted VoIP you don't get desktop call control. Well, today SimpleSignal introduced its Call Control Toolbar, which adds advanced business VoIP calling features while leveraging their hosted environment. Personally, I am very computer-centric and hate managing my phone calls using the phone. Why use an antiquated TUI (telephone user interface) to setup a conference when you can use a GUI and drag-and-drop conference participants? So I've enjoyed the use of several desktop call control applications over the years, including Comdial's CTI app, Artisoft TeleVantage (R.I.P),  Fonality's HUD client, and more. I've also used Vonage's (hosted residential service) Vonage Pro, a feature-rich desktop call control app.

simplesignal-call-control-toolbar .jpgIn any event, the Call Control Toolbar (image right) is an Adobe Air-based application that floats anywhere on a user's desktop for instant access to a range of calling feature controls. The Call Control Toolbar does not require users to launch a web browser.

"One of the biggest challenges we discovered in our customer research was that our Hosted PBX customers were not using incredibly efficient calling features like simultaneous ring, remote office and voice messaging, or even more basic functions such as speed dial," said Dave Gilbert, founder and president of SimpleSignal, "either because they didn't know how or didn't want to be bothered with logging into a web portal to set up or change their IP phone settings.

"The SimpleSignal Call Control Toolbar eliminates these hassles for users," Gilbert continued. "The controls are user-friendly and always available in the background on a user's desktop, so it's easy to make quick, on-the-fly changes to IP phone settings. "

The SimpleSignal Call Control Toolbar also features call notification screen pops. When a user receives an incoming call, a small window pops up at the bottom of the user's computer screen with complete caller ID information. Users can also answer the call directly from the screen pop.

The Call Control Toolbar is provided free of charge for SimpleSignal's Hosted PBX customers.

Features that can be managed by the toolbar include:
  • Simultaneous Ring for up to six numbers
  • Remote Office, which displays your office number on caller ID regardless of which phone is being used
  • Outbound Dialing by entering the phone number or copying and pasting into Toolbar window
  • Speed Dial for six numbers
  • Sequential Ring for up to six numbers
  • Voice Messaging Control (http://www.simplesignal.com/feat_Visual-Voicemail.php), including email notifications and transfer upon dialing "0"
  • Call Forwarding Always
  • Call Forwarding Busy
  • Call Forwarding No Answer
  • Do Not Disturb
microsemi-powerdsine-9001G-40-sp.jpg
Today, Microsemi is launching a new line of midspans which feature integrated lightning protection. This marks the first time a PoE midspan features lightning protection. As more and more PoE devices are used outdoors, including outdoor wireless Access Points (APs), outdoor security cameras, and more, having shazam.jpg lightning protection is critical. One lightning bolt strike and it can wipe out your entire networking and IT infrastructure. While lightning is great & all since it bestows Captain Marvel his super powers, when it comes to networking equipment, not so much.

The new PowerDsine 9001G-40/SP is a single-port, high-power Gigabit solution for remotely powering both current and emerging high-power networking terminals and equipment while also providing surge protection optimized for outdoor devices such as security cameras and wireless LAN access points.

microsemi-powerdsine-9001G-40-sp-back.jpg
                 Rear product view

According to Microsemi, "Until now, PoE could only be deployed safely outdoors with a costly surge-protection unit installed alongside it, to prevent direct or nearby lightning strikes from damaging or destroying both the powered devices and their network switch. Surge and lightning protection units can cost from $250 to $400."

"A high percentage of outdoor PoE installations, especially in the enterprise, do not implement surge protection units because of their high cost or a lack of awareness," said Sani Ronen, senior product manager for Microsemi's PowerDsine midspans. "Nearby lightning strikes are much more common than most people think. Strikes as far as a mile away can induce a voltage level that can damage outdoor devices. Without lightning protection, surges also can move quickly along the Ethernet cable to damage expensive indoor network switches. Our new midspan delivers that protection at a cost savings of as much as 50 percent."

Microsemi's newest midspan has been tested successfully to the GR-1089 lightning standard for surges up to 2500V/500A. It can power devices designed for either the IEEE 802.3af standard, or the recently ratified IEEE802.3at standard for high power over Ethernet applications.

The 9001G-40/SP offers up to 40W of power which means it can support power-hungry devices such as 802.11n access points, pan-tilt-zoom cameras and WiMAX customer premises equipment.

"The 9001G-40/SP was designed to deliver the latest benefits of the new high-power 802.3at standard while also offering a cost-effective, single-box solution for powering and simultaneously protecting outdoor PoE systems from lightning damage or destruction," Ronen said.

KEY FEATURES
- Interoperability with IEEE802.3af and IEEE802.3at devices
- Lightning protection up to 2500V/500A
- 10/100/100-BaseT support

Pricing and Availability
The PD-9001G-40/SP Midspan is available in production for a unit list price of $119. Additional information is available at Microsemi's website here: http://www.microsemi/PowerDsine.com.
ibm-smartcube.jpgIn a fascinating deal, IBM and Digium announced today that they are teaming up to offer Asterisk for Smart Cube, a customized version of Asterisk Business Edition. IBM's Smart Cube is very similar to Microsoft Small Business Server (SBS), a pre-packaged bundle of various IT & business applications - except in this case Smart Cube is based on Linux not Windows.

Businesses using Smart Cube can be up and running with a complete IT solution to run their business, including the operating system, integrated middleware, database, security and back-office functionality such as file, print, backup and recovery, and more. Extending the IBM Smart Cube to IP telephony and unified communications is a natural extension of the Smart Cube.

Asterisk for Smart Cube has administration capabilities built right into the Smart Cube Smart Desk GUI, which is perfect for the SMB. Asterisk initially ran on rPath but now uses the very popular CentOS distribution. IBM on the other hand is very partial to SuSe Linux. thus one of the technical challenges IBM and Digium worked on was getting Asterisk Business Edition to run on the SuSE Linux platform. Additionally they worked on seamlessly integrating it into the Smart Desk GUI.

How this affects Digium's own home-grown Switchvox SMB offering remains to be seen. But Digium gaining access to IBM's huge distribution and reseller channel is great news for Digium.

Via internetnews.com
If you're running Office Communications Server or LCS whatever you do, don't install KB974571 or you will cause the LCS or OCS server to state you are running the evaluation version and then say it has expired. Poof! - There goes your corporate unified communications (UC) followed by a parade of your coworkers carrying pitchforks & torches looking to find out why their UC isn't working. How widespread this patch is affecting OCS/LCS installs worldwide is unknown.

KB97451 is also known as  "MS09-056: Vulnerabilities in CryptoAPI could allow spoofing" and it was released just yesterday on Patch Tuesday. If you use WSUS or Windows Update to automatically download/install patches, you might be surprised this Wednesday morning to find your OCS/LCS server has failed. Didn't we have Patch Tuesday cause another major outage? (See: Skype Outage & Skype Outage caused by Windows Update)

If you apply the patch, the LCS/OCS services fail to start and you see these errors in the event log:

Event Type:        Error
Event Source:    Live Communications Server
Event Category:                (1000)
Event ID:              12290     
Description:
The evaluation period for Microsoft Office Live Communication Server 2005 has expired. Obtain the released version of this product and upgrade to the non-evaluation version by running setup.exe

Uninstalling the patch is the only way to fix it. According to the Microsoft OCS blog, "The issue is currently being escalated, but until a fix can be found, delaying the install of KB974571 is recommended."

You can also read more about it here.
The big news last week was how AT&T grants denies us mobile VoIP freedom. Just last week I wrote how AT&T had "granted" us mobile VoIP freedom by finally allowing SIP port 5060 on their 3G data network. I was able to use siphon, a SIP-based application on my jailbroken iPhone to register with TMC's IP-PBX and make/receive phone calls.

Well apparently, AT&T must read my blog or something, because once again SIP port 5060 is blocked. I tried siphon on my iPhone this morning and now it won't register. I tested port 5060 and it is indeed blocked. Alas, so no more SIP calls using siphon. The AT&T Overlord giveth, and the AT&T Overlord taketh away.

Hey FCC, can you give AT&T a little smackdown for me please? Just last week AT&T announced it was allowing VoIP over 3G, so why the inconsistency? Is it because siphon is a jailbroken app and not an "official" iTunes app?  I doubt it. I doubt AT&T is able to detect what iPhone application is making an outbound connection. That would be a form of spyware if AT&T knew exactly which iPhone applications you were using. So seems to me that AT&T is still blocking VoIP calling over their 3G data connection. AT&T is talking a big game about being "open", but they aren't backing it up. I'm sure they'll blame it on some network glitch or something. AT&T could be dragging their feet on allowing VoIP over their 3G network hoping the mere "huge" announcement of support for VoIP over 3G will get the FCC off their backs. It's time for the FCC to stop AT&T's stalling tactics. I want my mobile VoIP freedom and I want it now!
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