Recently in VoIP Category

Vipadia announced the release under the GPLv2 of Karaka, the open-source XMPP-Skype Gateway which connects the XMPP and Skype networks.

Karaka is a scalable distributed XMPP transport that bridges instant messaging and presence between a user's XMPP and Skype accounts. This will for instance enable Skype-to-Google Talk instant messaging. In theory AOL's AIM should work, since I believe they also support XMPP. In addition to full presence and instant messaging exchange, it also supports multi-user chat ("conference rooms"). Karaka implements the XMPP standards XEP-0100 for gateway support, XEP-0045 for multi-user chats and XEP-0144 for roster exchange.

According to Vipadia, "Existing Skype interconnect solutions focus on bridging voice even though the primary use of Skype is for instant messaging and associated presence data. Interconnecting with Skype messaging and presence has been a major stumbling block for many who wish to offer Skype interconnection to their network. Karaka bridges the XMPP and Skype clouds, removing this stumbling block by converting Skype messaging and presence to the popular XMPP protocol as used by, e.g., Google Talk."

Karaka is licensed under the GPLv2 and is hosted on Google Code at http://code.google.com/p/karaka/.

Check it out @ http://vipadia.com/products/karaka/.

In 2006, I came across a Network World article, which espoused the fact that Sam Houston State University (SHSU) had switched from the Cisco CallManager IP-PBX to open source Asterisk. I wrote about this news since 6,000 students and faculty were moved off Cisco to the open source Asterisk IP-PBX, which was great news for the open source Asterisk community. This deployment demonstrated that Asterisk could scale and put to rest one of the main complaints against Asterisk.

Well, 3 years have passed, and according to this thread written by Jason Fuermann, who is responsible for SHSU's IP phone system, SHSU has switched back to Cisco from Asterisk. Say what?

EZ Call, Inc. today announced the launch of EZCallerID.com, a new service that provides enhanced Caller ID, also known as CNAM, for VoIP calls. The hosted CNAM service gives you not just the phone number, but the name of the person calling.

Most SIP trunking providers do not provide the caller's name with Caller ID on inbound calls. EZCallerID.com solves this issue by simply having you route your inbound calls to their server. They insert the caller's name and send the call back to your IP-PBX.

How's it work? Simply put, EZCallerID.com connects to the national databases that contain the name associated with each phone number, perform a reverse lookup, insert the CallerID info into the From SIP header and then send the call back to you.

This is similar in concept to my recent article, CNAM (CallerID with Name) on Asterisk using Reverse Phone number lookup, but in this case, no Asterisk PBX is required. It works with any SIP-based IP-PBX. It is a hosted offering, so there is a fee of course, but it's not  expensive. They only charge $0.015 per call (or $1.50 for every 100 calls). The service is available on a pay-as-you-go basis ($10 minimum initial charge), with no recurring charges or minimum monthly commitment.

Head on over to EZCallerID.com if you want to sign-up.

Hat tip to Eric Hernaez for the news tip

Sony Computer Entertainment Europe (SCEE) announced that it plans to shut down Go!Messenger, a VoIP, video chat, and IM application that launched last February. Citing a lack of interest, Go!Messenger was an intriguing PSP app that leveraged the PSP Go!Cam camera for its video chat capabilities.

Sony explained "Although it proved a popular concept, achieving a significant number of registrations, it didn't gain the number of regular users that BT and SCEE were aiming for."

Go!Messenger will end on March 31. But all is not lost. Skype for PSP still works, but it's too bad Skype for PSP doesn't support the Go!Cam for some Skype-to-Skype video chat action.

Related articles by Zemanta

• Go!Messenger VoIP app for Sony PSP goes bye bye next month (crunchgear.com)
• PSP Go!Messenger No Longer Going Come April [Europe] (kotaku.com)

Rich met with ooma recently to see their latest wares and hear about their current business model. Recently, ooma ditched the 'P2P voice network' idea where users actually "share" their home landline with others and instead became a traditional VoIP broadband provider. Apparently, the privacy issues were too much to overcome, since users were concerns about fraudulent activity happening on their home landline by outside ooma users. I had my own reservations about the business model as well, since they claimed it would take 2,000 strategicly placed ooma boxes in all the various local exchanges to get good local call coverage for free P2P calls.

Besides becoming a traditional VoIP broadband provider, ooma is now going to start offering high-end media phones, that according to Rich Tehrani will in the future feature a picture frame, in-house sensors and cameras. As for what they offer today, in early January, ooma launched Telo, which offers unlimited, free VoIP-to-PSTN (U.S.) calls over the Internet along with a DECT 6.0 cordless phone that supports call screening, MP3 ringtones, 12-hour talk time, HD voice, speakerphone, two-line support, mobile transfer, and intercom. It supports up to eight phone numbers and six phones


                   ooma's Telo phone system with DECT 6.0 handset.

The Telo phone system is expected to be available in the first half of 2009. The next question you're probably thinking is "If it's free unlimited U.S. VoIP-to-PSTN calls, how does ooma make any money?" The answer to that is ooma offers ooma Premier, with advanced features that they hope people will opt & pay for. (See: http://www.ooma.com/company/how_we_make_money.php)

Some of the Premiere features include:

• Instant Second Line allows you to make or take two simultaneous calls from a single phone number
• Blacklists helps you protect your privacy and block telemarketers
• Multiring lets you answer calls from your home phone or cell phone
• Message Screening allows you to listen in as the caller is leaving their message
• Send to Voicemail allows you to transfer a call to your voicemail
• Voicemail Forwarding lets you forward voicemail so that you can listen to it from your favorite email program
• Do Not Disturb allows you to roll your calls into voicemail without ringing your phone
• Personal Numbers allows you to select additional phone numbers in any calling area in the US

The "free" unlimited calling puts them on par with magicJack, but the magicJack is much less expensive (magicJack costs $39.99 1st year, and $19.99/yr in subsequent years). Pricing for Telo has not been announced, but I'm sure it will be much more expensive since the hardware costs so much more. One advantage for Telo is that magicjack requires your PC to be on all the time to make/receive calls over its USB-based dongle. The Telo phone system is a standalone phone that has no such restriction. It's also a multi-line and multi-handset phone platform, so it's more suitable to busy households that require multiple lines or phone handsets.

Check out Rich's post for more on Telo and how the FCC is actually an investor in ooma.

I was driving to work in my Volvo V70 XC (XC=Cross Country) which has all-wheel-drive and Blizzak snow tires when I approached a T intersection where I had to make a right-hand turn. Now normally a Volvo with AWD and Blizzak snow tires can cut through any amount of snow with no problem. I've driven it through blizzards, so I know.

In fact, there was some light snow on the roads in my neighborhood, but once I got to the main roads, they were completely dry. The road I was on is also well-traveled and appeared dry in most places and wet in others. I didn't realize the "wet" spots I saw weren't wet at all, but were black ice.

I tried to stop at the T intersection, my car's ABS kicked in and the car wouldn't stop. I quickly spun my head to my left to see if I was about to cut someone off. I saw a yellow school bus bearing down on me. I decided to go over to the opposite lane, figuring I'd rather get hit by anything else but a 7 ton bus. I hadn't had time to turn my head to the right to see if there was a car coming from that direction. I had other things on my mind - like a big yellow bus headed straight at me!

Not knowing if there was a car coming from that direction and although ABS was still kicking in, I quickly turned the wheel to the right just in case I could slip in between and be riding the center of the lane - hoping if there was indeed a car coming from the opposite direction, it could ride the shoulder a bit to avoid hitting me. It worked! My car did indeed finally grip and take the center lane. Fortunately, there wasn't another car coming from the opposite direction, so I gunned the gas and rode illegally in the left lane for about 30 ft, just to be sure I wasn't going to get rear-ended by the bus, and then got back into the right lane.

I felt bad for the bus driver who obviously had to brake hard. The bus driver probably thought I was a lunatic for taking the turn without stopping. But I honestly wasn't going fast. I was slowing for the stop sign, but just couldn't stop. Although the bus driver probably saw it was an "out of control" action and I did fishtail a bit - so probably deduced it was black ice.

Wow, another 1-2s later and I was a goner.  There was also a line of cars behind the bus, so even if I avoided the bus, other cars would have hit me and there was an SUV just ahead of the bus. So basically, I was able to slip in-between the gap between the SUV and the bus. What if my wife didn't remind me to take the IPEVO Skype phone to work, which I had forgotten in the bedroom? I actually had to walk back down the hallway to get it.

The mere act of taking the IPEVO Skype phone home last night to test may just have saved my life. Thank you IPEVO & thank you Skype! They say Skype is a cost saver - well now Skype's a life saver!

Today, flaphone (formerly Flashphone) announced that users of their Flash VoIP application can now make a call from flaphone to skype. You simply need to enter sip:skype_username@skype after selecting "none"(global)" for the SIP account. I should mention that flaphone supports multiple SIP credentials, which is a really nice feature. I've been testing flaphone for several weeks now and have been meaning to write up their cool Flash-based VoIP application.

In any event, for my first test call I entered sip:tomkeating@skype and pressed the call button. The call was initiated and the call quality was superb!

You can also use this SIP-to-Skype feature for flaphone's CallMe widgets that you place on your website.

Similarly, Gizmo5 recently launched OpenSky which also enables SIP-to-Skype dialing. However, Gizmo5 calls are free only up to 5 minutes long. For longer calls they are offering a paid service. There is no such restriction that I am aware of with flaphone.

By leveraging Flash, flaphone is cross-platform, has minimal download time, and you can run it from any browser. That and the fast that it supports SIP-to-PSTN calling, SIP URI dialing, and SIP-to-Skype calling, means this is one VoIP app you should check out!

Goober Networks, recently launched CallingAmerica.com, which offers web-based free VoIP calls to any landline or mobile phone in the U.S. or Canada. The Web-based offering uses Flash for the audio output & microphone input. As for the business model for "free calls" CallingAmerica.com uses  advertisements on their website that you must watch before the call is initiated.

I decided to test it for myself to see how well it works. I simply went to their website, entered a phone number, and clicked the FreeCall now button, as seen here:


You'll be presented with a captcha code which you must enter to prove you are human, as seen by this clipped browser screenshot here:

Then, you'll see an ad and a short countdown (15s or less) before you can initiate the call as seen by this clipped browser screenshot here:

The countdown was pretty short, so surprisingly it wasn't annoying. After the countdown, the Flash application confirms your microphone source. Simply by talking into it, it detects the audio signal and then initiates the call. The call quality was pretty good - certainly on par with other web-based VoIP offerings.

I should point out that if you don't register, the calls are limited up to two minutes in duration each. Pretty useful if travelling and just want to make a quick free call. By registering for free at CallingAmerica.com, users can make an unlimited number of calls for a duration of up to 15 minutes. All in all CallingAmerica.com is worth keeping bookmarked for when you need to make a quick free call.

While we wait for Digium's official SIP-to-Skype gateway, Nerd Vittles today informed me about his very cool recipe that you can use today to build your own free SIP-to-Skype gateway enabling you to use your SIP-based desktop phones connected to Asterisk to make Skype inbound/outbound calls.

Part of the recipe uses SipToSis - SIP to Skype Gateway Bridge Proxy. SipToSis is a piece of software which Nerd Vittles points out "forms the lynchpin of Gizmo's offering and which lets any Asterisk user create much the same gateway at no cost other than the expense of any Skype Out calls you may choose to make."

Nerd Vittles explains in his tutorial:

When we're finished, you'll be able to call any Skype user in the world from any extension on your Asterisk server by entering either a Skype username or any 10-digit telephone number preceded by an 8 to take advantage of SkypeOut calling rates. You'll also be able to receive incoming calls from any Skype user on any extension of your Asterisk system. In short, what you get is a transparent interface to several hundred million Skype users from your Asterisk server.

In summary, with this tutorial you'll be able to dial Skype users, as well as receive incoming calls from any Skype user! Nerd Vittles' recipe should work on just about any Asterisk-based system. I might have to try this recipe myself later on today. Good stuff!

Skype and Nokia today announced that Skype will be integrated into Nokia devices, starting with the Nokia Nseries. The Nokia N97 flagship phone will be the first to incorporate Skype in the 3rd quarter of 2009.

Skype will be integrated into the address book of the Nokia N97, allowing you to see when Skype contacts are online and perform instant messaging (IM) or VoIP calls.

But here's the real kicker - the Nokia N97 will be able to use Wi-Fi and 3G to make and receive free Skype-to-Skype voice calls as well as Skype calls to landlines and mobile devices. The Apple iPhone on the other hand, restricts VoIP clients to just Wi-Fi VoIP calls and blocks 3G (data) VoIP calls.