VoIP Now

Voicemail-to-Text Application For Skype

SimulScribe's voice transcription services, which are already available for US mobile phone users, are now available for Skype users. SimulScribe is currently offering a one week free trial. Subscribers to the service will have their Skype voicemail messages transcribed using proprietary algorithms, then delivered via email or SMS text messages.

I'm wondering whether a service like this can last for long. While speech to text applications still have a ways to go, with more powerful computers coming, such ability could become native on a computer. In which case SimulScribe would become unecessary.

For that reason, I hope that their game plan is to develop an Extra for Skype, but not release it immediately. There's no reason they cannot charge for an Extra/ addon, like HotRecorder does, provided their application does far more than a similar freebie.

VoIP Roundup - Mon Feb 26, 2007

Apple TV Delayed
The shipping of the Apple iTV set-top box is being delayed until mid-March. The IPTV device allows you to wirelessly stream video content from your PC or Mac to a TV screen. I can hear TV junkie geeks everywhere going "damn!"

Some Communications Purchases
Two recent telecom purchases include Broadview Networks Holdings Inc.'s buyout of New York-based InfoHighway Communications Corp and phone maker Ericsson's plans to buyout Tandberg Television.

The Broadview combined company will serve 80,000 SMBs. Ericsson's planned purchase is in hopes of expanding their presence in the IPTV market.

How To Profit From VoIP
SmartBiz has an article with the enticing title How to profit from VoIP, which is actually an overview of Unified Messaging and presence. The crucial point of the article: it can't be done with the traditional PSTN system.

New Skype Integrations: Mindmeister

Skype is appearing in more and more web applications. The latest is Mindmeister, and web-based mind mapping tool that goes one step further than bubbl.us. Mindmeister not only allows real-time shared editing with collaborators, but you can use Skype click-to-call buttons to chat with each other during the edit process. [You can read a more focused review at Mashable.]

As a long-time, hard-core mind mapper myself, it's great that VoIP is enabling new forms of collaboration such as this. For example, there are a great many opportunities for VoIP in e-learning, and mind maps are a more intuitive way to brainstorm, teach or tutor. What I'd really like to see mindmapping combined with Learn Without Limits/ Tutors Without Limits, which does use Skype.

Now that said, standalone mind mapping packages like MindJet MindManager and Mindapp are considerably more robust than Mindmeister. And both allow publishing to a hosted web page for sharing with others. However, apps like Mindmeister allow real-time collaboration. It'll be interesting to see if a company such as MindJet who have a fairly mature tool, start offering competitve web-based collaboration tools. In other words, a convergence of functionality would be very, very nice.

What I'm really wondering, though, is when there will be similar SightSpeed integrations. They really need to open up their API.

Seinfeld's Kramer Predicted The Future Of Voice?

Didn't catch the title but in one episode of the TV sitcom Seinfeld, crazy Kramer predicts the future (2000), saying that we'll all be on permanent speed dial, and that calls will just come into our brains. This was in response to Jerry's new fling (Lauren Graham of the Gilmore Girls) putting him on her speed dial at #7, after two dates, then moving him to #9. Meanwhile, her stepmother comes after Jerry, ready to defend her #1 spot against Jerry.

Well, I'm happy to report that VoIP means never having to be a position on a speed dial, at least in soft clients. Everyone in your buddy list is #1. Unfortunately, brain-based presence features just aren't here yet, 8 years after Kramer's prediction. I guess we're waiting on a skull USB port first.

Switching To VoIP

Ted Wallingford's Switching to VoIP book from O'Reilly first came out in Jun 2005, but I noticed it being advertised on his site recently, while catching up on my reading. Now, analysts have declared at the end of 2006 that VoIP has now gone mainstream. So Ted's book might be a good one to pick up. Ted is also the author of the more recent VoIP Hacks.

I'm predicting that we'll see more and more VoIP books appearing on the market, several probably in the edit process already. But the good news for publishers and authors is that the lifecycle of VoIP books is probably going to be longer than a lot of computer-related books. My PHP + mySQL web programming book (designed by me but only co-authored) came out in Nov 2002 and was out of print by the next year because it was outdated. As long as VoIP books focus on features and hacks and are supplemented with a blog for updates, they might just stay relevant for an extra year.

The extra market that'll appear for VoIP books is in education, as more programs appear for training people in the high end of VoIP skills for niches such as IP PBX, installation, performance monitoring, load balancing, security, etc.

Video Campaigns: Can You Smell What Barack Is Cooking?

Senator Barack Hussein Obama must have the right-wing TV show hosts running scared if they're already taking xenophobic swings at his unusual name, despite his having been born and raised in the United States as a patriotic American - unlike California Governor Arnold Terminator, whom some people want to rewrite US laws for, to get him into a presidential race - shudder to think. But Obama has made a smart move: embracing web video for his campaign.

While I have a different preference for the next US Prez, I sent Senator Obama's campaign an email suggesting they follow Peter Csathy's wise recent advice about video blogging and video politicking. This was a few days ago, before I knew that the Senator is working with Brightcove on a channel. Apparently this was just before Brightcove pulled
in nearly US$60M in next-round VC funding?

At any rate, I'm paraphrasing what Peter has said: the next President will utilize Internet video better than everyone other candidate. Now if an IPTV/ video streaming company got smart, they'd create a special campaign channel and show paid content from all candidates. Just my feeling, but they could pull a great deal of web traffic and pay for it with advertising. Teaming up with Google on their Google Video or YouTube sites is one option.

Then again, it may not be necessary, as Senator Hilary Clinton, too, has just embraced online video chats. Well, let's hope that they all follow my video calling etiquette, as I'm sure no one wants to see the next President via video in their undies.

The All-New Ford/ Microsoft Car: Now With VoIP + Conferencing

Bwah ha ha ha ha. Just imagine it now: in the near future, all over North America, you'll see drivers talking to themselves.

What they'll really be doing is talking to Sync, the new automobile operating system from Microsoft, to become available in a number of Ford, Lincoln, and Mercury models starting in the 2008 model year. Like drivers don't have enough distractions such as mobile phones, now they can talk to a Ford car courtesy of Microsoft? What will Sync do when someone has a case of the road rage?

Couldn't Ford have come up with a better business partner? [NYTimes free reg needed] Like working with a company with operating system software that actually works? Microsoft's the company who in 2006 had 284 unsafe web browser days due to flaws through which malware attacks could be made (and they only issue software patches once a month). Do you really want to be driving a car that runs an operating system created by Microsoft? I'd considered a Ford for a future vehicle, but this move ensures I will never, ever buy a Ford.

Surely Stock Options Steve would have entertained the idea of an iCar. Or one of the Linuxes maybe? Ford Ubuntu. I like the sound of that: able to go where no vehicle has gone, and now with cron tables. You can grep your car. But Ford, being a fiercely loyal American car company, probably thinks Linux is for communists.

Let's just hope that when you have to inevitably reboot Sync, the car's engine doesn't shut off and restart. Unless you're already stuck in traffic Though the full Sync feature list is pretty impressive. Everything is voice-activated and Bluetooth-enabled, hence my quip about talking to yourself. There's VoIP, call waiting, call conferencing, a push-to-talk button in the steering wheel, and transfer of conversations from phone to car. But my feeling is that if a car company builds in conferencing abilities into their vehicles, the country has a serious workaholism problem. Can't wait until the first virus, worm or Trojan hits Sync. Color me unimpressed.

Network Physics VoIP Quality Monitors

VoIP sys admins will have another potential tool in their arsenal with new VoIP quality monitors
from Network Physics. The offering, called NetSensory Solution Insight for VoIP, works as an extension set for Network Physics' appliances. These extensions measure over 60 metrics related to VoIP call quality.

As I've pointed out before, there are many factors that affect VoIP call service, but I wouldn't have thought there were even 60 IP metrics, let alone that many that affect call quality. Things I haven't touched on before, which Network Physic's solution does, includes using the appropriate CODEC (Coder-decoder) algorithm. Essentially, there are different algorithms to compress and decompress digital audio data, and some perform on the fly better than others, depending on issues related to both network and computing resources.

The Indian Bluetooth Gambit: Or How To Cheat At Chess

Grandmaster Bobby Fischer caused a ruckus in the 1970s. when he he denounced the United States - where he grew up - and made pointed political comments. More recently, he renounced his US citizenship to avoid deportation to the US and a 10-year jail sentence. He also spoke in Iceland about President Bush's "regime". By comparison, Umakant Sharma, an Indian chess player, might be considered less trouble, merely cheating at chess using a Bluetooth device stitched into his cap. Now, this isn't the Bluetooth ski cap Motorola offers, but this certainly would be one unexpected way to use it. It's not like it's hard to configure Bluetooth headsets.

His accomplices would run chess simulations on a computer and relayed info to him. Sharma has been banned for 10 years. Maybe he can join Fischer on the fugitive lecture circuit.

Dream Phones, Voice2.0, Voice3.0

Luca Filigheddu describes "voice 3.0" for the future, and lists his dream mobile phone, inspired by an engaging post by Ken Camp. The phone would have several VoIP-related services, multiple
identities, video, configuration of availability, remotely stored contact lists, rich presence information and more. Sounds good to me. Sign me up. Except we're going to have to wait a few years for this before more than just us VoIP bloggers are using the term "voice 3.0".

For the present, I've listed my own VoIPmas wishes. Basically, I won't be happy with anything short of a Buck Rogers cellular wristwatch with WiFi for VoIP, Bluetooth (or Wibree), and a HUD (heads-up display) for video in. Not sure about video out, though the wrist watch should suffice. And so exactly what should we expect for voice 4.0? Portable Star Trek-like holodecks?

Microsoft Hearts VoIP?

You'd expect a software giant like Microsoft to have already been playing a big role in the IP communications market. But beyond the high number of users (active + inactive) of MSN/ Windows Live Messenger VoIM clients, it's hard to say they have any great involvement in VoIP and related services. They have announced this or that sort of IP communcations-related project in the recent past, but seemingly done little to compete with the dominant forces in the market (beyond launching Windows Live Messenger, which is okay, but not as good as Yahoo Messenger). But CEO Steve Ballmer indicates that they are going to change that with their new Vista operating system, come January 2007.

The Vista OS - which someone touted as not requiring anti-virus software - will integrate a variety of IP-related offerings including VoIP, IM (Instant Messaging), and video-conferencing in desktop and server applications. There will also be Microsoft PBX. * Yawn * I'm not buying it, sorry. They're late to the game (though Garrett Smith thinks otherwise) and can't offer anything new to the individual. However, the Microsoft brand name may encourage enterprise to get more involved in IP communications - so that could be a big sell for them. (In which case I do agree with Garrett.)

There was no word about whether Ballmer raged across the stage screaming like a lunatic while saying how much he loved "this company". This is your brain on Microsoft.

Think this will help your company's share price? Hmm. Could be. Dance little Ballmer, dance.

Why Switch To Asterisk IP PBX?

John Edwards lists 5 good reasons for switching to Asterisk for an IP PBX, and 9-step guide to getting started with Asterisk. Asterisk is of course open source software (OSS) so the full source code is available.

Open source IP telephony is getting a push by companies like Fonality, Digium (makers of Asterisk) and Polycom in terms of new partnerships, hardware and software. While I'm a big fan of open source, history shows a reluctance on the part of middle managers in larger enterprises to accept such software. I'm really hoping that this isn't the case with IP telephony, because such options afford robust, customizable solutions that have as much power as more expensive offerings.

If OSS IP telephony companies are smart, they'll go after the SMBs and not worry about enterprise, who'll want more "name brand" solutions from companies like Cisco or even Microsoft, who have vowed to integrate IP communcations into their new Vista operating system in Jan 2007. My feeling is that there are enough SMBs out there that would benefit OSS IP telephony, and would welcome such an option as well as the price. But since the cost and availability of "experts" who can add modules or debug problems may be a factor, solutions with a  set of plug-and-play SMB/ CRM modules will likely capture the most interest.

100 Projects For TrixBox/ Asterisk IP PBX

Nerd Vittles always has great information regarding open source IP PBXes, in particular TrixBox and Asterisk. Now Ward Mundy has put together a list of 100 projects relating to both, some on the Nerd Vittles site, and some elsewhere. (The links are not all projects; some are links to IP communications- related websites.) Categories are:

• Installing a free TrixBox server
• Installing the free Asterisk@Home PBX
• Customizing Asterisk@Home
• Asterisk server hardware
• Additional Asterisk hardware and software
• VoIP provider reviews and configuration tips for Asterisk@Home
• Securing Asterisk
• Additional Asterisk applications
• Where to turn when you need some help

Probably everything a DIY (do-it-yourself) VoIPr can think of. Great job, Ward. I wish I had the time to try some of these. They look like a heck of a lot of fun.

All I Want For VoIPmas Is SuperSkype

Skype just released their 3.0 Beta for Windows. Wow. From 2.5 to 3.0 in only a couple of months. What's this one got that's new? Supposedly Skypecasts and Public Chats, which I thought were already part of Skype. Click-to-call from websites through Skype browser extensions (for IE and Firefox web browsers). Again, this functionality already exists for Skype through other means. A more powerful user interface: new tabbing arrangements, wall paper. Skype Extras, for playing games, sharing desktops, and a whiteboard for notes during Skype calls. (Extras aren't new, but they have been integrated more tightly with the interface.) Enterprise compatibility? This should be interesting.

But to clarify, the Skypecasts are now browsable. Search for a Skypecast then join in. The Public Chat host can moderate conversation and reject or ban users. Public chats are promotable as links in email, web pages or Skype mood messages.

What'd be really cool, though, is built-in recording, with aim to satisfy podcasters. That means searchable meta tags on recordings (a la HotRecorder). Plus the ability to post the podcast to a weblog on the all of the popular blog platforms. There is Skype Recorder v1.2, which is free and activates automatically, but it doesn't have those extra features that would be great for podcasters.

Suggestion to eBay/Skype: buy HotRecorder and hire a team to do weblog posting integration work as well. At that point, I would actually pay for the Skype soft client (provided I can use Paypal, your sister company, to do so).

So, St. Niklas (Zennstrom), all I want for Xmas is a super Skype that does all of the above. Oh yeah. And the ability to mobile Skype from my PalmOS-based Palm Treo 650 running on CDMA EV-DO cellular wireless Internet access. I'm not asking much, am I?

Recording Multi-Person Conversations For Podcasting

Nick Wilson over at Performancing.com asks about "the quickest, cheapest way to do high quality multiple stream podcasts." Meaning, in this case, recording a VoIP conference call and publishing it to a website as a podcast.

There are hosted services and even some that let you initiate a conference call from their website. For example, Gabcast offers free podcast creation and hosting, but there are some limitations. There are toll-free numbers for the US, but in other places, you might have to pay for a long-distance call.

But in my opinion, the quickest, cheapest way to pull off a multi-person podcast is, arguably, with Skype. For a free solution, a Skype conference call will work fine provided a few factors are satisfied:

1. Everyone is using Skype. I.e., not landlines or mobile phones in use.
2. You shut down any unnecessary apps on each participating computer. (For best call quality.)
3. You all have a broadband connection and are not talking during a busy period locally (late afternoon, mid evening, before midnight).

There are a few other tips for better VoIP call quality.

For recording the calls, there's HotRecorder (HotRecorder.com, US$14.95) which is even geared for podcasters. There's a search function based on the text that you attach to each recording. HotRecorder, if it's running on your PC (Windows XP, 2003, only for now), will automatically start recording when a call comes in on Skype, or you activate it. HotRecorder works on Skype conference calls as well.

For a completely free solution, use the Audacity (audacity.sourceforge.net) audio editor (with support for Cubase VST plugins) to record the call. Audacity has to be manually started, but it's easy to edit your audio track afterwards - something HotRecorder does not feature. Audacity runs on Windows, Mac OS X and GNU/Linux.

Also see VoIPcasting: recording VoIP and Podcasting.

Voice Applications: You'd Better Recognize

One hot voice application space that will be useful in biometrics is voice recognition [Unified Communications]. It's likely less disconcerting to users than, say, fingerprinting, palm vein scans, or facial recognition. This type of app has been around for quite some years, but accurate voice recognition has been waiting expectantly, ready to be called upon - something that's only now happening due to more powerful computers. And there is the potential to use it in mobile phones in the future - at least in my estimation.

To my knowledge, voice biometrics is hypothesized as being accurate - i.e., that human voices are unique enough that they can be used for user authentication purposes in mobile payment, secure access, or other applications. If this is indeed true, or at least sufficient for most authentication purposes, say coupled with a verbally-administered PIN code or password, then all that remains is the horsepower needed for mobile handsets. We live in interesting times.

Nokia Opens US Mobile Apps Research Center

Palo Alto, California, is the home of the new Nokia Research Center. Nokia has a three-agreement with Stanford University to jointly work on research projects for "collaborative mobile computing and applications". The four areas that their research will focus on are:

• Context-aware content and communities.
• Wireless grids.
• Advanced user interfaces and visual media.
• Innovation radio and sensor networks.

Nokia recently bought an RFID company, and with research into wireless grids and sensor networks, it's possible that they will work on crowdsensing applications. In such apps, each mobile phone would have an RFID chip capable of sensing some environmental condition, such as moisture or heat. Each handset would be a node on a wide grid. If such apps are feasible they could revolutionize local/ regional weather reporting, possibly even traffic reporting.

The research center will initially employ 35 researchers, with plans to expand to 100 or more. Nokia recently introduced a new wireless protocol called Wibree, which is a low-power connectivity protocol designed for small objects and possibly mobile phones. Whether Wibree will play a role in the Palo Alto research center is unclear. They have also been planning VoIP on their line of mobile phones for quite some time.

[additional sources: Press.XTVWorld]

Ahead Of The VoIP Call Center Curb

While some call centers are still considering the use of VoIP, others are already on it. CampusUSA Credit Union installed their VoIP call center system five years ago, and did it in a single weekend. [SearchCRM] The key thing to rapid VoIP deployment is having a plan, of course - knowing what quirks to expect. While companies that have no phone system at all have an easier time of a switch over to VoIP, a relatively quick install can be done.

Small companies have an advantage. But larger companies and/or those with existing phone systems can reduce deployment time - whether for a call center or just a business IP telephony system. If you treat a VoIP system install as a software project, then planning and design should be 50-80% of the time spent. Do you know what network capacity you will need? Peak support hours? Backup systems? Call recording requirements? Start with a list of your required functionality and go from there.

VoIP Support Services Market To Grow

New research from IDC suggests  [CertCities] that the VoIP support services market will reach nearly US$1.3B by 2010. The growth will come partly from the use of non-proprietary VoIP software, which a single company will not have a stranglehold on in terms of services.

This would suggest that OSS (open source software) IP PBXes such as Asterisk will have a huge role to play in the coming years. Since the cost of startup is so much lower for IP telephony, there is likely to be a boom in the number of companies offering plug-and-play add-ons for Asterisk and other OSS VoIP solutions, as well as support. And since the support can be conducted using the solutions themselves (voice calling, video calling), support costs are lowered as well. Look into the future, and voice-recognition -activated animated avatars might be handling the support calls.

VoIP For Telecommuters

If you watch Jon Stewart and The Daily Show, you know that the United States hit 300 million people in October. As you might know, The Daily Show is a humorous look at World and US news. But VoIP providers are smiling, not chuckling. As total workforce population grows, daily traffic becomes an increasing nightmare in many cities and even towns. More people are telecommuting for work, or working at home. And VoIP is being promoted towards this particular niche of the residential market: the telecommuter.

Especially happy are the cable companies, who are promoting the benefits of their services over other types of broadband Internet connections. Since telecommuting typically requires an Internet connection throughout the day, the old slow 56K modem over dialup just isn't going to cut it for work. Some companies will pay for broadband installation in their telecommuting employees homes. Then there's the work-from-home types with their own business, who could loosely be classified as telecommuters (for lack of a more accurate term).

Telecommuting in either case requires communication with others, of course, and thus the potential for added long distance calls and costs. VoIP in both voice and video forms can save them a considerable amount of money over regular phone lines and mobile phone services. And cable companies are hoping to get a fair share of converts from dialup.

While other types of VoIP providers might promote themselves as well (they'd be crazy if they didn't), my money's on cable, simply because of the triple play they can offer: TV, Internet, VoIP. A convenient package. On the other hand, don't count out other types of broadband providers. If someone is working at home part or all of the week for their employer (as opposed to running their own business), the broadband service may be paid for.

Consider that in the average North American city, operating costs (office space, electricity, equipment leasing, telecommunications) for each employee is $40-60 per square foot per month, maybe more. Many employers would be happy to spring for broadband access for the occasional/ regular telecommuting employee - especially since their telecom costs are lowered, and there are even free or inexpensive video-conferencing options.

That may mean, for accounting purposes, that the employer wants a separate billing account for the broadband connection and any VoIP services. If an employee already has cable (TV or TV and Internet), they may have to get a second connection for work. Putting in a second cable line may or may not be an option in some areas. Which may mean that other VoIP providers, such as highspeed dialup or satellite types, may still be contenders in the market for VoIP for telecommuters.

[additional sources: Computerworld Networking, Telephony Online]

Video Politicking - Reach-Out Campaigning

Gadget Trail has a list of ten ways to use VoIP that you probably didn't think about [link below]. One of the tips, #4, suggests volunteering with you fave political party, then using Skype to make free outbound calls (to landlines) to registered voters in swing states. What a briliiant way to campaign without spending loads of money. Rock the vote. Save the campaign money instead to pay for the nasty, mudslinging TV commercials. I suppose you could post them on YouTube as well.

Go one step further and use videblogging using SightSpeed (or Skype). Or hold live video interviews that anyone can join in on. Even one-on-one video campaigning could be effective, as Peter Csathy discusses [link below].

The politician that captivates the web-savvy group, and maybe even uses something like Skypecasts or video-conferencing in SightSpeed or IPTVbroadcasts, is a person whose message will get out there, and who has a chance of running for high office. But that person will also be under intense scrutiny, so the best course of action is utilizing a "permission list" to send campaign videos to, if they're precorded. Consider also using "SkypeMe" buttons on a campaign website.

Whatever you do, don't follow step #10 at Gadget Trail to connect the White House with Cuba. Unless you're at a public computer that doesn't require signup of course :)

[sources: Digital Media Update, Gadget Trail via VoIP Telephony Service]

Enterprise: Ways To Marry Skype With IP PBXes

Not too long ago, Pika Technologies announced their bridging solution for Skype and Asterisk, perfect for enterprise use. And there's also VoSKY's solution which combines Skype and an IP PBX, aimed at SMBs. They even have a VoSKY Skype Call Center. And I won't pretend to know exactly what this does, but last week, Instant Solutions released their ChanSkype Skype channel driver for Skype. But from what I gather at O'Reily Emerging Telephony and other sites, it seems like you can use it to run Skype clients off of an Asterisk IP PBX. They tested it off of a Dual Xeon 3.0 Ghz with 6 Gb of memory. Hmmm. Wonder if it might also function as part of a Mac Mini IP PBX, since both Skype and Asterisk can run on one.

Build Your Own IP PBX?

Okay, don't get mislead by that title, but if you saw the slick iBlue IP PBX made from a Mac Mini and don't want to pay 3,000 Euros, Ted at MacVoiP mentions that in his new book VoIP Hacks, there are instructions on how to make your own. In fact, you can even use the open source SIP-based Asterisk IP PBX software. Save even more by using any USB stick; an iPod to boot the system is not necessary. Don't know what a Mac Mini costs, but I'm guessing if you can make your own iBlue-type of IP PBX, it'll cost you far far less than 3,000 Euros. I gotta find me an Apple store somewhere. And a bookstore. And lock myself away to for some VoIP DIY (do-it-yourself) projects.

Skype For Business VoIP?

The jury is still out on this one, as far as I'm concerned, but things are looking up. Skype generally has high call quality, and even polls users after a pc-to-phone call. Then there's Pika Technologies VoIP bridging solution for Skype clients and the Asterisk IP PBX, which holds great potential for business use: inexpensive calls in a great interface, coupled with a great SIP-based IP PBX. Then there's the customer testimonials. According to Jerald Downs, owner of a US-based fruit company

In the past all my business was done by e-mail. Now I use Skype a lot to keep in touch with my growers -- it increases the lucidity and trust between us and it's clearer than any landline.

Well said. Successful business relationships rely a great deal on human interaction. VoIP offers an interaction that supplements face-to-face meetings. And I know from personal experience that high-quality calls seem to engender more trust than when you have to put up with rattle and hum, crackle and pop.

So any VoIP solution that can satisfy the above conditions for business use has an advantage over all others.

[sources: ZD Net Australia]

Tips For Easing Enterprise VoIP Deployment

In the Oct 2006 issue of Networks & Servers, Mary Shacklett provides some valuable tips for easing the deployment of VoIP in the enterprise. Since the majority of businesses have little experience with VoIP installations, integrators and resellers have to be sought out, and each one is not the same as another in their skills. And it's not just about the lowest perceived price of a system.

Choose a person or team that will willingly make you aware of all the aspects of deploying VoIP, walk you through the process, and suggest what is best for your company, even if it means a hybrid VoIP/ PSTN system - a pure VoIP system is not always the best, especially for an established company, who might experience significant downtime if their entire telephony system were to be replaced.

Now That's What I'm Talking About: Custom Voice Mails

Not long ago, I was bellyaching about wanting a way to produce different voicemail messages for different callers (based on caller id). In fact, some other blogger mentioned something about wanting one voicemail message for his girlfriend/ wife (both?), another for business contacts, and yet another for friends and family. Well YouMail lets you do this. Their initial application rollout is for Verizon, Cingular and T-Mobile cellular subscribers only. Currently, there is only Windows support, with Mac coming soon. More details at YouMail. (As I'm not a subscriber of any of the above providers, I can't test it.)

I'm guessing that even if YouMail doesn't get into the VoIP niche, someone else will come up with similar features for soft phones. I mean, it can't be that hard. All soft phones already know who is calling, if the caller is at least on a soft phone. Now since I have not explored VoIP soft phone and VoIM voicemail all that much, I may have just missed the fact that some of them already have customized voicemails. I'm wagering that if Asterisk cannot already do this, that it wouldn't be all that hard to do so.

[sources: MobileCrunch, Technology Evangelist]

Bluetooth File Transfer Capabilities

The Bluetooth SIG (Special Interest Group) has declared October as "Bluetooth Transfer Month". They are promoting the fact that Bluetooth can be used to transfer digital content wirelessly between enabled devices including phones, computers, PDAs and other devices. Any two devices with Bluetooth capability and memory have the ability to transfer files to each other. (Whether they actually can is dependent on whether manufacturers have made the functionality available to users.) [via Wireless IQ]

Sample applications include passing digital business cards between phones and PDAs, capturing TV or stage show information from digital billboards and posters, sharing photos and music, and more.  A stage version of the Lord of the Rings included a promotion where special subway posters allowed people to download ringtones using Bluetooth. Obviously, there could be some very interesting social applications.

To help promote the file transfer abilities of Bluetooth, devices that are capable of this will have an "Experience"  icon on the device and packaging. But with VoIM becoming more common on cell phones, Bluetooth file transer usage might increase without the promotional campaign - if the ability is built into the next generation of VoIM clients, for short-range transfer.

Quantizing Voice Data For VoIP Applications

One of the great benefits about VoIP and IP telephony in terms of business use is that a voice call now becomes data. What that means, amongst other things, is that a VoIP system adminitrator can manage user accounts invidually or in groups. Access can be given to voice-related data - such as call recordings - in the same manner that computer file access can be given. It also means that a group of people can be given access to long-distance calling, file transfer, application sharing, or what have you, with relative ease. While traditional telephony offers some of these group-access features, VoIP telephony makes it fairly easy to implement advanced features without special phone lines or equipment. As well, VoIP calls are treated as a computer resource, so security is easier to implement.

VoIP As A Teaching Aid

VoIP is increasingly being used in a number of ways that traditional telephony never could. One such way is in online tutoring. Using either a VoIP or VoIM soft client, you have access to a host of free (or inexpensive) functionality ideal for remote teaching: text chat, voice chat, file sharing, video calling, conferencing. Some soft clients, such as AIM Pro, also have the ability to do desktop application sharing. There are even a number of options for collecting payment for your time: Ether or Skype + Jyve.

If you plan to tutor online, using VoIP or VoIM software is an ideal way to supplement the learning experience. See more details at 8 reasons to use VoIP and VoIM in teaching.

Protect Kids With IMSafer

VoIP blogger Alec Saunders talks about a new Instant Messaging monitoring tool for parents that has been created by his friend Brandon Watson. Called IMSafer, it would run in the background on a computer, discreetly monitoring IM text conversations and using lexical analysis to determine if the person talking to your child might be a sexual predator. The analysis techniques used are the same used by law enforcement.

I have no children myself, but this is a wonderful idea. It's unfortunate that we need these things, but we do. And with VoIP use becoming more widespread, maybe someone can marry voice-to-text translation with something like KishKish lie detector for Skype and come up with something that can protect people from vishers.

More VoIP Advantages Over Traditional Telephony

The Yankee Group has a report suggesting that soft phones (Skype, Gizmo Project, Sightspeed, Hullo) and IMs (Windows Live Messenger, Yahoo Messenger, Aim Pro, Google Talk) are moving into a new voice applications such as click-to-call, an area that tradtional telcos and cellular carriers just cannot follow into easily.

Click-to-call and other VoIP SOA (service-oriented architecture) offerings are gaining in popularity with service providers. Google is already planning a click-to-call service with eBay. Adobe plans to embed VoIP into Flash media players, which are fairly common on some websites - especially with the rise of embedded streaming video in personal and professinal weblog-style websites.

[sources: VoIP Magazine]

SkyNET: Single Geek Male Finds Single Toll-Free VoIP Number

Well, I didn't find it, exactly. After I posted my Single geek male seeks single toll-free number article, Michael Steverson from SkyNET-tel.com posted a comment saying that they can do what I was asking for right now: a single 800 VoIP number. Do my eyes deceive me? Really?

The deal is US$9.99/month for a Personal 800 Number. That has to be teamed with the One Cent Plan, which is $4.99/mth. Calls are then $0.01/minute. While I haven't been as much of chatty kathy lately, if I were to resume my old talk habits of 800+ minutes per month, well that'd still only be 14.98 + 8.00 per month. My old toll-free number cost me about $35/mth, if I remember correctly. So even if I used 1000 minutes per month, that'd still be just under $25/month. There's also the unlimited plan of $23.99/mth (first month free) or the unlimited business plan of $39.99/m.

Coupled with a personal 800 number, that's not a bad deal at all, if I can find a reasonable VoIP call-in number plan and suitable area code, then I'm set. The 800 number requires a local number, but if I can get a local area code with VoIP when I move to the big city, then I'm good. (That might be a problem, as most popular VoIP services do not cover the city I'm moving to, including SkyNET, from what I can tell.) But the 800 toll-free number is apparently good for 36 international locations. People from all of these locations can call the number as if it were local. Man, am I excited. I can finally enjoy vishing and annoying telemarketing calls from all over the world.

Sounds like a deal. Currently, most of my voice chat minutes are local. I've been taking advantage of Skype's SkypeOut free calling promo in North America, to test quality and generally freak friends and family out with my pc-to-phone calling. On the other hand, I did say I was moving. I would still need a soft phone Call-In number for the new locale. If I find one, basically for not more than what I used to spend only a regional 800 number, I can get pretty much what I was looking for: a single toll-free 800 number, not counting a local number. (SkyNET will have their own soft phone in the future. Just a suggestion, but guys/ gals, base it on SIP, so that it can communicate with users on Gizmo Project, iPhox, and others.)

Incidentals: There's a shipping charge of $25 for the free SkyBOX, which I assume is a VoIP adapter for the broadband connection. They're charging sales tax, even though it's the Internet. Maybe it has to do with where I am. And there's a $19.99 activation fee. Okay, I'll stop being a cheapskate. This still seems like a pretty good deal

I'm listening to Roy Orbison, the man with the soothing golden voice, right now as I write this. So maybe I'm a bit sentimental at the moment, but this might just be the beginning of a beautiful VoIP relationship. Thanks, Michael. The only things that worry me are (1) the secure HTTP server certificate on their website has expired. So I hope they'll fix this before I decide to commit to a serious relationship. And for those of you that don't use credit cards, like myself, they accept payment by Paypal. I'm not moving just yet, but when I do, I'm itching to try this. Although if Skype ever gets real mobile support going, I'll have a grand time combining Skype and SkyNET.

Skype From Mobile: SoonR - Take 2

Song Huang from SoonR responded in detail to my original post about Sooner, as well as a post about soft VoIP for mobile devices. SoonR is an application that lets you not only make Skype calls from your mobile phone or PDA, but it also lets you view your desktop applications. Apparently it can render all kinds of information on your phone including Powerpoint slides, AutoCad and Illustrator drawings, and PDF documents.

Except that I couldn't get it to work, other than being able to view my desktop's folders. I couldn't get a simple text file, nor could I use the Skype feature to phone a friend. Actually, I could, but when SoonR called my cell, I was still on data mode and it went to voicemail. So the friend I was trying to call heard my voice mail instead of me.

According to Song's response to my problems with SoonR on my Palm Treo 650, it appears that I missed a few details. Treos are problematic, especially on EV-DO networks. (At least, I think that's the issue. Palm devices using Microsoft Pocket PC don't have the problem.) SoonR allows you to set a delay so that you can switch from data over to phone mode. I missed that. But then, I missed that setting for a few apps. Delays are how Mino Wireless and EQO Mobile both get around the Treo data network problem. (Though at least Mino's is automatic, and EQO might be as well.) So if you are having similar problems with SoonR on Palm OS-based smartphones/ PDAs, try configuring the delay setting.

Now I guess I have to add SoonR to the growing list of VoIP/ voice apps that I have to try or re-try. But assuming that it will work for me now, with all the features it has, it's an incredibly cool application. At least in theory. While it'd be very nice to have access to apps like Outlook, Powerpoint, Illustrator, and PDF, it's like I said about Cognos announcement about running their business intelligence software on Blackberry devices. Basically, the app may be cool, but all of them suffer from the fact that mobile devices typically have such small viewing screens.

What I'd like to see - although I am a geek - is a HUD (heads-up display) that I can connect to my PDA, and a simple interface - possibly a wired glove (maybe even RFID) to actually interact with the application as simply as possible. This is about the only way I'd care about running complex apps or viewing complex data on the go. (That and a better cellular data plan.) Even my relatively large Palm Treo 650 screen won't cut it for me.

Single Geek Male Seeks Single Toll-Free VoIP Number

The VoIP Girl and others recently cancelled their Vonage account. VG is switching to something else. What VoIP service did she switch to? She hasn't decided yet, but it appears she has some choices, including using a virtual number call-forwarded to her softphone. She wants a local number (to her).

This approach could be interesting, but I'd want a single toll-free VoIP number. A few years back, just after I stopped working for a large telecom, I paid for a personal 800 toll-free number through their cellular division. It just happened that I did a lot of commuting: live in one city, work in two others, meetings for personal projects in others, hang out with friends in yet another, promote bands in still more. All in a single day or week. I didn't want people to have to spend a fortune trying to call me.

At first, I had a local cell phone number, but if I took that phone with me out of town and someone called me from my hometown, then it became a long distance charge for them. Then I got a second cell phone but with an area code in the city I spent most of my work day in. But that didn't quite work either. I then switched to a regional 800 toll-free number and my friends and business contacts were very happy. It only cost me about $35 per month, which beat the $200-300+ that I would have spent calling everyone myself. Except some weird politics developed between two big cellular providers and the 800 numbers on mobile phones option was cancelled in my area.

This was a whlie ago. Now I work almost completely from home, and don't travel much at present. Anyone I know that's geographically separated from me has a computer and I talk to them via IM (Instant Messaging), email, or a VoIP soft phone. Most of the time. But being the nomadic wanderer that I am, I'm planning to move yet again. Anyone I know locally right now would want to call me at my new town via a regular phone, not from a soft phone. (Don't ask. Maybe it's something in the water, but I can't convince anyone I know locally to get a soft phone. They don't mind IMs; soft phones they don't understand, or maybe don't like. Oh the shame.)

That means, to save everyone the long distance charges that would occur, I'd want a single call-in number linked to a softphone. Sure, I'll still have my Palm Treo 650, but it'll have a new local number after I move. I want something that isn't going to cost people money. Sure, Gizmo Project has a free 775 area code-based number, but that doesn't give you toll-free - as far as I can tell from the area 775 FAQ.

What I want is a single VoIP plan with a toll-free number attached to a quality soft-phone. Could be wrong, but I don't think anyone has that yet. This means that I'll probably have to get a landline (haven't had one in over 10 years) with an 800 number and forward it to a VoIP Call-In number. This isn't quite the same as VoIP Girl, since she wants a local number. I don't. I plan to be doing a lot of travelling before the end of this decade, for business and pleasure, and a single (toll-free) VoIP number would be best, for friends, family, and contacts. Of course, if I could get this number for fully-functioning mobile VoIP on my Palm Treo, I'd be even happier, and gladly pay for it. And before the end of the decade, please. Then I wouldn't have to worry about which VoIP soft client everyone was using.

Speaking of toll-free numbers, TipMonkies points to a site called Hardtofind800numbers.com. Speaks for itself, I think.

VoIP Roundup - Fri Sep 29/06

Should Web Traffic Be Prioritized?
Matt Brunk at VoIP Loop considers the types of web-based traffic and makes an argument for why certain types of traffic might need to be prioritized, especially since media convergence is pushing a lot of public services into IP-based access.

Testing Your VoIP And IMS
Ixia has just announced their IxVoice software for testing VoIP and IMS (IP Multimedia Subsystem) protocols. via Light Reading] IMS is a core part of media convergence. That is, offering a variety of media over via Internet Protocol (IP), and communication between networks.

Telepresence Via Video VoIP
Be Here is offering their TotalView "VoIP Collaboration Phone" which gives a full-room view for conference participants. TotalView was announced at DEMOfall 2006 earlier this week. [via VoIPLoop]

Skype File Transfer: Unusal Uses

Have a Skype-certified mobile device but no appropriate USB cable handy? You might be able to transfer files to the device using Skype's file transfer feature. Skype Blogs has a post about a reviewer who managed to transfer music files to a Sony Mylo media player and wireless Skype VoIP phone via file transfer. (Sony didn't send him an USB cable.)

You can also use this feature to transfer files between two of your own computers. Really, that's no different than if you were just sharing files with someone else. I work daily on both my laptop and desktop computers for overlapping purposes and have two different Skype accounts, and two different Google Talk and GMail accounts. (Google Talk recently got the file transfer feature.)

I suppose I could just as well use either of my wired or wireless networks and Windows Explorer for a file transfer, but this is easier because I don't have to expose my directories to sharing, and then unshare them each time. My home network stays secure. BTW, here's a link if you want a quick overview of various Skype features.

Let The Embedding (of VoIP) Begin

There's a lot of buzz the past few days about Adobe working on some VoIP project, possibly to embed VoIP from within a flash video player. Om Malik is credited with breaking the story. Bruce Stewart, Tom Keating, Ken Camp, Phil Wolff, and no doubt others have weighed in on the news.

Maybe it's Friday, when I tend to be jaded and play devil's advocate, but I don't see this as particularly surprising or big news. Isn't this really kind of an extension of click-to-call VoIP? At the least, it's embedded VoIP which, while a hot subniche, is already working from web browsers, Microsoft Outlook, and other programs.

But at least with those programs, there's some semblance of relevance for having VoIP-calling, as a phone number will be part of the information. Even if a Flash video has an extra tag for a phone number, how ubiquitous is Flash anyway? (Despite what the other VoIP bloggers are saying.)

Every web design site I've read in the past year cautions people to go easy on Flash-driven content because it's not indexable in the search engines. And it takes time to load, which drives away visitors on slower connections (not everyone has broadband yet).

Nevertheless, while I may not like that Adobe bought out Macromedia, it's an interesting idea and I wish them luck with the Flash-VoIP features. It could certainly be useful for live help sites that want to add VoIPability. [Note: After I wrote this low-key diatribe, I read Tom's piece, which says the Flash player has already had VoIP capability since early 2002. That's a surprise to me, but I still maintain my jaded opinion.] On the other hand, click-to-call types of embedded VoIP applications just might make soft VoIP clients obsolete.

Online Music Collaboration, VoIP Chatting, and Social Networking: Rype

About two years ago, I was helping a young musician develop some confidence in his guitar-playing abilities. (I had spent several years booking bands for shows and promoting local musicians in the past, so I decided to help this immensely talented young man.) Because we worked conflicting schedules at the time, we oftened chatted using MSN Messenger. in text mode. At that point, I'd forgotten that Messenger had rudimentary VoIP (pc2pc only) capabilities.

When my friend, A, initiated a voice chat, I was impressed. At least for a few seconds, until I realized how crappy call quality was (probably mostly due to my then poor wireless signal). But he pulled out his guitar across town and played for me some of the new songs he'd composed. I reviewed them with him. Despite the quality issue, it was quite a heady experience.

Fast forward a couple of years and VoIP call quality has improved - at least for some soft clients. I lost touch with A, because of his strange work hours, and last I heard, he was a bit disheartened about not being able to collaborate and thus gave up writing new songs. (Unfortunate, because he has the talent to be the next John Mayer or Dave Matthews, his fave.)

His biggest problem was finding people to collaborate with when he was actually at home, on his computer, too tired to go meet with anyone to jam in person. Well, budding musicians will be happy to know about Rype, a desktop application that appears to be the ultimate tool for musical collaboration in the global village.

Rype is from guitar.com, but it's not quite available yet, so what I'm telling you is based on the wee bit of text at the site, and the screenshots. And it really looks impressive. Rype will let you record, edit, and produce music, and has a built-in social network. So I assume that regardless of where you are, you'd be able to find someone awake to collaborate with. And when you do finish a song, you'll be able to sell them on iTunes. Brilliant or what?

This is one of those "killer" apps VoIP, and I can't wait to get my hands on it, even if it costs money. (No indication either way.) And if it's as good as it looks, or maybe even if not, it'll probably spawn a dozen copycats/ competitors. First it was online games using VoIP, now this. What's next?

[Found via Skype Journal, but the actual permalink doesn't work, so I haven't supplied it.]

LumenVox Speech Recognition Engine for Asterisk

Asterisk Business Edition will now be including [Asterisk VoIP News] the Speech Starter Kit and Speech Engine from LumenVox at no extra cost. Or you can get the Kit and Engine for US$245 if you're an Asterisk open source community member. Speech processing solutions can be built over the Speech Starter Kit. (LumenVox also has a number of other voice processing packages, including Speech Tuner and Speech Assistant. Their Speech Engine received a Best of Show award at TMC's Internet Telephony Conference earlier this year.)

This is good news for anyone wanting to build advanced voice data applications for VoIP systems using Asterisk, based on the open source VoIP standard, SIP. In fact, this Speech Engine could spur the development of inexpensive voice-triggered CRM (Customer Relationship Management) applications for SMBs and even SOHOs, not to mention enterprises. And now that Pika Technologies just built a seamless integration package for Skype and Asterisk for enterprise use, I can see some pretty sophisticated VoIP call center solutions being created as well. Call Center/ IVR and CRM in a box anyone?

By the way, you can try out the IVR demo for ordering a pizza or checking the weather. Neither my SkypeOut call nor a call from a cordless phone seemed to register too well, but the "woman" for the pizza demo is pretty darn funny. So since she psychically knows where I live, I should be getting a gigantic pizza in about a half hour.

Online Role Playing Games Add IP Communications

RPGs (Role Playing Games) are a type of online game that involves multiple players online at the same time. MMOGs (Massively Multiplayer Online Games) have been popular for several years and have spawned a whole subculture. One acquaintance of mine would play for up until 30 hours straight when he was out of work. Now, as a baker, he has to get up early and can't play as often. But on his days off, he's back to the mega-sessions, playing up to 15-20 hours straight.

One of things he repeatedly asked me to check on (before I started writing about VoIP) was a way for his clan (forgive me if that's the wrong term) to be able to talk to each other simultaneously without paying a fortune for some company's subscription fee. Now that was last year, before I knew about free VoIP conferencing. But his clanmistress was ultimately happy with her choice. However, their choice was not integrated into the game they were playing - meaning that while playing the RPG, they would have to use a separate web browser window (or tab) to start a conversation using another service.

Enter a new generation of RPGs, with integrated VoIP. A new RPG, Fallen Earth, by Icarus Studios, will have IP communications integrated right into the software. Another company, BigWorld, is producing a new RPG development suite which will have VoIP capabilities built-in. Both are a couple of new customers [Mass High Tech] for Vivox Inc.'s integrated IP communications platform and development software.

While there are a growing number of voice data applications, I believe this is a new direction for VoIP. I'm not otherwise aware of any of the more popular online games having this ability. Though I wouldn't be surprised to see, in a few years, RPGs with video capability and even video avatars, where a person appears as their character, in real-time. And then a whole new generation of sleep-deprived players will be swept in.

Enterprise VoIP: Pika Combines Skype and Asterisk

Big news on the enterprise VoIP front. Pika Technologies, Inc., a Canadian call center services company, has come up with a solution that seamlessly combines Skype and Asterisk. Free plus free equals free. Get more details on Bill Campbell's post at Skype Journal, or the Pika press release.

Wow. It's amazing how flexible Skype seems to be, considering it's not open source like Asterisk. Regardless, anyone with a small business should be salivating at this news. This sounds like the perfect enterprise VoIP soution, both for SMBs (aka SMEs) and large enterprises. And here even Skype was saying last November about how it wasn't suited for enterprise use. Pika (and Asterisk) save the day. Expect to see an aftermarket of solutions and hardware for Pika.

Large-Scale Enterprise VoIP Migrations

As VoIP systems grow in favor with enterprises, the size of projects also seems to increase. Take, for example, a commercial bank in China, the Agricultural Bank of China (ABC).They have over 50,000 branches and plan to consolidate their regional call centers into a single VoIP call center. [ Sci-Tech Today via Asterisk VoIP News]

ABC has a fairly hefty list of requirements, including: switchover to PSTN lines, if the need arises, and no change or upgrade to the existing IP network. Already over 100 offices have completed the switch - in just 30 days. There is no indication in the Sci-Tech article of how much ABC is spending on the project, but with assets of US$250B, it's probably worth it to the bank if the rollout reduces their phone bill and saves money in the long-term.

So initial project costs alone shouldn't always be the determining factor in deciding whether to switch. Return on investment is often far more important. Consider that Virgin Entertainment Group of N. America saved US$700,000/year in long-distance costs after they switched to VoIP. Their cost is estimated at $330K for year 1, and a total of around $1 milion. However, they have plans to utilize the network in ways which will ultimately give them a good return in terms of savings.

SMEs (Small and Medium Enterprises) shouldn't fear these project costs, though, as there are a variety of options for IP telephony systems. As mentioned in other posts on this site, knowing what functionality you intend with an enterprise VoIP system will take you a long way towards determining what type of software and IP phones you'll really need.

More On-The Fly Language Translation

New software designed for laptops, intended for Army and medical personnel in Iraq, translates English-Arabic audio conversations in near real time. The software, called IraqComm, records spoken words, translates them, and plays the translations. The process takes a few seconds. The predecessor to IraqComm was a handheld device called Phraselator. [via Technology Review]

While IraqComm is currently for military evaluation only, it is also intended for a variety of other users. It shows the potential market for automated language translation tools. It certainly would be nice to have something like this for Skype which, to my knowledge, only has something like ULRTMT, that translates text nearly on the fly.

On-The-Fly Language Translation?

I've posed the idea before: how nice it would be for a VoIP data application that provides on-the-fly language translation. Well, there's already such a plugin for Skype in beta format. It's called ULRTMT - Universal Language Real-Time Message Translator. [Mathemagenic via Skype Journal]

Although before you get too excited, it's for text conversations only. Surprisingly though, it handles a whoppingly big list of language conversions. I didn't count, but there's probably close to thirty translations, some of which don't even involve English. And the software supposedly works on both active and archived Skype text chats.

Unfortunately, as Mathemagenic indicated, it takes a bit of effort to install. Follow the instructions carefully and it actually does work. Use Internet Explorer. (I didn't try Firefox browser, because the actual translation window runs in an IE browser window. Don't forget to bookmark the link; there doesn't seem to be a trigger from within Skype.)

I tested the service with English-to-Japanese and English-to-French on archived chats. Then I ran Skype on two different profiles on two computers, with one set to French language (although this isn't necessary for languages that use the Roman alphabet).

Unlike most Skype plugins, the meat of the service runs in a browser window. [Like I mention above, the instructions mention IE Explorer, so that's what I went with.] After refreshing the translation browser window, I selected the active conversation on my desktop - the computer with the so-called English speaking user. Then I typed simple French greetings on my laptop. The translation window immediately showed both my French text and the English version. The desktop's Skype chat window, however, showed the French text as typed from the laptop's Skype session.

Verdict: Unfortunately, the Japanese translation does not use the Romaji letter set, and my knowledge of the other three Japanese letterforms has disintegrated with disuse. So I don't know how accurate the translation is. The French-to-English translation, however, is reasonably accurate, if a bit literal. I assume other translations will undergo the same problem. It's part of the reason why machine translation of a language is generally a last resort if a human translator is unavailable. Still, it's a nice start, so bravo to ZOverLord for a great effort, and to a product that just might one day be the closest thing to an IM Universal Translator. At least in text mode.

Show Me The Money In VoIP - Still More Thoughts

Telesyte reports that Australian PSTN telcos will lose more than US$5 in revenue for every US$1 earned. [via 21Talks] So even if they start offering VoIP services, their overall revenue will go down.

As mentioned in our Show Me The Money... In VoIP and More Thoughts posts, the money seems to be in hardware - both handsets and adapters - and integration. But as Fonality is showing, there may also be some revenue potential in VoIP PBXes.

Fonality is a company that makes Asterisk-based IP-PBX systems. While Asterisk itself is free, Fonality's PBXtra has additional features aimed at enterprise. In fact, this is why they are purportedly tops in the Asterisk PBX market. [via GigaOm]

Overall, though, this is still a young market providing a valuable and essential service. Since the service is mostly free or inexpensive, it's everything else associated with the service that will provide revenue opportunities. But my proverbial money's on VoIP data applications. 

VoIP Roundup - Fri Aug 18/06

Skype has released version 2.1 beta of their client for PocketPC smartphones, which will actually work on either Windows CE or Windows Mobile 5 devices. [via The VoIP Weblog]

The question of how VoIP calls get routed to their proper destination over the Internet depends on several methods, none of which are standardized. Some people think that this hinders adoption of VoIP for enterprise. So a set of protocols called ENUM (tElephone NUmber Mapping) was devised which is tied directly to domain names or IP addresses in really clever, simple way. Read more at Extreme VoIP.

I'm not the only who makes nearly all of my calls via VoIP or a cell phone. Phoneboy does so as well, but uses Gizmo Project whereas I use Skype for the free SkypeOut in Canada and the US. Although the pc2phone  call quality of Skype (and other soft clients) is pretty bad, as Phoneboy points out.

But using Gizmo does have some shortcomings, too. Go have a read (it's short) about how he got around a not being able to mute his handset during an 800 number-based conference call.

Examples of VoIP Data Applications

Martin Geddes talks about an telephony industry mag called Receiver (sponsored by Vodafone). In his write up, he speculates on the idea of your voicemail being able to distinguish who a call is from, giving different people a different message.

Of course, if you've followed any of the recent posts here about data applications being where VoIP could really shine, you know that it's more than possible - probably already available. (I'm still looking).

In fact, since VoIP-based CRM (Customer Relationship Management) software can presumably retrieve customer records based on who is calling, I can't see why Martin's idea can't be implemented. That means you could have a single VoIP call-in number, usable on a Wi-Fi phone, to conduct all your conversations, business or pleasure.

Of course, in the scenario Martin was talking about, nightclubbing, you'd need widespread Municipal Wi-Fi if VoIP was to be in the equation.

AppCritical VoIP Assessment Tool For SMBs

A new troubleshooting tool from Apparent Networks will help assess VoIP network problems prior to deployment. AppCritical already exists, but a new version aimed at SMBs (Small and Medium Business). [via eWeek]'

The tool is said to have a low-startup curve and requires little training. But at US$40,000, I can't see a lot of SMBs - especially those falling into the "S" category - being able to afford this. What I do see happening is for VoIP solutions integrators/ consultants purchasing the tool and hiring themselves out. Less headache and cost for SMBs.

InnovAlarm VoIP-based Alarm System

It's always nice to see VoIP being used in unique new ways, and that's exactly what InnovAlarm is doing. Imagine home and security alarm systems, but which use Skype or another soft client instead of regular phone lines. The service is in pre-beta. [via Read/Write Web]

The only drawback with this application is that your computer has to be turned on. I'm wondering if there's a market for a similar solution using phone2phone with a VoIP bridge, using hardware such as Digifone's plug'n'play adapter. Phone2phone VoIP calls generally seem to have better quality.

There's obviously a perception that there is a market for InnovAlarm's method. In fact, Read/Write Web reports that the company will be getting $10 M of venture cap in Q4 2006.

Maximizing VoIP Functionality For Your Home

Thinking about adding VoIP service at home? Here are a few tips and options for maximizing the value of your setup.

Plan on keeping your regular phone line for the time being. While some companies are developing solutions for e-911 emergency calling, most providers don't offer this. If you have children or elder family members, I suggest you keep your current line, or maybe a cell phone.

If you already have a broadband Internet connection, you don't need to get your VoIP through pure play providers such as Vonage. If you still want to be able to take regular phone calls, try one of the plug'n'play adapters that are popping up. They let you use your existing handsets and come with VoIP service, usually by the minute. Most of these VoIP adapters have a bypass feature which allows you to take/ make regular phone calls as well.

Alternately, you can set up a wireless router and purchase Wi-Fi VoIP phones, which can then be used pretty much all over your house, and possibly even outside, within range. Since this setup makes use of soft VoIP clients, the only way that people can call you from a PSTN or mobile phone is if you have a call-in number and service such as SkypeIn or Gizmo Call-In, or something similar.

Evidence suggests that phone2phone calls using a VoIP bridge tend to be of higher quality than pc2phone or phone2pc calls. So keep that in mind when deciding what type of setup to go with, and consider ways to improve call quality for pc2pc and pc2phone.

VoIP Tips: Phase In Telephony Changes

Planning to move from POTS/ PSTN to a VoIP system? Howard Berkowitz says that the move can be incremental, and in fact recommends that approach rather than a wholesale change. Incremental changes, he suggests, reduce the chances of technical problems that come from installing a complete VoIP system. [via Techworld]

One of his key pieces of advice is that any size business that switches to VoIP should also keep one regular PSTN line or mobile phone. That's exactly what I do. I make as many calls as I can using VoIP, but currently keep my PDA phone for inbound calls for anyone who does not use any of the multitude of VoIP soft clients that I use.

Good planning of your move to IP telephony will reduce the problems that are some times inevitable for a new VoIP system implementation.

VoIP System Implementation Tips

Not everyone who has switched their business to VoIP is happy with their results. A Detroit-based law firm switched their telephony a couple of years ago, but has had regular system problems, including crashing. The VoIP system was provided by a client of the firm.

The firm spent US$750K on their six-office VoIP project for a couple hundred lawyers, and had considered ditching it because of all the system problems. However, a software services firm, Compuware Vantage, helped them solve many of the problems. Compuware's management tool reduced support calls from lawyers by 50/ day down to five/ day. The law firm's additional expenditure was just under $100K. [via Computer World]

Project management practices often tell you to essentially not throw good money after bad. In this case, the extra expense was worth it, to make the initial investment bear fruit.

These problems bring some key issues that businesses considering a VoIP system should consider:

Firstly, plan to run a VoIP system on a dedicated computer server. In fact, you may need more than one server. (See steps 2 + 3.)

Secondly, make sure that you run network diagnostic tools to analyze and report on peak network times. Any server worth its salt, whether for VoIP or just a website or database, has to be able to handle peak traffic, not just average performance.

Thirdly, if your company's business is phone-based, you're probably going to need backup VoIP servers, where overflow calls get shunted at peak times. This a technique that high-volume websites, including search engines, use. Unless you are running a call center, you will not need dozens of VoIP servers, but you may need a few.

This sort of information is something any good VoIP system provider/ reseller/ consultant will tell you, but knowing this makes you more aware of what potential problems your IP telephony network might encounter. More knowledge means you're less likely to be cheated or run into problems later.

VoIP Roundup - Thur Aug 10/06

Successful personal development blogger Steve Pavlina wrote recently, in an article detailing 10 reasons why it's worth learning some technical abilities, that he disconnected his entire house from traditional phone lines and switched fully to VoIP. [via Steve Pavlina] He does not say anything about e-911 emergency calling nor the service he's using.

Riverside, California is initiating a pilot project for muni Wi-Fi. It's also being touted as a public safety network. [via Xchange Mag]

Got a GSM-based cell phone? The new CelluNet gateway allows mobile- to- mobile VoIP calls on a GSM network, via a SIP bridge, which should produce a cost savings forproviders of GSM. [via Asterisk VoIP News]

If you're an AOL subscriber, you may be pleased to hear that their parent company, Time Warner, is changing their fee structure to provide email, IM (Instant Messaging) and VoIP free of charge. But only to broadband users. So if you're on their outrageously priced dialup plan, it's time to quit and move up to broadband. [via Teleclick, CNBC TV]

KishKish Skype VoIP Lie Detector Test

KishKish has a new feature for Skype called SAM, which effectively functions like a lie detector. Or so they claim. Voice Stress Analysis is the principle on which lie detectors work. SAM can do this for VoIP calls recorded from Skype.

SAM was orginally just a voice answering machine for Skype. If you're away, it'll record the call and notify you with a list of messages, as well as access to the recording. Now it also detects voice levels on recorded Skype calls, to help determine if the person is potentially lying. [via Skype Blogs]

On their webpage, they have a video of President Clinton talking about the allegations levelled at him re Monica Skankinksi.. uh Lewinski. While the video is playing, a little graph shows P-Willy's voice level fluctuating, synced to his facial and hand gestures. Yet I saw no stress in Clinton's face nor heard any in his voice, despite what SAM suggests. They have a "Skype Me" button to a profile named "clintondenial". If you've downloaded and installed SAM, you can record the call and try the VSA feature yourself. (There's a 10-day free trial, the installation's simple, and SAM is very easy to use, as is the VSA feature.)

Keep in mind that lie detector tests are often disallowed in court in the US. Still, there are a few other presidents and prime ministers I'd like to VoIP and record when KishKish comes out with their real-time version of SAM.

How To Record VoIP Calls - Reader Q+A

There are many reasons to record VoIP calls, especially in a business setting. But even for home use, it can come in handy. (Just have the courtesy to notify the person you're talking to that you are recording the call. In fact, in some countries, recording a regular telephone call without the other person's consent is illegal. Unless you're the government.)

On a previous post about recording VoIP calls, one reader, Richard, asked how he can record his calls using 3rd party audio recording software:

I have been reading your site about how to record VOIP calls. I have Nero Wave Editor and have tried recording. However, the speaker is a fair way from the microphone and I cannot hear the other person when I play back the recording. Would I have to place the speaker close to the microphone or is there another way where the recording can be done perhaps internally through the sound card. If so, would Nero Wave Editor enable me to do this or would I need something else?

Richard, you don't say whether you are using a softVoIP client, such as Skype, Google Talk, Gizmo Project, etc., or if you are using hard VoIP through some PBX device. Let's discuss both scenarios. Regardless of your setup, you want to combine the audio of both people at the same volume.

Recording From Soft VoIP Clients
If you're using something like Skype, there are 3rd-party plugins and overlays. I'm using HotRecorder. With most other softVoIP clients, such as SightSpeed and Yahoo! Messenger, etc., audio recording is built-in. You just have to activate it. So I'm assuming that if you are using a soft client, you do not have built-in recording. In this case, you'll need to employ an external mixer.

First, I don't recommend placing the speaker near your mic. If you do, you're likely to get screeching feedback. Instead, you'll have to send the audio output of your computer to an audio mixer. (I use inexpensive, good-quality Behringer mixers, but they're popular and sometimes hard to find.)

You don't need multiple channels or anything fancy. You're simply going to reroute the audio of your conversation by sending it out of your computer, to the mixer, and back into your computer's audio input, and thus into your recording software - in this case, Nero Wave Editor. I haven't used Nero, but I'm assuming that you will have to manually trigger it, when you start a conversation.

Recording From Pure-Play VoIP Phones
If you're using a regular handset and have VoIP via a service like Vonage, or are using a VoIP PBX, etc., this is a bit more difficult to answer. As I said in previous post on recording, there are special solutions. Otherwise, it depends on the specific phone you are using, but you might be able to output the audio of the conversation from the handset straight to your computer's audio input.

As before, you'd have to trigger your recording software manually. Unless your phone has a MIDI (Musical Instrument Device Interface) port, in which case you'd have to have a sound card on your computer with a MIDI port as well. This is a very unlikely situation. I haven't heard of telephone handsets with MIDI ports because they serve no existing need of musicians and composers. But in case they exist and you find one, the MIDI signal from the phone would trigger your recording software - provided it has MIDI sync capabilities.

But generally speaking, whether using soft or hard VoIP, you basically want to route the conversation's audio directly to your recording software, and this may require a multiplexer or a simple channel mixer as an intermediary device. If you're recording calls for podcasts and want to mix in other sounds, you're better off doing after-call sound editing.

In either case (soft or hard VoIP), you'll have to do a bit of planning to determine the most efficient way to record your calls.

Aside: For general audio recording needs, I use a variety of software. But for the price, you can't beat the free, robust, open source, high-quality Audacity audio recording software. It's supposedly written by industry insiders for garage/ basement/ bedroom musician, but can be used for any audio recording - up to 16 channels simultaneously, if your computer's RAM can handle it. It accepts Steinberg Cubase's VST plugins. (Cubase is a high-end music composing/ sequencing software package.) There's also a built-in programming language, Nyquist, in Audacity, with which you can write your own audio effects. Audacity runs on Windows, Mac OS X, and GNU/ Linux.

Advanced VoIP Apps For Enterprise, SOHOs and SMBs

According to CIO Today, VoIP adoption is getting a boost through advanced features such as broadcasting, presence, find-me/ follow-me, and conferencing. But the real promise of VoIP, they say, is in the integration of voice and data applications.

VoIP also gives advanced CRM (Customer Relationship Management) power to SMBs (Small to Medium Businesses) and even SOHO (Small Office/ Home Office) owners. [via TechNews World]

By treating VoIP as a data application, some incredibly sophisticated CRM experiences can be produced. Throw in auto-answer attendants and CCXML and VoiceXML support, and you have a powerful voice-driven VoIP-based IVR (Interactive Voice Response).

You can also take advantge of VoIP data. For example, if you're an SMB that sells services or products over the phone, each VoIP call becomes data you can store: which country or city the call came from, how much and what they purchased, etc. All of this data can be quantized, stored, and then geo-analyzed. For larger businesses that have multi-language operators, you can transfer calls to the correct operator by applying language defaults based on IP addresses.

It's not that you cannot do this via existing phone systems and computers, but it's not integrated, and thus requires more human data entry. With VoIP data applications, most of the process becomes automated and thus less prone to error. Imagine getting a monthly report, via, email, from your VoIP system showing sales by region. This is just a glimpse into the potential of advanced VoIP data applications.

VoIPing For Profit - Jyve Talking

Like Ether, Jyve is an Internet-connected voice commerce application that lets you consult via phone calls and earn money. Unlike Ether, Jyve is directly plugged into VoIP. In fact, it's an application layer over top of the Skype VoIP IM client.

Jyve's another great idea, like Ether, but they've gone a step further by creating a free searchable , structured directory of "experts" who will consult with you via Skype at a given rate. You can search for listed experts by categories and sub-categories, or by tags. (Ether has a community forum, but no consultant listings that I could see.)

If you're an expert yourself, you can signup, setup, and list yourself on their site, under a variety of categories. They create a "Click & Buy" billing account for you. You can then download Skype and Jyve buttons to post on your website. These buttons display your availability. (Jyve availability can be configured differently than Skype availability.)

Once you get a Skype call from a potential client, you generally spend a few minutes negotiating a price, then use Jyve to switch the free call to a paid call. I think that this feature alone makes Jyve a potentially better service than Ether, since the latter requires two separate numbers to pull this switch off. It's seamless in Jyve - or at least in theory. If you're unavailable for consulting, clients can leave you a voicemail or an email by clicking on the appropriate button on your Jyve listing page.

Jyve-Skype calls can be pre-paid, or metered by the minute, or in blocks of time, etc. You can also sell digital content via Skype's file transfer feature. For example, you may want to record conversations and supply a copy to your clients at a later time. (To send free screen snaps, use TechSmith's free Skype profile for SnagIt. You can also run live screen sharing using WebDialogs Unyte's free Skype plugin. I'm just not sure you can meter either of these for profit. Warning: to use SnagIt for Skype, you must already have SnagIt installed. If not, install the Snagit 30-day demo first, then the Skype profiles version. If you've previously tried the demo and passed the trial date, you're probably out of luck.)

The major drawback to Jyve is that it's purely web- and Skype-based. The calling party also has to be using Skype. So you cannot take calls from a regular phone/ mobile through Jyve. Thus the smart consultant will set up both Ether and Jyve accounts. And like Ether, you're not limited to just talk-only consulting. There are all kinds of professions, including writing, listed in the Jyve experts directory.

So what does Jyve get out of this? They take a 20%, which is higher than Ether's 15%. But in any commission-based industry, 15-20% is pretty standard, unless you're Elvis and your agent is Colonel Tom Parker - in which I case Tom gets 60%. Hmmm. Gives me an idea: Elvis-By-Skype. Need an Elvis to perform for an event? Hook up your speakers to your computer and Jyve-Skype me. Though I think I'd much rather perform Led Zeppelin's Communication Breakdown, or maybe Blondie's Call Me (in Spanish?). Though I'll throw in Electric Light Orchestra's Telephone Line. Hint: voice commerce can be used in a lot of ways, particularly for musicians to communicate with fans, besides corny references to musical communication.

VoIPing For Profit - Ether Consulting

Ether is a voice-based service, though not necessarily VoIP-based, that lets you essentially set up a consulting business online, with the help of a phone, email address and website (free-hosted is fine). I'd all but forgotten about Ether until I stumbled across Amit Agarwal's post a couple of nights ago.

Ether is a brilliant concept. They give you a free toll-free number (and personal extension) that clients can call, which you advertise on your website, email, or business card, along with your rates and availability. At the Ether site, you can login and configure your availability throughout a single day. Calling clients will be notified that you are unavailable at present, if necessary.

If a client want to talk to you, they pay upfront, with their credit card, through Ether's billing system, and the call gets transferred to your desired phone number (home, cell, etc.), if you're configured as being available. If you've set a fixed time limit for a call, the call will end.

Your rates can be set by a variety of time periods, including custom (max $1,000 for a max of 120 minutes). You can even specify that minutes are free after a certain duration. So, for example, I could charge for the first 45 minutes, then allow the rest of a call to be free. (Although there's no way that I saw when I signed up for the beta where you could limit the free time. That's something that would have to be managed manually.) If you've set recurring rates, such as $30 for every 15 minutes, the client will be billed before the call can continue.

It appears that you can setup multiple phone profiles from a single Ether account. So if you do a variety of consulting work and have different websites to promote that work, you can post a different Ether extension # and call rate on each site.

Ether went live near the end of June 2006. I signed up months ago during the beta trial. Because of technical and personal reasons, I never got around to actually fully setting up my account. However, I did come across a couple of websites where the owners had set up. One site owner had two profiles/ numbers. One was something like $100/hour consulting. The other was 30 minutes free, available for a couple of times each week, first-come-first-served.

It's a great concept, and I had intended to set up for business. In fact, I even bought my Palm Treo 650, and the calling and wireless data plans, with Ether consulting explicitly in mind. Unfortunately, since I don't have a landline (haven't for nearly 12 years now), that means I have to use up my costly cell phone minutes. Either that or I need to purchase a SkypeIn, TalqIn, or Gizmo Call In type of plan.

So while Ether might be using VoIP in their phone system infrastructure, it's not a VoIP service from the end user point of view. However, if you have a "call in" phone number for Skype or one of the handful of other softVoIP clients, or even a hardVoIP phone number, there's no reason why you cannot enjoy VoIP benefits from your end.

In fact, because Ether also lets you sell digital content to clients via email or by downloading from your website, you could offer extra services. For example, if you are using a SkypeIn number, you can record calls and offer clients a copy for $0, or even a small fee. If you have voice-to-text software, you could even offer a text transcript, maybe in PDF form, for later download from your site - again for free or fee. Additionally, you could offer language translations of the transcript.

You can essentially set up a consulting practice for nearly any type of business (there are a few restrictions) for next to no cost. (For example, you can use a free-hosted site, but I wouldn't recommend it.) You can do followups by email or downloadable documents, if necessary. The options for businesses are endless, even if you don't want to do a lot of talking.

For example, let's say that you do web analytics work, say with a basic package rate of $500. Set up one Ether profile that gives a limited number of free 15 minute calls. Then set up a second profile that provides a 10-15 minute call for $250, but provides the content via email or download at an agreed upon date. (I have yet to see the non-phone Ether interface, so I'm speculating about the email/ download setup.)

That means that a client calls for free and describes what they want done. The call is the equivalent of a free estimate, but in this case, the price is fixed. If they think you can do the job, and you want to, they call back immediately on the other Ether extension, pay for your service up front, and finish providing the project details, etc.

It might take you a week to finish, or whatever, but when you do, the client calls back on the agreed upon date for a second $250 call, and you complete the transaction. The client has their work and your Ether account will have this additional $250, as well as the $250 from the second call. You could obviously get more sophisticated in your setup and break things down into four calls.

Ether takes a 15% commission from each transaction, which doesn't sound too bad for the service they offer. Hopefully they'll consider integrate with a softVoIP client such as Skype (because of it's Paypal connections) or an open source client such as Gizmo Project. For video calling, there's also Sightspeed, which would make it possible to offer consulting services with visual instruction, such as language pronunciation lessons. To summarize, Ether's a great concept, with room to grow in the VoIP arena to become a killer application.

Is VoIP-Based On-The-Fly Language Translation Possible?

Any sort of voice-based application is eventually going to beg the question of whether there is more than one language in use, and whether languages can be used interchangeably. In countries such as Canada and the United States, that have a sizable immigrant population, and where VoIP applications like Skype have really taken hold, this is an important question.

While many countries in Europe, Asia and Africa are officially bilingual, Canadians and Americans (and probably citizens of Australia, New Zealand and the U.K.), are for the most part unilingual, speaking primarily English. Some do speak French or Spanish as a first or second language. But there is a sizable portion of recent immigrants - particularly the young - or 1st- or 2nd-generation born, who may lose fluency with their mother/ heritage tongue, or never gain it in the first place.

It's often the latter citizens who while trying to uphold their culture and keep up contact with any family back home, often find a language gap. (Sociologists claim that clothing and then language are often the first characteristics of immigrants to change.) They'll speak in broken English blended with their mother/ heritage tongue, instead of fully in the latter. Technology such as VoIP-based language translation may be able to help them, or anyone else who wants or needs to communicate with people in another language.

For example, I can understand 95% of what is spoken to me in my mother tongue, but when I try to speak it, my words are often garbled. So if I want to converse over the phone with my grandmother, my words have to be translated to her. Or my grandmother has to do most of the talking and questioning. I respond to her in fragments, with poor tense and possessive nouns. But being the quiet woman my grandmother is, she won't do that. That means I rarely speak to her, beyond a hello.

No doubt I'm not the only North American to lament my lack of fluency in another language. There's a problem desiring a solution. So consider: what if you could simply speak English, or some other widely-spoken language, and your call would be translated on the fly? There already are voice-to-text and text-to-voice translators in several languages, as well as language-to-language text translators. The next step is to efficiently translate voice-to-voice in two or more languages. Universal translators anyone?

While universal translators to cover every Earth language may be improbable, natural language processing and speech recognition could foreseeably be combined to offer on-the-fly translation between two to five languages simultaneously - especially between languages that are closely related to each other. In a nutshell, language translation works on grammar trees. Once the grammar trees of two languages are properly paired, it's easier to write software to do the translation.

True, there are translation issues such as the difference between colloquial language and literal meaning to be worked out. That is not an easy or even short-term process. Also, processing power and grammar tree storage space are factors. But VoIP-based systems are far more likely to achieve speech translation than PSTN (Public Switched Telephone Network) ever could.

Seems to me like there's an application opportunity just waiting to happen. And not just once - over and over in different pairings of languages. All someone has to come up with as a starting point is some VoIP- and XML-based protocol that everyone else can follow, hopefully in an open-standard that can be shared.

Digital Audio Voice Signatures for Payment Authorization Via VoIP

VoIP ubiquity in software and hardware [1, 2, 3] is just around the corner, and it's likely to come in (now) familiar packages. Some of these VoIP voice applications are already here, some just arrived, and countless others are on their way. Imagine being able to initiate a VoIP call via Microsoft Outlook, just by clicking on a contact's name in your address book. Your familiar email client becomes a VoIP client. Or maybe you want to send a Paypal payment via Skype, or track and buy something from an eBay auction via Skype.

Of course, you can already do all of those activities, and many people have. I don't have sales figures for Skype-based Paypal payments, but it's pretty obvious that electronic payments in general are increasing. That's true whether via the Internet, through RFID-enabled smartcards or smartphones, or with biometric devices that incorporate RFID. In fact, it's said that India will have the largest market for contactless electronic payments via cell phones, with possibly up to 100 million users.

While I have a bit more faith in the security of hybrid biometric-RFID contactless payment systems, I'm not so sure I'd want my cell phone, or Skype or Outlook software, to be able to make a payment without my explicit authorization. So it made me wonder if there could be some way to authorize e-payments via VoIP, in terms of a digital audio voice signature.

The theory's long been put forth that each human voice is unique (notwithstanding comedian and impersonator Rich Little). While that theory has had a bit of difficulty in courts of law in the past, newer research suggests that it's true. It wouldn't be all that difficult, then, to take a voice scan for authorizations as an alternative to fingerprints.

It's my feeling that such an alternate will be more welcome than biometric scans. The reason for this may be purely psychological. Human beings have been familiar with voice recordings for decades. So recording their own voice does not make them uncomfortable. Biometrics, on the other hand, is a new science and the general populace does not have first-hand familiarity with it, unless they work in secure-access offices, military bases, or laboratories.

Of course, biometrics could be combined with VoIP technology for secure authorizations. However, my feeling is that such a combination would be unnecessary and more costly when digital audio voice signatures could be used reliably instead, and would probably have wider acceptance.

Sources: Owl Investigations - Aural Spectrographic, TC-Helicon - Voice Modelling Parameters.

Skyping and VoIPing With Music Fans

Alt-Rock band Coldplay, in a brilliant marketing move, embraced VoIP via a promotion launched by Skype and chatted a few minutes with a couple of  adoring fans. Could this be the music promotional tool of the future? Want to watch a bit of a live show, anyone? Throw in video to get V²oIP, and fans will be going crazy, maybe to the point of wanting to pay for a few minutes of access.

Talking with musicians was something I did quite extensively in the early 90s, sometimes in person, often on the telephone, for all the interviews and profiles for my now dead print rag. And while musicians usually called me, on the record company's dime, sometimes I had to call. And it was costly back then. VoIP could have save me a bundle, had it existed.

A common problem, though, was that bands sometimes had to cancel calls because they were on the road and they'd lost their calling code. Or it was nasty cold outside and standing in a phone booth wasn't an option. Or insert any reason here. VoIP means not having to stand in a phone booth. Unless you want to.

Now, with an application like Skype, bands can do meet + greets with journalists/ bloggers, participate in interviews, run CD promotions, and generally communicate with fans and other interested parties. Having multi-person chats with bloggers - bloggasm anyone? - could reduce or even replace the mass interview sessions musicians sometimes have to do, and typically detest, on a daily basis during a tour.

Keeping in touch with fans, in particular, is more than possible with VoIP applications, and it might just be a way for record companies to revitalize the industry and deflect attention from music downloading.

Heck, I'd pay a few bucks to talk a few minutes with some musicians. And now that eBay owns Skype, I wouldn't be surprised if musicians started auctioning off a few minutes to play a custom tune over Skype for some lucky bidder.

Let's Make An eDeal - Online Gaming Gets VoIP

With online poker, especially Texas Hold'em, being so popular these days, it's not surprising that one of the first online gaming applications of VoIP is for poker.

Playwize's PokerWize will online players not only see their 3D avatars - like many online games - but they'll also be able to talk to each other.

This is probably just the beginning in a string of VoIP-enabled game applications. Take things one step further and imagine being able to play word games with someone across the world, or have a chess game where VoIP enables you to control the pieces.

While Video over IP (also VoIP) is not something online poker players will want, it's a possibility as a proxy for live games. Maybe it's a stretch, but I can also see several frivolous V²oIP (Voice + Video over IP) applications, such as online game/ quiz shows.

Can you imagine shows like Jeopardy or The Dating Game in online form. All-time Jeopardy champion Ken Jennings wouldn't have had to worry about running out of clothes. Although an eDating Game might need something like telepresence suits to make it worthwhile.

Sources: PocketLint.

SingTel to deploy VDSL2 broadband technology

With SingTel preparing to deploy VDSL2 broadband technology, Singapore is all set to receive a high-speed internet access network that will allow 100Mbps over existing copper infrastructure.  Based on Ericsson's EDA VDSL2 technology, the network will make possible a triple play service (Video, Voice and Data) over the legacy local loop. The striking feature of this latest VDSL2 standard is that it uses Ethernet that facilitates the service provider to use VLANs as the delivery apparatus across the entire access network.

Via: [VoIP News]

Samsung launches Enterprise Networking Solutions

In an effort to mark its presence in the VoIP market, the leading mobile phone manufacturer Samsung has developed an enterprise networking solutions. The networking 'Ubigate iBG' provides routing, switching, security and voice integration (both analog and digital). The product also supports VoIP.

The company designs four types of iBG product lines, two for small business and one each for medium and large enterprise. The company will release the fourth model meant for large enterprise second quarter of next year. The remaining three models will be available this year.   

Via: [IP Telephony]

Oversea students develop FTS for long distance calls

Using VoIP, some of the former oversea students in America have developed an internet telephony system called FTS.

People can make free long-distance calls by connecting FTS with a broadband router or modem. They just need to pay for the device not for the service. The device cost comes around US$150 per set. 

The students claim that they developed FTS for their relatives and friends to effortlessly keep touch with those living overseas to skirt long distance phone costs.

Via: [THANHNIEN News]

CSS, G&T deploy SteelheadR appliances

Contech Stormwater Solutions (CSS) and G&T Conveyor have installed SteelheadR appliances of Riverbed Technology Inc. The deployment is aimed at bringing rapid improvement in application performance, enhance VoIP and video quality of CSS and G&T. In addition to these, CSS and G&T also want to transfer of large design and database files over WANs thorough this deployment. CSS is a renowned company with $100 million business. Its offices are located in Maine, Maryland and Oregon .

These are linked by T1lines having 768 kbps of that bandwidth designed for video and another 128 kbps diverted towards VoIP. G&T is a manufacturer and designer of bagging handling and explosive detection equipment for most important airports of the world.

Via: [dBusinessNews]

Cistera launches VoIP application for Call Centers

Cistera Networks has launched two new products - the Cistera ConvergenceServer 7500 and Cistera CallCenterEnterprise v1.7 . Off these two products, the Cistera CallCenterEnterprise v1.7 is highly useful for the call centers. It allows  the Call Center admins and supervisors some extra features like remote monitoring, screen capture, integrated instant messenger client with better presence support, coaching and desktop remote control. In addition to these, CCEv1.7 also ensures reporting, remote Web-based monitoring and QA sampling. While the Cistera ConvergenceServer 7500 supports 100,000 directory users and 4 Intel Xeon Processors operating the Cistera v1.6 platform.

Via: [VoIP Central]

Viewqwest deploys Tekelec 6000 VoIP Application Server

Singapore-based Viewqwest has deployed the Tekelec 6000 VoIP Application Server in the Singapore to provide its One Voice hosted business-class voice, video and data services. This is the first and foremost deployment of the server in Singapore. Tekelec, a leading network applications company has supplied about 190 Tekelec 6000 systems throughout the world. 

Viewqwest, CEO Vignesa Moorthy says,

Tekelec's VoIP application server is a highly efficient, cost-effective solution that offers us a range of business-class, next-generation services and the scalability to customize feature sets for a virtually unlimited number of subscribers while maintaining our competitive rates. 

After getting the reorganization from Viewqwest, Tekelec is now planning to expand its networking through out Singapore.

Via: [VoIP Central]

MSAG Based Routing For VoIP 911 Calls by TCS

TCS would be integrating a Master Street Address Guide based routing feature in its existing VoIP Positioning Center. This solution can be used for both static and nomadic VoIP emergency calls.

As per Roger Hixson, NENA technical issues director:

With the introduction of this new call routing feature, TCS demonstrates a commitment to improving the quality of E911 service for VoIP providers and PSAPs. This development also supports the intent of NENA’s recent VoIP Position Statement on use of MSAG-based call routing.

Via tmcnet

CE-Infosys Releases High Security VoIP Solution

CE-Infosys has released free secure VoIP solution, [Closed Talk] which includes enhanced features and maintains highest standard of security for free internet based phone calls. It incorporates modern high end security technologies and focuses on giving customers the highest security and privacy required for VoIP communication over the internet.

Combined with highest security and ease of use, [Closed Talk] brings benefits to the customers along with the advantages of VoIP without complicated setup and needs of additional software. This solution reduces VoIP security risks and vulnerabilities for both organizations and home users. It can be freely downloaded at www.ce-infosys.com

Via hardwarezone

Cable & Wireless Deploys NexTone’s MSX

Cable & Wireless has deployed NexTone Communications Multiprotocol Session Exchange (MSX) which would allow Cable & Wireless to utilize VoIP for its international carrier customer base. MSX is helping the company tackle new market opportunities while building a substantial base for IP service offerings for the future.

Dan Dearing, V.P., NexTone said:

NexTone's session management technology goes beyond basic IP interconnects to enable Cable & Wireless to intelligently manage all aspects of the real-time IP sessions running on the company's networks. This gives Cable & Wireless the option to add new revenue-generating services, routes and partners to its network for greater service reach for customers, and help maintain its competitive edge.

Via tmcnet

GlobalTouch Deploys Video over VoIP Application

GlobalTouch Telecom has deployed a plug and play video over VoIP application which combines video telephony hardware and SIP software functions. It worked with Leadtek research for developing the videophones and overcame video signal handling and compression challenges.

There are two ways of making video calls, Leadtek desktop phone or GlobalTouch’s video softphone which comes with its VoIP service. The software has been developed only server side T.38 Fax support. It implies that the company can provide double fax transmission speed and higher level of transmission reliability.

Via xchangemag

SigProvider 2.0 Announced By Signate

Signate has announced SigProvider 2.0 which is a major enhancement to its hosted VoIP PBX solution for telephone providers with up to 3,000 extensions. SigProvider enables service providers to offer both consumer and enterprise telephone service.

It includes a new forms based interactive voice response builder so that customers can configure their own voice response system. It can be scaled for increasing call capacities by adding additional call management servers. The price of SigProvider starts from $8,000 for the first customer provided 32 bit server which includes installation and billing integration through a customer provided Radius server.

Via businesswire

Vistula to Provide VoIP Solutions in Mexico

Vistula Communications has been selected by Protel i-Next for providing VoIP solutions for providing VoIP solutions.  This solution would be launched in Mexico in the second half of 2006 and would be made available to Protel’s customers’ base of call centers, cable operators and small and medium sized enterprises.

Alejandro Pineda Mathus, Marketing Director, Protel said:

We will provide our customers with significant savings on their domestic and international long distance phone calls. We chose Vistula's V-Cube IP PBX technology as the platform for these products for its scalability, low cost of entry and advanced features.

Via upi

China's Taxation Bureau Adopts VoIP

Xinjiang Uygur Autonomous Region Taxation Bureau has adopted Cisco's Unified Communications solution. This up gradation is being regarded as the largest Unified Communications project based on IP telephony within China's taxation system. Around 2,600 Cisco Unified IP phones are being installed at fifteen state level and one hundred and fifteen country level offices. The completion of the installation was done at the end of March 2006 and the bureau is expecting the project to bring profits within one and a half years.

It is expected that the system would reduce the communication costs of the taxation bureau and would increase its operational efficiency.

Via: [VoIP Central]

Verizon Unveils VoIP Solutions for International Markets

Verizon has introduced a broad portfolio of VoIP solutions for the international markets which offers a simple and an efficient way for global customers to transit their voice and data services to IP technology.

When the services would go over the managed networks the customers could realize the cost, features and productivity benefits of VoIP without compromising on the reliability and quality which they would have come to expect from traditional business telephony. The solution also includes IP Trunking, Hosted IP Centrex and managed IP PBX as well as an enhanced IP Integrated Access offering to the Verizon VoIP portfolio.

Via itnewsonline

Is Vonage Developing Wi-Fi Calling Solutions For Sony And Nintendo?

T3 has reported that a source close to Vonage has confirmed that the company is developing Wi-Fi calling solutions for Sony PSP and Nintendo DS. It was already known that both the systems were capable of implementing VoIP but for the first time we are getting clear indications that a major VoIP player is making a move in this direction.

Even if we consider it to be a rumor the details offered are quite less as regards how Vonage is planning to implement VoIP in the disparate devices. The possibilities which can be considered include a firmware update for the PSP or DSpeak like software or an Opera like cartridge for the DS.

The company has refused to comment on the matter so keep on guessing till some formal announcements are made.

Via engadget

Next Generation Active VoIP Recording Solutions from NICE

NICE Systems has released the next generation of VoIP active recording solutions. The latest version of NICE's solutions incorporate the latest VoIP enhancements in the market.

NICE's new active VoIP recording solutions is included in Nortel's new Duplicate Media Stream over IP (DMS-IP) architecture.  They are also present in active recording for Cisco's CallManager, and for Avaya's Communication Manager API, and IP-phone applications.

NICE's scalable VoIP software solutions are certified by all major VoIP switch vendors and are aimed at all segments of the market.

Via SecurityPark

VOIP-based call center

Supershuttle, a transport operator in NewZealand is the latest organization to switch from the old PBX-based communication system to VoIP services.

Supershuttle aims to save money and much more by switching over to VoIP. It has 14 offices in New Zealand, all of which using different phone systems. The company had to pay huge communication bills because of toll calls, afterhours diverts and other necessities of the business. A company official says that their phone bill used to be their second largest expense, including all those internal calls and diverted calls.

Supershuttle not only plans to save money by switching to VoIP, the new VoIP system will also help it in tracking on-duty performance and activities of workers.  Moreover, if it receives too many calls on any day, Supershuttle might use remote call-takers who can work from home using broadband connections.

Via ComputerWorld

How to track VOIP

The FCC rules has made it mandatory for all VoIP service providers, each provider, at all times, must know where the customers’ VoIP phones are and who is assigned to each phone, each time a call is made. 

eTelemetry has introduced Locate911, which is a plug-and-play device. Locate 911 can help VoIP service providers comply with the FCC’s e911 mandate, without any hassle.

Very simply, Locate 911 is able to provide real-time VoIP location tracking as it automatically links the VoIP phone to building/room and the person.

Via Processor.com

VOIP on Mobile phones soon

We are talking about VoIP on Wi-Fi to be more precise. This will be made possible by a new technology called Unlicensed Mobile Access, or UMA.

Wi-Fi telephones and walkie-talkie-like communicators are common in hospitals and offices. However, manufacturers and mobile carriers plan to bring it to the mainstream market. They will link standard cellular networks to the mishmash of Wi-Fi hotspots. This will undoubtedly result in cheaper mobile minutes and coverage over a larger area.

Unlicensed Mobile Access, or UMA, will help consumers who have high-speed Wi-Fi routers deal with instances of poor coverage in their houses or apartments. Moreover, UMA helps mobile operators as they are able to expand their coverage without having to install any piece of expensive new infrastructure.

Via Seattle PI

VoIP over CDMA2000

AudioCodes, a VoIP solutions company from Israel, will work in tandem with CDMA giant QUALCOMM to demonstrate a solution for end-to-end VoIP over CDMA2000 1xEV-DO Rev. A technology during CTIA Wireless 2006.

If that appears too technocal, let me translate that. The two companies make VoIP calls using EV-DO Rev technology. This is now possible since AudioCodes now supports the VoIP Enhanced Variable Rate Coder (EVRC) codec, developed by QUALCOMM. If this demonstration works, mobile operators will have one more reason to implement VoIP over their networks.

Via Compact PCI Systems

A new approach to securing VOIP

Secure VoIP services are a must especially for business consumers. It is a new field so experts have not yet listed out all the dangers. Consequently, not all the responses are yet ready.

Recently, Certicom Corp. launched Certicom Security for VoIP. This solution integrates everything - multiple, integrated modules that implement key security protocols such as IPSec (IP security protocol), SSL/TLS (secure socket layer and transport layer security) and DTLS (Datagram Transport Layer Security protocol). The solution also provides you the underlying cryptographic algorithms, trusted boot, secure provisioning and code signing technology.

Via Thomas Net

VoIP in the Financial Vertical market

This lucrative segment of the market is sure getting hot. For example, recently, Franklin Collection Service, Inc. has chosen to implement Cistera's GoVertical Financial Solution at its call center. Franklin Collection Service, Inc. is a collection agency headquartered at Tupelo , Mississippi, with offices in Tulsa, Oklahoma and Nicaragua . 

Franklin has selected the Cistera ConvergenceServer™ and CallCenterRecordEnhanced™ to provide monitoring and recording capabilities for its call center. Moreover, the Cistera ConvergenceServer™ adds critical competency and features such as text and audio broadcasting, messaging, recording and content streaming within a Cisco Communications environment. Cistera ConvergenceServers may also be configured to support small, medium and enterprise installations.

Via MarketWire

Sonus leads the Japanese VOIP solutions market

Analysts say that Japan is setting the pace for Telecom Operators regarding the migration to all-IP networks. That means supplying equipment and stuff to Japanese Operators is a profitable business. And Sonus does exactly that. According to a report issued by Synergy Research Group, for the full year 2005, Sonus led the Japanese VoIP solutions market with a 36.1% market share position, more than twice of what any other equipment provider. Sonus has also cornered around 76% of the high-density gateway market and accounts for more than 61% of the total gateway market in 2005.

Sonus has formed lucrative deals with all the big operators in the country.  It recently announced a multi-million dollar contract with KDDI, the 13th largest service provider in the world, according to Total Telecom's 2005.

Via Yahoo Finance

COTRAN i-BCS from COTeL: Presence Collaboration meets VoIP

COTeL has introduced the COTRAN i-BCS, VoIP communications solution built for Small and Medium-sized businesses. It enables businesses to create their own Private Enterprise VoIP Network. Using COTRAN i-BCS, businesses can create private Skype™-like corporate VoIP networks complete with Meet-me-Conferencing and Corporate Instant Messenger (IM).

Via PRWeb

Adtran launches NetVanta 7100

Adtran is set to expand its VoIP portfolio by launching NetVanta 7100. It is a VoIP office in a box solution that would integrate voice, high speed internet and security in a single compact platform. It would be addressing the North American IP private branch exchange market which is valued at more than $3.5 million.

It is a complete IP telephony and data networking solution for business locations up to fifty employees. It is an all in one platform which includes an IP PBX, fill features IP router, voice mail, firewall, 24 port powered fast Ethernet switch with Gigabit uplinks, VPN and two expansion slots for Network Interface/Voice Interface Modules.

via [TheStreet]

Top Global introduces world’s first 3G mobile wireless Skype solution

Top Global has introduced World’s first 3G Mobile wireless Skype solution based on their 3G MobileBridge platform. The company has developed and patented the flexible MobileBridge platform which serves as a gateway/router that links 3G network and WLAN/LAN networks which enables customers to enjoy the flexibility and convenience of wireless internet in truly mobile and remote environments. It works anywhere within cellular coverage. It is not limited to office, home, cafe or open public. Users would be able to make free domestic and international calls, call non-Skype users and regular phone numbers for a small fee with the SkypeOut feature.

via  [BBWExchange]

BroadSoft’s Mobile PBX Application to give boot to major mobile operators

BroadSoft has made an announcement regarding the new deployments of the Broadworks Mobile PBX Application to give major mobile operators around the world the ability to deliver an enterprise communications solution. The application would unlock the enterprise market for its mobile operator customers and trial deployments. Mobile operators who use the Mobile PBX application would increase value to end users. Major service providers in Asia, Europe and North America are using the Mobile PBX application that would offer new productivity and mobility communication services to enterprises. Mobile operators have been deploying BroadSoft in order to enter the largely untapped mobile enterprise market through Broadsoft’s strong channel relationships for IMS solutions.

via  [Unstrung]

babyTEL introduces a comprehensive VoIP solution for small and mid sized enterprises in Canada

babyTEL has announced a comprehensive VoIP solution which would make it easier for small and mid sized enterprises in Canada to make transition to this cost effective and beneficial technology. babyTEL is taking steps to ensure that small and medium sized businesses get the benefits of this technology as generally smaller firms are not able to reap the benefits of  the new technology quickly. The babyTEL Enterprise is meant for firms having employees between 2 and 300, multiple offices and a flexible workforce that works remotely from hotels, home, airports and other locations. There is a great potential for VoIP services in the small and mid-sized enterprise market.

via  [NewsWire]

CYGCOM and Paraxip Technologies introduces next generation VoIP Gateway Bundles

CYGNOM and Paraxip Technologies have announced the availability of new VoIP gateway product bundles optimized for speech, IVR and contact center applications. It includes the Parapix Gateway Software Version 2.0 which offers Open Protocol Translation, dynamic call transfer selection and advanced call progress analysis over the SIP protocol. It also includes the Intel NetStructure Digital Interface Card which offers wide PSTN protocol support for comprehensive support of PBXs, Switches and ACDs at a reduced price. It also has the Intel NetStructure Host Media Processing Software which offers highly flexible software based media processing functions which leverage the power of general purpose processors.

via [BusinessWire]

Meru’s Radio Switch receives Internet Telephony’s 2005 Product of the Year Award

Meru Networks Meru’s Radio Switch has been chosen by Internet Telephony magazine as the recipient of 2005 Product of the Year Award. The radio system can deliver up to 648 Mbps of WLAN bandwidth which utilizes 802.11 standard radios and scales up to 1.2 Gbps of bandwidth across its coverage zone. The radio switch can achieve this performance by layering up to twelve channels in a single coverage area. Meru’s radio switch family includes the four radio RS 4000, eight radio RS 8000, twelve radio RS 12000 and a built in omni directional antenna which is awaiting a patent.

via  [PRNewsWire]

Korea all set to be on the forefront of International IPTV Standardization

The proposal for composition of special group for standardization of IPTV in Korea by Telecommunication Standardization Sector (ITU-T) will enable Korea to be on the forefront in the telecommunication equipment standardization sector. ITU-T has proposed a focus group for standardization of IPTV set top box, gear and service which would be launched by July. The Ministry of Information and Communication is planning to set up a special group to arrange for a focus group in partnership with TTA and ETRI in the first half of 2006.

via [EtNews]

VOIP Starter Platform from Netcentrex

Netcentrex has started selling its VOIP starter kit in the United States.

The Starter Kit consists of an entry-level VoIP platform for residential and corporate customers. The Starter Kit contains a feature-rich administrative and provisioning environment and makes end- user customization possible.

Its is a scalable platform and can be configured for as few as 1000 subscribers or several hundred thousand subscribers. It works on all types of broadband connections.

Via [teleclick]

Quality management tool Piqua to address VoIP issues

Quality management tool, Piqua from Texas Instruments will make VoIP reliable and better sounding. It will help service providers scrutinize devices, services and networks. In many cases it will permit the infrastructure to heal itself. This will help providers to reduce costs, scale up voice and other multimedia services in order to meet the demand. It will be available later this year in products from console developers and carrier equipment vendors. It will also replace the diagnostic tools built into the traditional circuit switched phone system.

Via [Computerworld]

Combination of VoIP and Wi-Fi adopted at Rhode Island Hospital

Doctors and nurses at Rhode Island Hospital are making use of Wi-Fi and VoIP system that allows them to access patient data, test results, consult expert sources and order procedures at bedside. Wi-Fi connects the Vocera units around the necks of nurses enabling them to communicate to each other hands free even while giving an injection to somebody. Patients and families can access the internet while they are in the premises of hospital and track critical pieces of equipment by locating their electronic tags. A combination of VoIP and Wi-Fi is being adopted because of the simplification it offers.

Via [PBN.com]

New website portal released by VoIP-Internet-Products.com for complete VoIP solutions

VoIP-Internet-Products.com has released a new website portal for complete VoIP solutions for multiple choices of major brand providers for IP Broadband and Telecommunications service solutions. This portal is customer focused and it presents the best value in communication products and services. It offers the convenience to compare different carriers and providers based on their location to get access to the inexpensive and high quality services. It also offers additional services ranging from Conference Calling solutions to e-Fax along with traditional local and long distance land line services.

Via [PRWeb]

VOCAL Technologies offers iPBX and PBXtender solutions

VOCAL Technologies has introduced its new PBXtender and iPBX solutions for home and office VoIP telephony services. These will help VoIP users to improve telephones in their offices or homes. These instruments will prevent the need for extra wiring or Wi-Fi networks. These solutions will be beneficial to those users who didn’t want a single phone, were cynical about shifting from traditional phone service or didn’t want to spend huge amounts for availing VoIP service. These solutions will provide to the customers hassle free VoIP service. iPBX and PBXtenders will provide VoIP service to other phone lines in office or homes without infringing FCC power regulations or disturbing telephone services or DSL.

via [TmcNet]

Yak Communications introduces unlimited VoIP calling plan

Yak Communication has launched an unlimited VoIP calling plan which would allow Canadian customers to have a U.S. phone number. This plan would provide subscribers a U.S. or Canadian local phone number to make unlimited outbound and inbound calls in Canada and continental U.S. in order to utilize this service one must have a broadband internet service. This plan also provides features such as unified messaging, call waiting, caller ID, voice mail and conference calling. The plan is available for $ 24.99 per month plus VoIP gateway cost if required. Computer would be required for making calls but if one wants to make calls using a traditional phone VoIP gateway would be required.

via [DigitalHome Canada]

VoSKY : World’s First Skype Call Cente

ActionTEC has announced VoSKY Call Center in CES 2006 which would enable users to call his/her Skype buddies and also receive Skype calls to his normal phones or mobiles without the need of a broadband connection. This could be the world’s first mobile Skype Solution which will help VoIP/Skype to reach the mass market more effectively.

This product and service can be used by anyone who doesn’t have a broadband connection by installing the VoSKY call center, and users can call the numbers to make Skype, SkypeOut Calls. VoSKY call center can also be very effective in the countries where broadband hasn’t reached the core and can get good business for Internet Café Owners to provide their customers long distance callings at negligible costs. 

via [Mobility Today]

Uniden to display products at the CES

Uniden America Corporation is displaying its innovations in VoIP and Bluetooth technology at the CES, which is being held from January 5-8 in Las Vegas. Uniden has also teamed up with Microsoft and the partnership will introduce a digital cordless phone that will function as a traditional phone and also enable users to receive and make Internet calls by means of the Windows Live Messenger phone. businesswire.com reports:

With the WIN 1200, customers will have three calling choices with the solution.

Read More: Launching Peer-to-Peer, Cutting-Edge VoIP and Bluetooth Offerings, Uniden Marks 2006 Trends at CES

VoIP Phone reference platform from PMC-Sierra

PMC-Sierra has released a VoIP Phone reference platform which facilitates toll-quality voice services and PSTN-like quality. The MSP2020 secure VoIP processor and certified voice processing software have a simple design; thereby bringing down the total BOM. tmcnet.com reports:

"Communication is key to every-day business and understanding," said Jim Jacot, VP of sales and marketing at Acoustic Technologies.

Read More: PMC-Sierra Announces Secure Wired and Wireless VoIP Phone Reference Platform

VoIP in schools

VoIP is making its presence felt everywhere and has even managed to impress educational authorities. The Clark County School District (NV), David Douglas School District (OR), and Saugus Union School District (CA) have deployed VoIP and are reaping benefits in the form of flexibility in communication, low costs, and better connectivity. thejournal.com reports:

In 2001, frustrations with this system led to a call for widespread improvements, and Information Systems Manager Keith Seher responded immediately.

Read More: VoIP to the Rescue

ECG to provide telecom solution for TelCove

TelCove is a company providing business critical telecom solutions. The company had selected ECG for a turnkey project that will cover installation and integration of a VoIP platform for TelCove. The platform included Broadsoft Broadworks and Lucent LCS Gateways and was deployed and rendered serviceable in its Beta phase inside seven weeks. The call routing plan was developed by ECG, which also created the project plan and laid down the engineering prerequisites.

Avaya to develop contact center

Showtime Arabia, a major pay TV provider, will be deploying an IP-based contact center solution from Avaya. The contact center will enable Showtime Arabia to improve communications with its subscribers in 20 countries. tradearabia.com reports:

Given the twenty-four hour nature of Showtime Arabia's business, the company is operating across the Middle East to support subscribers.

Read More: Avaya to lift Showtime customer service

NetIQ® by NetIQ Corp.

NetIQ Corp has stated that it will be providing its NetIQ® VoIP Management solutions to MSPs like Analysts International, Calence, and Getronics. The solutions from NetIQ will help the MSPs to deploy VoIP applications quickly and also to ensure that the performance and availability of the applications. sanjose.dbusinessnews.com reports:

"We chose NetIQ because we look for best-of-breed vendors with whom we can partner to achieve the successful management of our customers' environments."

Read More: Managed Service Providers Deploy NetIQ for VoIP Management

Hosted solution from M5 Networks

M5 Networks, which is a hosted VoIP provider, and IP blue Software Solutions are jointly offering a solution that will allow VARs to keep in touch with remote workers. messagingpipeline.com reports:

The IP blue softphone software turns any PC or laptop into a fully functioning VoIP telephone.

Read More: M5, IP BLUE MOBILIZE HOSTED VoIP OFFERING

VPNremote by Avaya

Avaya has released a new software referred to as the VPNremote, which will provide Avaya IP telephones with VPN remote capabilities. Upon integrating the software with the IP phones, telecommuting employees will be accessible all the time. The facility is available with the 4600 IP Telephones series from Avaya. The software enables home-based workers to plug the phone to a power source, connect to a broadband router, and the phone becomes operable. voipplanet.com reports:

One recent customer of the product line was the U.S. Red Cross, which deployed the VPNremote and IP phones as part of its disaster recovery work after Hurricane Katrina devastated the Gulf Region last fall.

Read More: Avaya Loads Up on Unified Communication

VoiceRD by Novacoast Inc.

VoiceRD by Novacoast Inc. is a Rapid Deployment solution for the open source Asterisk IP PBX. VoiceRD integrates the hardware, software, security, and identity management. The VoiceRD bundle also includes codec licenses and a web-based management tool for the IP PBX. voipplanet.com reports:

VoiceRD includes the recently released Asterisk version 1.2 and runs on a Linux appliance from PogoLinux with Novell SUSE Linux installed. On the hardware side, the Linux appliance also has a 4 port T1 card from Asterisk's corporate sponsor Digium Inc.

Read More: Novacoast Integrates Identity Management with Asterisk

VoIP management tools

Vendors such as Empirix Inc and Agilent Technologies Inc. offer VoIP management tools that enable enterprises to solve VoIP networking issues by acting on the advance indications that the tools provide. The tools monitor common VoIP-related problems such as jitter, delay, low QoS, packet loss, volume issues, etc. informationweek.com reports:

"We'll correlate for you in real time, and it'll say, yeah, the issue is at this particular port on this router," says Qovia Inc. COO Steve Mank.

Read More: Explosive Growth Expected For VoIP Monitoring Tools

ShoreTel Converged Conferencing 5.6

ShoreTel has launched the ShoreTel Converged Conferencing 5.6. It is an integrated conferencing and collaboration platform that works alongside the ShoreTel 6 IP telephony system and offers facilities such as integrated audio and web conferencing, enterprise instant messaging and document sharing. The platform offers greater security by virtue of being an onsite embedded Linux appliance and is not Internet-based. home.businesswire.com reports:

"ShoreTel Converged Conferencing eliminates the expense of using outside web conferencing services for our internal and customer collaboration," said Jason Colburn, IT manager at Optimal Solutions, one of the fastest growing suppliers of broadcast software in the United States.

Read More: ShoreTel Delivers Collaboration Platform

Semiconductor package from Ceva

Ceva has launched a semiconductor package for incorporating VoIP functionality into SoCs. The package, Ceva-VoP, consists of a fully programmable 200 MHz DSP engine and xpertteaklite 2, which is an integrated subsystem that has a memory cache. The solution enables users to avail voice coding, echo cancellation, and telephony interfaces. The Ceva-VoP is available in two versions; a co-processor version that uses a host CPU for voice processing and another version that uses the DSP for voice processing and network support.

The package can be used to support up to 8 voice channels with support for protocols like G.711, G.726, G.729AB, and G.723.1. Ceva-VoP provides echo cancellation, a wide range of telephony features, fax relay, and functionality for signaling and user interface management.

IP PBX solution from Linksys

The Linksys One Hosted Small Business System is an IP PBX solution targeted at hosted service providers and VARs. The system, which provides network discovery and automated configuration, features a 16-port services router and an analog VoIP gateway. Linksys One will be available in the US in the first quarter of 2006; MCI and NeoNova are two of the companies that plan to provide hosted services using Linksys One.

Net Satisfaxtion IP

Net Satisfaxtion IP is a T.38 boardless fax server launched by faxback. It can work with the existing IP routers and also with legacy PBXs.

IMS testing suite from Radcom

Developers and service providers can now make use of the IMS testing suite provided by Radcom to conduct pre-deployment evaluations of IMS network devices, services and architectures. The suite can be used for tracing transactions carried out via wired and wireless interfaces. The network suite can be used in conjunction with the Radcom SIP simulation tool for building call flows for conducting stress tests on the media gateways and SBCs.

Tenor MultiPath from Quintum

Quintum Technologies has added enhanced software to its Tenor MultiPath switches and gateways. The additions will enable automatic deployment of Tenor upon addition to a network. The software supports the mapping of ports enabling PBX extension functionality for branch offices. Support for DNS addressing, failover capability, and automatic load balancing are other facilities provided by the enhanced software. The software is available for download at the company website.

iLBC voice codec from GIPS

GIPS, which provides embedded solutions for processing speech, will be providing its iLBC voice codec to several leading companies in the VoIP market. HelloSoft and CIRPACK are two of the many users of the royalty-free software. iLBC-enabled solutions benefit from interoperability across different platforms. globalipsound.com reports:

These newest customers, who join other well-known companies such as Texas Instruments and AudioCodes on the growing roster of iLBC-enabled solutions, are planning to implement iLBC in a variety of existing and next-generation voice switches, gateways, and VoIP software solutions for various VoIP client devices.

Read More: Major VoIP Suppliers Select GIPS iLBC Royalty-Free Voice Codec

VoIP solutions from Marvell

A family of integrated VoIP solutions has been launched by Marvell. The offering, Marvell 88W8618, is targeted at the fast growing Internet voice communications market and is particularly meant for the VoIP residential gateways. The products on offer integrate VoIP processing, low power WiFi circuitry, a high-performance host CPU and peripherals into single system-on-a-chip (SoC) solution. yenra.com reports:

These VoWLAN SoC solutions significantly reduce overall system cost for OEMs because of extremely high levels of integration. In addition, Marvell has increased the performance of these offerings for their respective applications.

Read More: VoIP SoC

M-series for mpower

Juniper Networks has provided mpower broadband with routing and security solutions that will help in providing reliable and secure VoIP services. mpower provides hosted VoIP services and applications for business and residential clients. The M-series multiservice routing platforms, VF-series session border controllers, and Integrated Security Gateways with IDP have been provided by Juniper networks. The solution ensures simple and seamless deployment of a feature-rich VoIP service. Customers of mpower will gain network connectivity, automated billing facility, and VoIP portal customization. This offering by mpower is a useful addition to its already well-received suite of carrier and enterprise services.

The M-series multiservice routers are deployed in the core of the mpower broadband network. The network acquires advanced IP/MPLS capabilities and the capability of delivering voice-specific quality of service (QoS) policies. The M-series router works alongside the multiple VF-series SBCs that offer a high level of security and hosted NAT traversal. In order to ensure the security of the network and protect it from DoS and network attacks, the ISG 1000 is used. It integrates a firewall, VPN, and IDP.

XelorRate Service Quality Manager

The XelorRate Service Quality Manager from Xelor Software is a Linux application for Red Hat. It offers a QoS solution for Avaya's Communications Manager and Cisco's CallManager. The software uses SNMP and provides network configuration, dynamic bandwidth allocation, and automated packet prioritization. The software enforces the QoS policy from end-to-end after the initial configuration. It configures Class of Service and Differentiated Services for Layer 2 and Layer 3, respectively.

The bandwidth availability and the delay bounds are evaluated by using an SAC algorithm. Upon receiving a positive response from the SAC, the software starts to control the network resources. It uses the ACL entries on the access ports that are nearest to the point of origin and termination of the call in order to allocate bandwidth and manage the flow treatment. XelorRate is available at rates $ 14,700 and $ 61,000 for companies with 200 and 1500 seats, respectively.

Converged Mobile VoIP

Converged Mobile VoIP is a platform that allows mobile operators to provide PC-to-PC and PC-to-phone VoIP calls. It has been developed by Tatara Systems. 3GPP/2 compliant architectures are required for this technology.

Features like network authentication and authorization, presence and location management, etc are available with the platform. Other features include NAT and firewall traversal, SIP session management, voice call information, CDR generation, etc.

VoIP access solutions

VoIP access solutions from Quintum provide businesses with the opportunity to connect to a VoIP network without having to go through an expensive integration overhaul process.

BCM3368 by Broadcom

BCM3368, launched by Broadcom, is a VoCable SoC based on Viper, which is an enhanced RISC processor architecture. The chip does not need a dual processor or dual memory architectures for supporting multiple voice lines. The solution offers good scalability and wireless mobility. An ICSA certified firewall offers security. The solution supports peripheral interfaces such as multi-port Ethernet, USB, Bluetooth, and storage devices. The cost of the chip in sets of 100,000 is $ 25 each. It comes in 426-pin PBGA packaging

Norfa Lite

Norfa Lite is a billing and call management platform introduced by SysMaster. It works with managed IP Centrex and IP PBX systems that use end user CPE like IP phones for processing calls.

VoCable by Net2Phone

The VoCable solution offered by Net2Phone will be used by Cable & Wireless for providing VoIP solutions to a population of 3.6 million in the West Indies.

IP video phone solution by Ittiam

An IP video phone and video conferencing solution designed by Ittiam can be used in stand alone devices and can also be added to a set-top box. The solution uses SIP and the TMS320DM642 DSP from TI, which comes with a media engine, media controller, an adaptive jitter buffer, etc. The TI DSP comes with a built-in keypad, mic, speaker, and LCD display. HDD and storage devices that work with a CardBus interface are supported by the platform.

VoIP platform from Motorola

The VoIP open application enabling platforms introduced by Motorola use the FACT-SIP software and ComStruct CompactPCI voice boards. The platforms will aid in the development of IP PBXs and VoIP access gateways. FACT-SIP can be integrated with codecs, tones, and echo cancellation. ComStruct boards can support up to 2,016 voice channels and facilitate a broad spectrum of I/O functions.

Scientific Atlanta chosen by COMCOR

Scientific Atlanta will be providing headend and optical transport equipment and systems to the Moscow Telecommunication Corporation (COMCOR) to enable it to provide cable programming to Moscow residents. Equipment from Scientific Atlanta will enable cable operators to access signals from COMCOR. Scientific Atlanta will be the exclusive supplier to COMCOR throughout its two years of network expansion. COMCOR hopes to avoid interoperability issues and save on time by using a single vendor.

Fax over IP

Organizations are opting for new applications that can help them to converge their networks. Converged messaging ranks high on the list of applications that can provide tremendous benefit to an organization. Companies that have deployed VoIP infrastructure are looking to incorporate additional features that will help in improving operations and cutting costs.

Fax through IP is an alternative that several organizations are exploring as fax support (T.38) is already inbuilt in several IP routers such as Cisco, 3Com, Alcatel, etc. Fax over IP (FoIP) allows enterprises to leverage IP networks for transmitting documents over secure and boardless fax servers. Faxes can be sent from the desktop where the content travels in a digital form till it reaches the PSTN endpoint gateway. FoIP integrates well with existing applications and works with multi-function devices.

FoIP offers fast ROI which is a function of the faxing volume in a company. It eliminates the use of paper for faxing which itself is a major cost saving. There is no longer any need for dedicated PBX line cards and additional telephone lines for the fax machines. Since FoIP does not require dedicated hardware, fax boards, toner, paper, etc it results in a lower total cost of ownership. Boardless solutions also lead to less downtime as installations are easier and no reboots are required.

Boardless fax solutions also provide VoIP encryption and virus-free transmission that facilitates compliance with regulations like HIPAA, Sarbanes-Oxley, and Gramm-Leach-Bliley. A centralized archiving system ensures easy tracking and management of documents. techweb.com reports:

As the market heats up there are a range of VoIP solutions with various price points. For such companies also looking to roll-out network fax functionality, a review of the actual cost of a VoIP system as well as a comparison of a boarded fax server versus one that is boardless should be added to the IT checklist.

Read More: Boardless Fax over IP

IMS functions

IMS enables service carriers to combine several Internet applications with already present wireless services and has hence generated a fair amount of interest with the vendors. The IMS architecture is being developed by the 3rd Generation Partnership Project (3GPP). 3GPP has laid down several specifications that deal with various issues such as requirements, service aspects, technical realization, etc.

Call control, media processing, and gateways are three functions of IMS. The Call/Session Control Function (CSCF) handles the call control function. Signaling messages from SIP are processed by the CSCF. The three CSCF functions include Proxy CSCF which performs like an SIP User Agent and handles the forwarding of SIP requests and responses; the Interrogating CSCF is also an SIP proxy and functions as a contact point for the operator’s network and the users that are roaming that network; the serving CSCF is an SIP server that helps to maintain the state of a session as desired by the network provider.

The Media Resource Functions (MRF) manage media processing. The Media Resource Function Processor (MRFP) handles functions such as the processing of mixed incoming media streams, audio transcoding, announcements, etc; the Media Resource Function Controller (MRFC) performs a controlling function for the media stream resources within the MRFP.

The gateway interface consists of the Breakout Gateway Control Function (BWCF) which is an SIP server that performs routing functions to allow calls between networks. This enables calls to begin on the IMS and finish on the PSTN; the Media Gateway Control Function (MGCF) transforms the signaling protocols such that they can be used by a given network; the Media Gateway (MGW) is used for changing the media streams that are used on a network such as RTP to a format that can be used by another network such as the PCM.

DigiVoiceXE from Davacord

Davacord, which is based on Oklahoma, has stated that its DigiVoiceXE call and screen recording system now offers VoIP support. VoIP compatibility in the product enables concurrent recording, tracking, analyzing, and monitoring of voice and data. The product also facilitates a move to VoIP using VoIP switches from Cisco, Avaya, Nortel, etc.

VoiceEngine embedded technology from GIPS

Gemtek technology, which provides wireless broadband systems, is using the VoiceEngine Embedded technology developed by GIPS for its FreePP VoIP service. The VoiceEngine technology provides support for managing voice quality across the different platforms used for the FreePP service. Jitter, packet loss, delay, and echo are some of the problems that will be sorted with the help of the VoiceEngine Embedded technology. Different systems and platforms are used to provide global P2P VoIP services.

PMC-Sierra, VoIP device provider, uses the GIPS NetEQ for its MSP VoIP Processor family in order to obtain better sound quality and minimize jitter buffer delay.

VoiceConX ro use Edgewater products

VoiceConX, which is a hosted VoIP service provider, will be using VoIP products from Edgewater Networks for its business VoIP customers. Edgewater Networks will provide VoiceConX with customer premises products, network-based session border controllers and a VoIP-specific network management platform.

VoiceConX will be using the EdgeMarc device and EdgeView VoIP Support Systems for quality management at both the customer’s end and at the core network. EdgeView VoIP Support System will facilitate remote management of the ConX-IP subscribers. EdgeMarc improves VoIP service quality for real-time media. tmcnet.com reports:

Edgewater made news last month when it partnered with VoIP software supplier Atreus Systems to combine the EdgeView network management system, EdgeMarc converged network appliances, and EdgeConnect managed PoE switch with Atreus’ Multi-Service Provisioning Solution.

Read More: Edgewater Delivers ConX-IP

Solution from MetaSwitch

MetaSwitch has introduced the VP2510 softswitch, the MG2510 media gateway, and the SG2510 signaling gateway. The solution is meant to be deployed in large networks and can be used while migrating to IMS networks.

The softswitch which also supports up to 15,000 subscribers can be configured with an integrated call agent, media gateway, media server, etc. The price of the solution will vary with the configuration.

The J7844A

Agilent Technologies has launched the J7844A, which is a VoIP call trace signaling analyzer. It is used for identifying signaling inconsistencies between ends. It can be used with voice technologies such as VoIP, 2G, 2.5G, 3G, etc. It facilitates quick detection and resolution of issues because of its real-time call trace functionality. The J7844A will cost $ 11,000 and will be in the market in December.

Signaling gateway from Stratus

The Inter-Network Services Signaling Gateway, which has been developed by Stratus Technologies, is used for allowing traffic to pass between SIP, legacy, and 3G networks.

TR1000 media processing board

A quad density T1/E1 version of the TR1000 media processing board developed by Brooktrout Technology is now available. TR100 is also available in versions for analog, BRI, and single/dual T1/E1. This latest board release is compliant with the Restriction of Hazardous Substances directive.

SessionSuite Architecture

A major concern for traditional PBXs employed in an enterprise is the development of products developed by companies such as Skype and BlueNote Networks. Skype has not targeted the enterprise market and it does not support the SIP and SOA. The application developed by BlueNote is called the SessionSuite Architecture and does not require any hardware to be installed. It was a result of interest shown by the company as well as a prospective customer.

The SessionSuite Architecture consists of seven software modules that execute functions such as delivering the GUI. It supports SOA and could well enable the integration of enterprise applications with telephony. Industry watchers feel that if Skype has to grow in telephony it has to start supporting the SIP protocol soon.

D-Link to launch solution for SMBs

D-Link is launching a VoIP solution for the SMBs. The solution will have an IP PBX, trunk gateway, and a VoIP phone. The solution is called the D-Link xStack IP Telephony Solution for Small Business. The DVX-1000 is an SIP-based device offering functionalities such as voicemail, call hold, call forwarding, etc, it can support 25 extensions.

Conferencing is enabled by a web-based interface. Security is provided by an integrated firewall and an MD5 SIP authentication encryption coder software. The DVG-3004S SIP Analog Trunk Gateway connects to the PSTN and it has one LAN port and four RJ-11 ports.

The company feels that a pay-as-you-grow pricing structure is ideally suited to foster adoption by SMBs. The solution can be upgraded to a 25-user license. Another driver for adoption by SMBs is easy implementation, maintenance, and ideally round-the-clock support from the vendor.

Unlicensed mobile access

Sweden-based TeliaSonera has stated that it has successfully executed the first phase of its pilot of the unlicensed mobile access (UMA). The pilot was carried out in Denmark. The UMA consists of mobile and IP calls in a single solution. tmcnet.com reports:

Technical testing of the UMA concept is now also being carried out in Sweden in cooperation with Ericsson, the company said.

Read More: TeliaSonera says Danish IP telephony successful

ADSL VoIP in Italy

Teleunit S.p.A, which provides services such as fixed line telephony, wireless broadband, etc in Italy, has launched GoVoIP. The service is an ADSL VoIP service intended to address communication requirements of residential and SoHo markets. The service is available online and through resellers. GoVoIP enables users to make VoIP calls over an ADSL line without having to purchase extra equipment. The facility covers a range of bandwidths. ccnmatthews.com reports:

"The launch of GoVoIP is another example of our innovation, capitalizing on an existing technology to offer new and exciting services to potential new customers. VoIP now represents a viable alternative for SoHo and residential users to make economic savings by placing voice calls over the Internet."

Read More: Teleunit Launches GoVoIP to Further Strengthen Fixed Line Division

VoIP Telephony with Asterisk

The second edition of the book VoIP Telephony with Asterisk, written by Paul Mahler is now in the market. Readers can learn about the process of creating a reliable VoIP telephony system by using Asterisk and Linux-based computer hardware.

The open source software, Asterisk is a PBX that is interoperable with all standard telephony devices. Both American and European standards for signaling types are supported by Asterisk.

Knesset to deploy VoIP

The Israeli parliament, Knesset, will deploy an advanced IP telephony network provided by Nortel. The network will enable users to enjoy services such as long-distance calls, voicemail, and unified messaging. The deployment will also bring down the cost of operations for Knesset. Taldor, which is Nortel’s partner for this project will deploy 400 lines by the end of 2005. 1600 lines will be deployed in all.

The network will also have the Nortel Switched Firewall for protection against worms and viruses. Users will be able to access voicemail, email, and fax messages in a single mailbox by using either their phones, PCs, or any email enabled device. This feature is made possible by the CallPilot unified messaging product.

iGLASS Networks

iGLASS Networks and QoVox Corporation have come together in a strategic partnership that will lead to the integration of the QoVox Network Assurance System and the iGLASS Status Monitoring service. The companies intend to develop a “best in class” QoS product. tmcnet.com reports:

A major issue for service providers is maintaining high-quality services while minimizing the need to dispatch technicians to investigate and troubleshoot problems, commonly called "a truck roll," an often costly proposition.

Read More: iGLASS and QoVox Team to Assure VoIP Service Quality

MontaVista Linux CGE development platform

The MontaVista Linux CGE development platform by Artesyn Communication Products is compliant with the Service Availability Forum (SA Forum) Application Interface Specifications. tmcnet.com reports:

The KosaiPM modules can act as a control plane processor for optical and wireless infrastructure, and may also be used for augmenting packet processing and routing performance in voice gateways.

Read More: Artesyn Supports MontaVista Linux for KosaiPM AdvancedMC Modules

ICE

Microsoft Corp. has joined hands with Cisco Systems Inc. to incorporate Interactive Connectivity Establishment (ICE) to Microsoft software programs that will be released from 2006 onward. ICE will presumably help in solving the problem of NATs in firewalls. NATs change the IP address of a machine which makes VoIP communication difficult. ICE enables the VoIP software to know the new IP address of the machine.

RangeMax 240

Netgear is introducing RangeMax 240 routers and adapters that use MIMO technology developed by Airgo. The equipment is designed to increase the transmission speeds of wireless LANs to 240 Mbps. This allows concurrent use of VoIP, streaming audio, and video. The range and throughput achieved with this equipment is almost five times greater that that achieved by deploying the 802.11g wireless networks. It enables users to access data at high speed anywhere in the premises.

Businesses can use the RangeMax 240 to replace wired Ethernet networks and be able to hold a video conference anywhere in the office. According to Airgo, the True MIMO Gen 3 chipset is twice as energy efficient as the Intel Centrino chip. It consists of one processor and two integrated radios. MIMO technology employs multiple antennas at the two network points. This curtails latency and boosts bandwidth.

OfficeServ Wireless

The OfficeServ Wireless is a VoIP communications application developed by Samsung. It works with the company's iDCS 500, iDCS 100, and OfficeServ 7200 platforms.

A wireless line interface card and a wireless access point are some of the major components of the application. The application runs on the 2.4-GHz band and can support 240 wireless handsets concurrently. OfficeServ offers the advantage of mobility to its users and clarity of sound even during peak data traffic.

inContact™

UCN, Inc, which provides on-demand contact handling software and business telecom services over the UCN national VoIP network, will be providing its inContact™ to the SCO group for the management of inbound calls to the customer support teams. The SCO Global Services support clients such as NASDAQ, McDonald’s, and several SMBs. tmcnet.com reports:

"In terms of our call handling equipment, we had a huge, old Aspect ACD and a large Nortel PBX. Just to move the ACD system and re-program it for our new operation, we were looking at a $30,000 cost," explained Mark Colley, Director of Information Technology for SCO.

Read More: SCO Selects inContact(TM) From UCN for Global Service Department

IPtimize

IPtimize, which is a Managed VoIP Solutions Provider catering to SMBs will merge with Simmetech, Inc. tmcnet.com reports:

Upon closing of the proposed merger, which is expected to occur shortly, the continuing entity shall be known as IPtimize, Inc., a publicly traded entity (new ticker symbol to be determined) possessing core operations as a Managed VoIP Solutions Provider.

Read More: IPtimize to Merge with Simmetech, Inc.

IPTV system

Nortel has launched a new IPTV system that service providers can deploy. The system delivers services such as television and video by using the broadband network of the service provider. IPTV enables users to communicate by voice, video, IM etc by using their televisions. tmcnet.com reports:

"Nortel strives to provide more than a superior IPTV network for our customers. Our goal is to enable new and exciting IPTV applications that give our customers a competitive advantage to win and retain IPTV subscribers," said Walt Megura, general manager, Broadband Networks, Nortel.

Read More: Nortel Powers IPTV for Service Providers

CommuniGate Systems and Pingtel

CommuniGate Systems has launched CommuniGate Pro version 5.0, which is a solution that enables messaging and VoIP. The company also delivers the CommuniGate Pro version 5.0, which enables companies to host millions of subscribers. The server aims to integrate voice applications into the existing SIP infrastructure of a company. It performs functions such as call queuing and acts as an auto-attendant server. The solution enables the user to get services such as voicemail and email from a single mailbox.

Pingtel, which uses open-source code developed by SIPfoundry, has released the SIPxchange Enterprise Communications Server (ECS). The ECS includes an integrated presence server and support for additional SIP phones. Pingtel is making a conscious shift toward real-time communications and is looking to incorporate IM, video, and pure SIP applications as well. The company now has the architecture to manage independent components running on different servers.

SiteScape

SiteScape has now announced the release of Forum ZX, which brings together the features of Forum and Zon, which is a real-time collaboration package, developed by SiteScape. Forum uses a workflow engine that delineates the work, assigns tasks, reports the current status, etc. The workflow is menu-based and uses HTML forms. The workgroups can share and edit content in the context of the team; check-in/check-out control is used for this activity. The history of a document is recorded by the document management system and can become a part of the knowledge management system. voipplanet.com reports:

Fox went on to explain that with common presence-enabled applications, like IM or peer-to-peer VoIP, you accumulate a buddy list of people you talk to all the time, but when it comes to expediting business processes, who you talk to all the time usually isn't the right list to work from.

Read More: Synchronous and Asynchronous Collaboration

ASUREcall

OfficeMate Eyecare Business Solutions DBA Marchon Software Solutions, which is a subsidiary of Marchon Eyewear, will deploy the ASUREcall VoIP solution developed by NetWolves Corp. Marchon Software Solutions has already deployed the solution to link its head office in Irvine, California with another location in Ohio.

The ASUREcall solution consists of VoIP gateways, remote monitoring equipment, etc and allows international companies to minimize their telecommunication costs. The solution facilitates a smooth transition to a VoIP environment without disrupting the existing installations and in a cost-effective manner. The QoS delivered by the solution ensures uninterrupted toll-quality voice and optimum network performance. In fact, the QoS application is important for controlling the voice traffic over the data circuits. The application uses techniques such as bandwidth capping, flow queues, etc to manage the direction and breakup of network traffic.

NetWolves delivers a solution that is transparent to the user and enables the user to access the most up-to-date telecommunications technology without having to install software-based phones. The end-user does not require any special training. The monthly billing system helps to reduce the cost of ownership by eliminating variable expenses. The solution provides services such as voice and energy detection. It can support up to 1,000 users per gateway simultaneously and can be used with DSL, cable broadband, frame relay, etc.

Unified Core Network solutions from Nokia

Nokia has developed a range of innovative Unified Core Network solutions that facilitate fixed-mobile convergence (FMC). With the launch of its Push to talk over Cellular (PoC) solutions, Nokia is set to make a statement in the field of convergence. Nokia is known for providing core solutions such as push to talk and mobile softswitches. The Nokia MSC Server system is in use in more than 20 commercial networks by over 70 clients. The company has a major presence in the PoC market in GSM. 43 networks are using its commercial PoC systems. tmcnet.com reports:

"Nokia's portfolio of core network products and solutions offers operators one of the best fixed-mobile convergence propositions in the business. With a firm foundation in mobility and IP-based solutions as well as new offerings for Voice over IP and Unlicensed Mobile Access, our Unified Core Network is an ideal enabler for FMC."

Read More: Nokia Launches OMA-Compliant PoC and Presence at the Nokia Mobility Conference

Spanlink Managed Services suite

Spanlink Communications has developed the Spanlink Managed Services suite, which enables companies to manage their VoIP-based Customer Interaction Systems. The new suite which includes administration support and remote administration is on display at the Internet Telephony Conference & EXPO at Los Angeles. tmcnet.com reports:

Spanlink Managed Services "enable businesses to supplement their internal administration resources or eliminate the need for dedicated internal administrators, depending upon their business needs and technical competencies," noted the company's news release.

Read More: Spanlink Intros Managed Services for VoIP

Unified Messaging

SIP is facilitating the marriage of asynchronous applications like email with real-time applications. This enables unified messaging and helps companies stay connected. Interactive Intelligence, 3Com, etc provide unified messaging applications that are capable of working on a message before it enters the email system. The Find Me, Follow Me (FMFM) functionality checks for pre-configured contacts before moving on to the voice mail.

Unified messaging allows companies to prioritize their communication. The cost of these applications is not very high. The application provided by Siemens costs $ 80 per user per year. The cost of SIP phones are falling as well. informationweek.com reports:

Network Computing's poll paints an interesting picture of unified-messaging adoption. Only 17% of 686 respondents use unified messaging. But every one of our respondents from businesses with more than 5,000 employees say their companies have implemented it.

Read More: Get The Message Out

ShoreTel 6

The latest version of the distributed IP telephony platform by ShoreTel Inc, ShoreTel 6 stresses interoperability and security among other things. It is a suitable advanced voice platform for SMBs.

ShoreTel 6, which supports SIP, can be used with a range of telephony devices. ShoreTel 6 provides presence capabilities by means of Office Anywhere and encryption for heightened security. For $ 200 per user, ShoreTel 6 provides extension software, mailbox software, unified messaging, etc.

Networks have to use a central ShoreWare Server and the Jet database included with it in order to access the advanced telephony services. Business-class networks are assured of uptime and scalability by means of the distributed architecture used by the ShoreTel system.

VoIP at the racetrack

In the fourth quarter of 2002, Infineon Raceway was weighing its options regarding a new telecommunications system as the existing system did not provide services such as caller ID, voice mail, etc. According to Sara Grafals, V.P Finance, Infineon Raceway, the company was looking for a telecommunication system that could be installed before the racing season began. eweek.com reports:

What's more, Grafals said that her telecom consultants would soon discover that the track was paying $200,000 a year for track event phone lines, which were ordered for each event and then left in place long after the events were over.

Read More: Racetrack Wins with VOIP

Hammer Call Analyzer 1.6

Network assessment or prequalification may not be sufficient to check for voice readiness and quality throughout the life of the voice network. Hammer Call Analyzer 1.6, which has been launched by Empirix Inc., is intended to help organizations to assess voice quality on the spot and pinpoint voice-specific issues.

The software is available at $ 9,900 and a one-year subscription can be had for $ 1,950. A hardware-software solution is also available that supports hybrid TDM and IP voice systems. eweek.com reports:

We used Hammer Call Analyzer 1.6 vigorously during our tests of ShoreTel Inc.'s ShoreTel 6 VOIP solution, leveraging the tool to help isolate any signaling or call-quality issues we encountered as we deployed the ShoreTel network.

Read More: Empirix Checks VOIP Call Quality Over Time

FoIP

According to a research report published by Synergy Research Group, the Service VoIP market will be worth $ 2 billion in 2005. The growth of VoIP has firmly placed it as a viable option for mainstream telephony. As companies enter their upgrade cycles, they will migrate to IP telephony.

According to Morgan Stanley, a big share of business outlays is going to be increasingly directed toward VoIP. In the corporate sector, VoIP software and hardware sales will cross $ 11 billion by 2009. An estimate by AMI-Partners puts the spending by SMBs on IP telephony at $ 4.5 billion by 2008. In 2004, SMBs spent around $ 1.2 billion on VoIP, out of which $ 1.1 billion was spent on routers and SIP phones.

A survey conducted by Savatar in which 300 industry decision-makers participated yielded interesting information on the perception patterns in industry regarding VoIP. Reduced monthly bills were cited as a major attraction by 74% of the respondents. Reduced cost of ownership and simplified administration were cited as reasons by 73% and 68% of the respondents, respectively. Bundling of services, such as fax, with VoIP is also a good reason for 40% of the respondents for switching to VoIP.

Major telecom players have initiated moves that will result in reduced subscriber rates. Verizon, for example, is replacing its copper lines with optical fiber networks so that all products finally converge on to a single IP network. Analysts feel the VoIP adoption has gone mainstream. David Lankelevich, eMarketer.com, feels that VoIP will gain mass acceptance in the next two years.

The growth of VoIP is being fostered by factors such as quick resolution of the interoperability issues, increased broadband connectivity, etc. A single IP network for voice and data promises simplified management as there is only a single technology to understand; the reduced number of network elements is easier to manage with fewer IT systems required. A VoIP system helps in performing time-consuming and tedious moving and shifting tasks in a quick manner. In a TDM network, shifting a user while maintaining his extension number entails physical alterations in the network.

In an IP network, a user is not tied down by a physical connection to a specific port. User identification is achieved by the IP address of the phone. This allows users to work with their desk phones by simply plugging them into the LAN from anywhere in the office.

The hurdles to implementing VoIP include ensuring network capability to handle latency-sensitive voice traffic. This can mean expensive upgrades in terms of increasing WAN bandwidth and changing switches and routers. A study conducted by Nemertes Research in November 2004 revealed that the startup costs depended upon factors such as the size of the company and the IP vendor selected to provide the solution.

For a company with 100 users or less, the cost of deploying VoIP can come to $ 763 per person. The costs include IP PBXs and handsets as well as planning and implementation. The cost per user falls down to $ 525 for an organization with 1000 users or more.

Fax is an important service that can be bundled with VoIP. Fax offers advantages in terms of being compatible with several technologies, no changes in format, not editable by the recipient, etc.

The prospect of streamlined and integrated messaging services by deploying VoIP is of interest to a lot of organizations. 58% of the respondents, in a survey conducted by Empirix in February 2005, stated that they intended to run messaging applications on their converged networks.

Fax over IP (FoIP) is easy to deploy for companies that have an IP network in place. IP routers from companies such as Cisco, 3Com, Alcatel, etc are available with a built-in fax component (T.38). bitpipe.com reports:

For smaller organizations exploring VoIP, the big question is “how can we do this without breaking the bank?” As the market heats up there are a range of VoIP solutions with various price points. For such companies also looking to roll-out network fax functionality, a review of the actual cost of a VoIP system as well as a comparison of a boarded fax server versus one that is boardless should be added to the IT checklist.

Read More: Boardless Fax Servers in VoIP Environments

VoIP solutions by ACE*COMM

According to Dittberner Associates Inc. (DAI), the Voice over Packet (VOP) market is set to grow beyond $ 20 billion by 2012. The US, along with EU, China, and Japan will remain the major markets for VoIP. acecomm.com reports:

In Western Europe, the overall annual growth rate of 17.9% reflects different patterns in each major country, with Germany, France, and Spain clearly in the lead. The United Kingdom, Italy, and the Netherlands indicate slower growth, but according to Lilian Tau, VP of Consulting & Market Research at DAI, this is due to their having already made significant investments in VoP technology.

Read More: Proven Mediation Solutions for VoIP Environments

Sterling Internet Solutions

Sterling Internet Solutions first considered offering a managed VoIP solution to its clients in 2002. Its offering Sterling Voice has been available from April 2004. It is a managed VoIP service for SMBs. voipplanet.com reports:

"We saw the market evolving and migrating in that direction, but the product offerings that were out there were just blisteringly expensive and did not cater to a multi-tenant environment where you've got multiple companies with different needs and desires all managed and hosted by a single vendor," Gillihan said.

Read More:Sterling Internet Solutions

VoIP in government organizations

Even though revenue and competition are not the driving forces that government bodies are subject to, they too need to cut costs. Convergence of voice and data provides government bodies with the opportunity to reduce costs and reap the benefits of enhanced functionality.

Government networks run on highly efficient TDM networks that use the same compression algorithms as VoIP networks. The difference lies in the fact that unlike VoIP compression, TDM voice compression is not accompanied by any other overheads. VoIP calls do have the advantage of bandwidth savings due to silence suppression. According to studies as much as 62% of a voice call is silent.

In a TDM setup, the bandwidth is dedicated to a call at the beginning of a call. The IP overhead that leads to increased bandwidth and the reduction in bandwidth due to silence suppression evens out the bandwidth consumption in VoIP calls and makes it comparable to that of TDM calls. Frame packing is an effective technique used to reduce the header size of VoIP packets. It involves loading several frames of voice packet into an IP packet.

The frames can be loaded onto a single packet in two ways. One method is to add several voice frames from the same voice call. A drawback of this method is the limitation in terms of the number of voice frames that can be added; too high a number can lead to delay. Another method is to use voice frames from different calls that are taking place at a given time. This technique allows frames from 60 different calls to be present in the same packet.

By minimizing the extra bandwidth used due to the IP headers, the advantage of silence suppression is felt more keenly. The bandwidth utilization of VoIP calls reduces to half that of compressed TDM calls. However, the commercially used standards such as H.323 and SIP do not offer the facility of frame packing. A typical characteristic of voice traffic is the small size of the multitude of packets generated. 33 small packets of 50 bytes each can be generated every second with VoIP calls. In contrast, a data packet has a maximum MTU size of 1500 bytes.

In order to maintain the QoS of voice services, voice traffic is differentiated from data traffic by DiffServ. STU-III and STU-IIB are standards used by U.S government agencies and NATO respectively, for the purpose of securing voice communications. The voice call is transferred as encrypted data over a modem. Modem calls do not support speech compression and therefore PCM (64 Kbps) is used for transferring them. Use of differential waveform coding (ADPCM) can reduce this to 32 Kbps but it impacts the modem transfer rate.

The problem of maintaining bandwidth efficiency as well as the security of the call can be solved by not using ADPCM and terminating the modem signal at the entry to the network. By extracting and transferring only the modulated data from the signals by means of a Secure Call Relay, bandwidth consumption can be managed. A buffer can help smooth jitter to a large extent with a minor delay in traffic. However, this is not possible with secure modem calls as delayed signals can lead to the termination side reconstructing the data sent by the originating side in an incorrect manner.

Error correction is utilized for prompt correction of corrupt VoIP packets without retransmission. Error correction is particularly useful with secure calls. A satellite hop can introduce a 250 ms one-way delay in a voice call and a half-minute round trip delay. Delay also increases the chances of the secure modem not remaining in sync. If satellites, such as INMARSAT, are being used in the government voice network, a BRI data interface can be used to connect the satellite transmitter to the network equipment.

Implementing VoIP in government networks can often mean having to deal with legacy systems that may have to be supported as well. The PBXs run on CAS protocols and applications such as MLPP may require support.

Cisco CallManager Version 4.0

Nonverbal communication is an important aspect of any conversation and can account for up to 60% of the communication. Therefore, it is important that video be made use of to supplement audio communication. The promise of convergence made by IP networks is set to take video out of the boardrooms and make it available on the desktops.

The Cisco CallManager Version 4.0 is an IP-based system that facilitates video telephony. Cisco IP phones have gained the functionality of video telephony due to CallManager Version 4.0 and Cisco VT Advantage. The IP phones can be used to add real-time video telephony to a call in a transparent manner. The CallManager Version 4.0 is software-based and along with the VT solution enables video conferencing at an incremental cost of less than $ 200 per seat. cisco.com reports:

Cisco VT Advantage works with Cisco’s midrange and high-end IP phones, including the 7940G, 7960G, and 7970G Cisco IP phones. Video endpoints are configurable from 128 Kbit/s for low-resolution video, to 4.5 Mbit/s for broadcast-quality displays.

Read More: Communicating in an IP World

PLC

The technique of Packet Loss Concealment (PLC) is used to cover the impact of discarded packets. It is useful in situations where the number of packets lost is less, amounting to not more than 20-30 milliseconds of speech time.

The degradation in voice quality occurs primarily because of bursty packet loss that can go on for several seconds and result in a loss of up to 30% of the conversation.

PLC employs algorithms to either replay the last received packet or generate speech using previously used speech samples. The bandwidth consumption increases with the increase in the sophistication of the algorithms. This reduces the capacity of the gateway.

Managing VoIP voice quality

The voice data in packetized voice communication systems, such as VoIP, frame relay, and ATM, is digitalized and lossy compressed. The voice data for a given unit of time, for example, 30 milliseconds, is represented by a frame. The compressed frames that are transmitted across a network are decompressed at the destination.

The header size of a voice data frame has a minimum size of 20 bytes. The User Datagram Protocol consumes another 6 bytes. This means that if a voice frame is encoded using the G.723.1 algorithm, it is 24 bytes long yet the total packet length is of 50 bytes. The additional 26 bytes are header-information. This means that only 48% of the bandwidth is utilized in an effective manner.

In order to manage the latency, the voice packets need to be very small in size. An increase in the number of VoIP packets can constrain the routers with respect to their bandwidth and processing abilities. A voice call that is compressed with the G.723.1 codec at 6.4 kbps generates 33 packets per second, which can overwhelm a Cisco 2500 router.

In order to increase the payload for a header size, multiple frames from a call can be grouped into a single packet. However, this method has the drawback that it leads to an increase in latency. SHOUTIP™ open telephony platform uses a system of frame packing that increases the efficiency of bandwidth utilization without increasing the latency.

Interlink Global

Interlink Global, a Florida-based VoIP provider obtained approval for its CLEC license from the Florida Public Service Commission in the first fortnight of September. This should allow the company to price its data infrastructure more competitively as it can now negotiate rates directly with Bell South instead of approaching it through a third party. voipplanet.com reports:

InterLink is employing both acquisitions and partnerships to expand its international presence. In May of this year, the company announced the formation of a partnership with CosmoTelco of Athens, Greece. In July, InterLink announced that it had acquired Venezuelan telco NGTV for $6 million, followed last month by the formation of a subsidiary in Ecuador.

Read More: InterLink—a Global Niche

Siemens and Genesys come together

Siemens and Genesys have introduced a new SIP-based IP call-center solution which integrates the Genesys 7 set of call center applications and Siemens HiPath 8000 IP system. The solution has been positioned as a centralized platform for communication solutions aimed at large enterprises.

According to Jonathan Zaremski, Product Manager, Genesys, the Genesys SIP Communication Server has enabled the development of a software-driven contact-center solution. HiPath 8000 and the Genesys suite are integrated at the SIP server. In the past, efforts at implementing an enterprise-wide call-center solution have been hindered by the absence of scalable IP solutions that are based on open standards.

This solution offers the advantage of easy user addition, thousands of users can be added without an increase in the number of agents to man the resources. There is no need to add a PBX at every site and maintain proprietary hardware. The solution also facilitates easy migration from a TDM to an IP network. It also enables the running of TDM and hybrid TDM/IP networks simultaneously.

The company can plan its migration in a manner that allows it to get the best ROI out of its legacy systems and not be hurried into it by the infrastructural demands of the new network. Other benefits of the SIP-based solution include excellent disaster recovery and savings in administration costs.

Cisco solutions for SMBs

In the third week of September 2005, Cisco Systems announced its new Cisco Business Communications Solution, which is targeted at the SMBs. It is an end-to-end solution that incorporates all the standard features and financing options as well. Cisco Unity, which executes the auto attendant function, and CallManager Express version 3.3, which handles conferences for up to 96 users, are utilities included in the IP Communication Express suite.

The Cisco Network Assistant (CAN) 3.0 includes GUI-based web tools that assist SMBs in experiencing VoIP convergence. The early adopters of IP technology in the SMB sector are realizing the benefits of reduced costs, better efficiencies, and improved customer management skills. Cisco offers the SMBs the same products that its enterprise customers purchase. The SMBs get a high level of security and features that have been customized to their requirements.

W-Series systems

According to a study by Frost and Sullivan, VoWLAN operators are set to experience a CAGR of close to 160%. The major factors governing the acceptability levels of VoWLAN include voice quality and interoperability. 802.11 wireless devices like VoWi-Fi are tested using tools such as the W-Series developed by Azimuth Systems. voip-news.com reports:

W-Series systems provide the ability to configure an entire WLAN network in a bench top chassis designed for complete Radio Frequency (RF) isolation and control. The flexibility of the W-Series allows for the thorough evaluation of wireless LAN equipment under varying mobility and traffic conditions, as well as precise analysis of the results.

Read More: Voice over Wi-Fi Voice Quality Assessment Test

LogiSense Corporation

LogiSense Corporation has released an enhanced version of their EngageIP VoIP billing solution that automatically executes the rating, provisioning, etc of VoIP services. voip-news.com reports:

“CLEC, ILECs and ISPs today are faced with the challenge of bill presentment to their customers that may be in different geographical locations, taking advantage of limited time marketing offers or simply using bandwidth from multiple carriers,” said Flavio Gomes, president and CEO of Logisense.

Read More: LogiSense announces EngageIP VoIP Billing and Rating solution

Echo Cancellation

Sangoma Technologies Corporation has filed a patent application titled "Echo Cancellation Controller", which deals with the methods of measuring the echo on incoming voice streams and the proper control of the echo canceller. voip-news.com reports:

Sangoma's EDAC (Echo Detection and Control) is an algorithm that examines each call as it is connected, and within about one second, determines whether the call has echo or not. It then enables or disables the echo canceller as necessary.

Read More: Sangoma Technologies Registers Patent Application on Echo Detection and Control System

Open source VoIP products

Open source VoIP products include

• sipXpbx, which is an SIP PBX aimed at SMBs.

• sipXregistry, which is an SIP registrar and redirect server.

• sipXpulisher, which is a subscribe/notify server for SIP event subscriptions.

• sipXcal, which is a call processing library.

• sipXvxml, which is a voice processing engine. eweek.com reports:

The first open-source release under SIPfoundry will be Pingtel's source code for its SIPxchange IP PBX and Instant Xpressa soft phones, under the GNU Lesser General Public License.

Read More: Push on to Make VOIP Open Source

BCM1161

The new integrated VoIP processor chip announced by Broadcom, the Broadcom® BCM1161, is a second generation VoIP processor that consumes low power and provides advanced multimedia functionalities such as a 2 megapixel digital camera, voice and video record/playback, etc. The BCM1161 also allows conferencing and high-fidelity voice capability. It has a single chip that integrates the direct microphone and the speaker interface with the analog voice codec.

SurfUP™ver4

On September 22, 2005, SURF Communication Solutions® announced the launching of SurfUP™ver4 in the market. The product is a universal port solution that supports video transcoding, conferencing, recording, and also has video toolbox capabilities that enable resizing, frame rate change, inserting logos, etc. SurfUP™ver4 offers DSP chip–level solutions as well as DSP–farm resource boards.

The version 4 runs the different media types on one DSP, which is TMS320C64x™ generation by Texas Instruments. The DSP provides scalability to the solution and helps in reducing the Host-DSP bottleneck by supporting UDP/IP, RTP, and H.323. Telecom manufacturers can integrate user-defined channels with the help of the open DSP framework that Surf provides.

As Surf runs voice and video on the same DSP, it provides quality in a cost-effective manner. The product supports dynamic speaker detection, gateway applications, and CTI messaging and recording applications. Moreover, cPCI and ATCA carriers will provide an integrated solution for the Surf PTMC/AMC daughter cards.

VoIP affects network performance

According to a survey by Enterprise Management Associates that covered 100 companies, it was found that VoIP performance was affected by the fact that it competed with data applications for bandwidth. Close to 90% of the respondents who were polled in the survey felt that having VoIP performance monitoring capabilities was critical in ensuring smooth functioning of VoIP. The survey reinforced the thought that although VoIP is being rapidly adopted by industries, its scalability can be hindered by the absence of management policies to regulate the effect of VoIP and data communications on network performance.

According to Jim Vale, Product Manager, Network General, most companies know that they have to fulfill rigorous performance criteria to ensure real-time VoIP transmissions. Yet, they do not seem to grasp the significance of the effects that VoIP implementations can have on the network. This can handicap them in their efforts to manage "mission critical" applications that run on the network simultaneously with VoIP. The fact that VoIP is not like other applications that run on TCP/IP and its transmission requirements are different makes VoIP deployment a slightly tricky issue. VoIP transmissions are high priority and have poor tolerance for dropped packets and retransmissions.

Network General and Fluke Networks are two companies that are working toward developing network performance tools that can be integrated with the regular network management activities as a part of the system. Network General has focused on the L2-L7 protocol analysis and provides tools for application-level analysis. The VoIP Lifecycle Solution offered by Fluke Networks manages and troubleshoots voice as well as data traffic.

Packet loss due to burstiness

Packet loss in IP networks happens in bursts and there are various models that analyze the cause for packet loss and bit errors; usually it is congestion in the network and jitter.

Bernoulli Model: It is a popular independent loss channel and assumes that packet losses occur with a probability Pe. Therefore, if the number of packets in the network is N, the number of packets lost is N.Pe.

Gilbert Model: It is a well-known and widely used burst model. The model has two states, these are a gap state with a value 0 and a burst state with a value 1. The gap state is a zero loss state and the burst state is referred to as a lossy state.

Markov Model: It is a multi-state model and the system keeps oscillating between states. Short term dependencies between lost packets can be studied with help from the 2-state Markov model. It can be combined with the Gilbert-Elliott model to understand consecutive losses as well as the longer events that occur due to link failure and can last for more than 10 seconds.

VoIP quality suffers due to bursty packet loss even if the loss rate is as low as one percent. This is because the packet loss occurs in short and dense bursts that leads to degradation of sound.

Carrying voice over frame relay, IP, and ATM - Part 2

Voice over Internet Protocol (VoIP) uses IP, which is a connectionless protocol. IP facilitates efficient bandwidth allocation as packets do not follow a preallocated path between endpoints. Paths that are open and are less congested can be used for transmitting the packets. To ensure a high QoS level, it is preferable that the packets get transferred on the same path. Headers used in IP traffic consume bandwidth because of their size, they can be up to 20 bytes, headers in Frame Relays and ATM cells are of 2 bytes and 5 bytes, respectively. The headers contain information that ensures the arrival of the packets at their desired destination as well as in rearranging the packets at the receiver's end.

Fragmentation, jitter buffering, prioritization, voice compression, silence suppression, and echo canceling are some of the methods used in an IP network to increase bandwidth efficiency.

Prioritization: Prioritization is closely linked with QoS. At present, there is no widely accepted QoS standard for IP services. RSVP was an IP QoS protocol under which a sender could try and obtain permission to dispatch his data in a particular manner. It has led to the development of the Differentiated Services Model that uses Type of Service (ToS) to determine the type of traffic at the gateway between the user and the service provider.

Fragmentation: It is carried out to minimize the delay of voice traffic and is performed in a similar manner as in Frame Relay. However, this leads to an increase in the number of IP headers, which means that IP voice traffic may require up to 50% more bandwidth than Frame Relay voice traffic. Improvements in header compression and router technology should help in minimizing bandwidth consumption in IP.

Voice compression: As voice traffic usually travels over links that do not have a very high speed, for example VPNs at many SMBs run only at 28.8 kbps. The ITU G.723.1 standard supports voice compression over IP for dial-up modems and ensures toll quality voice.

Jitter Buffers: These store the packets that arrive so that the delay in the variations is minimized. The setting of the buffer can affect the quality of the conversation. The maximum size of an adaptive jitter buffer can go up to 100 ms to 200 ms. The ideal size is between 30 ms to 50 ms.

Echo Cancellation: Echoes occur because of a mismatch in the impedance in the circuit-switched network or a faulty coupling between the microphone and the speaker of a telephone. VoIP networks can face greater delays than circuit-switched networks and consequently require better echo cancellation techniques. G.165 and G.168 are some of the specs recommended to counter echoes.

Silence Suppression: It is also known as Voice Activity Detection (VAD) and implies the ability to refrain from sending audio packets on an RTP stream during the silent periods, which include the pauses between words and the natural pauses in a conversation. Silence Suppression can help in reducing the bandwidth requirement by 10%.

Voice over ATM: Asynchronous Transfer Mode is an ITU-T standard that lays down the specifications for cell relay of information such as voice, video, etc. The information is relayed in small cells of a fixed size. The technology has the advantage of being high speed and scalable. However, it is an expensive technology. ATM is being increasingly used by corporates to transfer large amounts of voice, graphics, and other such information. ATMs use fixed-size cells that consist of 53 octets/bytes each. The cell consists of a header and a body, with the header consuming 5 bytes and the body taking up the remaining 48 bytes. The small packet size makes ATM suitable for transferring voice and video data as these data types require a steady flow and large data packets take time to download.

Fragmentation: The ATM network uses high-speed switches to run the data through its course. This is possible primarily due to the fragmentation that is built-in into the network. ATM networks use high bandwidths that help in minimizing congestion problems and ensure reliable delivery of data packets. This helps the ATM providers in delivering a high QoS.

Prioritization: VoATM follows the standards laid down in the Adaptation Layer 1 (AAL1) protocol as per the Constant Bit Rate (CBR) service. CBR provides Circuit Emulation Services (CES) that transmits a continuous stream of information; this enables the network to apportion the desired bandwidth to a connection for the transmission period. However, as with circuit switching technology, the voice quality that comes from a regular transmission comes at the price of efficient bandwidth utilization. On occasions, CES can transmit semi-filled cells instead of waiting for the cell to fill. This can lead to wastage of up to 20 bytes of bandwidth per ATM cell. Dynamic Bandwidth Circuit Emulation Service (DBCES) is similar to CES except that it transmits only when the receiver is off the hook.

A Variable Bit Rate (VBR-RT) service as specified by AAL2 is the accepted standard for VoATM. Packets of size 1 to 64 bytes can be transmitted by following the AAL2 standard. These packets are also known as minicells and can be incorporated into an ATM cell. AAL2 supports a variable payload, which helps to improve bandwidth efficiency. AAL2 also supports voice compression and silence suppression and enables multiple voice channels over one ATM connection.

Interoperability between networks: Achieving interoperability between the various networks will allow users to benefit from the best that each network has to offer. ATM offers a high QoS and a good speed, Frame Relay provides an installed base, and IP has a global reach. The extent of compatibility is limited by the prioritization methods and signaling protocols, even though these networks follow similar fragmenting techniques. The level of interoperability will increase with the introduction of standardizations within the protocols, which will facilitate the interworking.

Currently, the Frame Relay Forum has set standards for transmitting voice over Frame Relay; however, there are no standards for voice switching between VFRADs. This has led to the development of proprietary solutions that limits interoperability between the products of different vendors. The use of Switched Virtual Connections (SVC) would entail that paths are defined dynamically, this would increase the scope for interoperability between different solutions.

The interoperability standards for voice and multimedia over IP are defined by ITU H.323. These include endpoint negotiation and the format of the information but not issues such as encoding and security. Also, given the fact that the definitions as given in H.323 can be interpreted in more than one way, a guarantee of interoperability between the products provided by different vendors can not be given. Efforts are underway to provide interoperability between the gateways and gatekeepers provided by different vendors, this is being done by creating an interoperability profile using H.323 and H.225 Annex G standards.

In the absence of standards for these networking technologies, the interworking solutions of the near future will be proprietary. This entails that the users be aware of the technological aspects and that interoperability issues be made transparent to the users. Situations in which an interworking of technologies is desirable include corporate networks that run on Frame Relay and need to communicate with a remote location, their problem can be solved by implementing VoIP, without the need to install a Frame Relay infrastructure. The ability to use multiple voice technology over the same platform also means that migration to another technology need not mean a loss of investment. A new product being developed by RAD will facilitate VoFR - VoIP signaling conversion. This interworking between Frame Relay and IP should be advantageous as the ubiquity of IP services increases.

Carrying voice over frame relay, IP, and ATM - Part 1

Voice can be transferred over frame relay, IP, and ATM. The rate of growth of data networks has been greater than that of voice traffic, primarily due to the increased availability of broadband. This has led to the phenomenon of voice being sent more regularly on data networks. Voice over Frame Relay, VoFR, has been developed from the Frame Relays that were developed in the 1990's but were not suitable for carrying voice.

The growth of the Internet fuelled the demand for networks that would enable carriage of voice data in an inexpensive manner. Frame Relay, IP, and ATM are examples of packet networks designed for carrying voice and data, thereby satisfying the market need for a universal and cheap convergent technology. The next step is to improve the integration standards for the ubiquitous delivery of voice over Frame Relay, IP, and ATM.

Frame Relay, IP and ATM differ from PSTN, which is a circuit switching technology and has been specifically designed to carry voice. ATM differs from Frame Relay and IP in the sense that it breaks the data into small cells, this helps to speed up the data switching process in the network. Cell switching technologies can perform dynamic allocation of bandwidth depending upon their activities, this is known as statistical multiplexing. There is no reservation of bandwidth for a particular use, at any given time a bandwidth may be allotted as per the requirements of the network. As opposed to to cell switching networks, in a traditional circuit switching network, a line is dedicated to a call for the duration of the call and even when the call is on hold. This has the advantage of a consistent voice transmission but the bandwidth utilization is inefficient as the dedicated line cannot be used for other data transmissions even in the absence of voice transmissions.   

The packet switching networks were originally developed to handle traffic that moved in bursts. This means that packet switching networks are by default not as efficient as circuit switching networks as far as handling voice is concerned. The shared nature of a packet switching network implies that the delay in voice packet transmissions is erratic and not as less as desired for achieving a satisfactory quality of voice. Voice transmission should ideally be an accurate representation of the speaker's tone, inflection, etc. Delay in the delivery of the voice packets can lead to a communication gap, which may be further accentuated by packets being dropped along the network path.

Delay in the network can be reduced by increasing the bandwidth but it is an expensive alternative and the benefits of a shared bandwidth cannot be availed. Ideally, traffic congestion should be managed at the customer's end and at the backbone by prioritizing the flow of traffic. This had led to the development of smart access equipment for implementing procedures that help in reducing packet loss due to network congestion.

Voice over Frame Relay (VoFR): Frame Relay finds application in data networks in companies as it offers flexible bandwidth, easy accessibility, and a mature technology that supports a variety of traffic. Frame Relay offers the advantage of a predictable performance, it runs on the principle of Permanent Virtual Connections (PVCs) and is well suited for star configurations and closed user groups. Voice Frame Relay access devices (VFRADs) connect the router, SNA controller, and the PBX to the Frame Relay Network. This helps in achieving the integration of voice into the data network. MAXcess is a VFRAD that helps to surmount the difficulties in transmitting voice data over the Frame Relay without having to increase the bandwidth. It does so by employing the techniques described below:

Prioritization: VFRADs mark applications as per their reactions to delay. Voice and other time-sensitive data like SNA is given higher priority. Since voice transmissions are short and do not require much bandwidth, they do not have a detrimental effect on data traffic; in fact, they can be sent alongside other information like emails, graphics, etc that travels on the network. Different QoS packages are provided by the Frame Relay service providers. Clients prefer to purchase the best QoS for voice and SNA traffic; Non-real time variable frame and a slightly lower QoS, it is preferred for a LAN network and Internet connections. VFRAD can also be set to recognize traffic that can be dropped in case of network congestion. It achieves this by utilizing the Discard Eligibility bit. 

Fragmentation: VFRADs have the capability of fragmenting data packets to allow voice data higher priority in terms of transmission even if it leads to stopping the other transmissions. However, increasing fragmentation leads to reduced bandwidth efficiency due to an increase in the number of data frames. Applications such as RAD FR+ make it possible to send complete data frames and fragmenting occurs only if the voice data reaches a switch in the midst of a data transmission.   

Fluke Networks offers VoIP Management Solution

Flukes networks announced that it has prepared a comprehensive VoIP lifecycle management solution. It will help the network managers to deploy, monitor, troubleshoot and manage VoIP networks. Fluke network's VoIP lifecycle management solution will manage the VoIP infrastructure for planning a future growth. It will act as a cost-saving formula and enhance competitive advantage through the new productivity tools.

Users expect IP phones to be more reliable than the voice-dedicated landlines. They also expect the VoIP systems to have high quality and performance standards. The network should provide a flawless performance in the ever-changing network environment. To minimize all the problems, Fluke networks has made a VoIP lifecycle management solution. primezone.com reports:

This lets IT managers verify before deployment that the infrastructure can support VoIP. It also allows the thorough examination of all system elements during deployment and management of the VoIP system proactively after deployment, including ongoing monitoring, troubleshooting, and planning for future growth.

Read More: Fluke Networks Offers VoIP Management Solution for the Entire VoIP Lifecycle

NETGEAR selected Centillium to produce VoIP Platform

NETGEAR has selected Centillium's newest Atlanta(TM) system-on-chip (SoC) solutions to produce its latest VoIP platform for the consumers and small-scale business market. Centillium Inc is a leading provider of broadband access solutions. Its Atlanta series includes four unique devices and easy-to-use application software. It will enable equipment manufacturers to save development time and preserve software investments across multiple platforms.

NETGEAR is committed to provide its customers the most advanced solutions. Atlanta's high performance routing and excellent voice quality will be proved beneficial for the customers. The Atlanta series are built on Centillium's Voice Services Platform (VSP). It is to be noted that VSP is an award-winning platform. home.businesswire.com reports:

The Atlanta family's innovative application software is field-proven to meet the demanding characteristics of carrier-class networks, and has been designed to be interoperable with the most widely deployed soft switches and leading VoIP service provider networks. Most importantly, its unique set of software features offers turnkey functionality for "plug-and-play" installation and service activation, allowing equipment manufacturers to start providing complete VoIP product solutions within as little as three months.

Read More: NETGEAR Selects Centillium to Speed Next-Generation Consumer VoIP Platform to Market

H.323 protocols and their applications - Part 2

H.450 is a set of standards similar to the QSIG standards that detail services for ISDN networks. H.450 defines the Supplementary Services for the H.323 protocol. These include Call Transfer and Call Diversion. The H.450 protocol works with the H.225 protocol and its messages do not have a header.

H.450.1: It relates to the procedures for controlling the supplementary services between H.323 applications. It elaborates a signaling protocol that is common to all H.323 supplementary services and is based on the generic functional protocol for Private Integrated Services Network (PISN).   

H.450.2: It is based on H.450.1 and is a Call Transfer supplementary service in H.323 networks. This protocol enables a user to transfer his call with another user to a third user without having to establish a call with the third user.

H.450.3: It is based on H.450.1 and is a Call Diversion supplementary service in H.323 networks. Call Forwarding Unconditional (SS-CFU), Call Forwarding Busy (SS-CFB), Call Forwarding No Reply (SS-CFNR), and Call Deflection (SS-CD) are the services that are included in this protocol. These services are applicable during call establishment and enable the diversion of a call to another specified endpoint, which may be a voicemail or a cell phone number. The reason for the diversion, such as unavailable or busy, is specified by H.450.3 to the destination endpoint, thereby allowing it to respond accordingly.

H.450.4: It is based on H.450.1 and is a Call Hold (SS-HOLD) supplementary service in H.323 networks. It enables the interruption and re-establishment of communication between users. SS-HOLD is applicable to the audio as well as video data streams. The caller can put the receiver on hold to carry out other activities and at the other end, the other person can too initiate another call without disturbing the held call, if he so wishes.

H.450.5: It is based on H.450.1 and is a Call Park (SS-PARK) supplementary service in H.323 networks. It enables a user (Parking User) to park a call (parked to endpoint); it results in the parking endpoint achieving idle status. The Parked User experiences filler music, video, or images while he is parked. A user can pick up a parked call or an alerting call using Call Pickup. Every authorized user of the H.323 network can avail this supplementary service irrespective of the gatekeeper zone.

H.450.6: Call waiting (SS-CW) operates when an endpoint is busy with another call or another application, such as emails. The caller is made aware of the endpoint being busy and has the option of either ending the calling or leaving a message waiting callback. The potential receiver can accept, reject, or ignore the call. SS-CW occurs when all other options such as active, waiting, etc have been exhausted.

H.450.7: It deals with Message Waiting Indications that may be voicemail, fax, teletex, etc.  It is a general purpose mechanism in which a Message Center notifies the Served User at whose end A Message Waiting lamp lights up. Message related information such as the type of message, subject, and relevance can also be highlighted. Automated message retrieval and a callback request are possible in an H.323 environment.

H.450.8: It deals with Calling Party Name Presentation in which the receiver gets to see the name of the caller. The calling endpoint or gatekeeper provides the name of the caller for gatekeeper routed calls. The gateway obtains the name of the calling party from the switched circuit network and passes it to the packet network.

H.450.9: It elaborates on the Completion of Calls to Busy Subscribers (SS-CCBS). SS-CCBS notifies a caller if the receiver is busy, the receiver's endpoint can monitor activity at its end and inform the caller when it is free, upon receiving the information, the caller's endpoint then attempts to complete the call.

H.450.10: It is a Call Offer (SS-CO) supplementary service that allows a calling user to wait till the receiver reacts to the call. The receiver can accept the call after the resources become available to him. The receiver can ignore the call offered or try to make resources available by releasing or placing on hold other calls.

H.450.11: A served user can interrupt an established call by invoking Call Intrusion (SS-CI). A Call Intrusion results in the third user either being held, invited to a conference call, or force-released.  The options available with a Call Intrusion depend upon the level of authorization with the served user. 

H.450.12: It deals with the ANF-CMN service that allows for the exchange of Common Information, such as Feature Identifiers, Feature Values and Feature Controls. This information can serve as a foundation for the indications to the local user or for filtering requests. ANF-CMN endpoints can receive solicited and unsolicited services that can be offered as a combination.

Infozech allied with BroadSoft to offer VoIP Solution

Infozec software Inc. has integrated its offerings with BoradSoft's BroadWorks platform. Infozec is a leading provider of telecom billing and settlement solutions. The integration will allow the two companies to provide an infrastructure for a VoIP service provider in Mexico, Telco. Infozec's offerings have the ability to introduce new and innovative services in real time.

Infozec's solution for Telco consists of various modules. These include mediation, customer management, order entry, credit control and accounting. I will allow its subscribers to make online payments and access other Web self care options. The integration of BroadWorks and and Infozech's Billing and Settlement server will satisfy both the prepaid and post-paid billing needs. businesswire.com reports:

The BroadWorks VoIP application platform predominantly uses an open client interface; by virtue of which, service providers and solution vendors can seamlessly integrate their solutions and deliver best-of-breed services without the need for large customizations. BroadSoft's interoperable platform, coupled with Infozech's highly reliable, scalable, platform agnostic and secure products, give users the flexibility to add new and innovative services rapidly and easily.

Read More: Infozech Offers an Integrated Solution for VoIP with BroadSoft

Nortel brought VoIP to the Rural North American Market

Nortel announced the successful deployment of its next-generation, SIP-enabled DMS-10 platform. It is seen as a step to bring VoIP into the rural North American market. This solution is designed to expand revenue opportunities for the rural service providers. The rural service providers now will be able to offer services enabled by Session Initiation Protocol (SIP) such as VoIP at a low cost.

Nortel's rural VoIP solution gives DMS-10 subscriber the option to use traditional or VoIP primary phone service. It also allows user to add the cost-effective VoIP secondary phone service. Subscribers can also take the advantage of VoIP mobility capabilities. This will allow them use their phones while travelling provided they have broadband access. lightreading.com reports:

Nortel's DMS-10 platform allows independent service providers and rural market carriers to seamlessly advance their networks to a new, cost-effective packet infrastructure at their own pace without expensive upgrades to the network. With the new SIP-enabled enhancements, service providers now have the capability to offer new revenue generating services such as primary or secondary line service over any broadband facility, mobility options with IP phones and clients, and service bundles such as VoIP coupled with the subscriber's existing broadband or long distance service.

Read More: Nortel Deploys Rural VOIP

Cable One selected Sigma system's VoIP solution

Cable provider Cable One selected Sigma Systems as its OSS vendor to deliver new PacketCable telephony to its subscribers. Cable One currently has over 700,000 subscribers. The reason behind choosing the Sigma Systems based on the fact that it has better infrastructure and long experience in providing OSS installations for VoIP. By choosing Sigma, Cable One has scored a point over its rival cable operators. Sigma's OSS solutions for VoIP and broadband IP services are the most trusted ones in the cable industry. Sigma is confident that its VoIP Service Package and other features will enable Cable One to easily deploy new next-generation services to subscribers.

Cable One will use Sigma's Service Management Platform (SMP) and VoIP Service Package to provide services like dial tone, voice mail, long distance calls etc. Sigma has already proved itself in the field on VoIP deployment across the globe. tmcnet.com reports:

The role of OSS service management is essential as we deploy VoIP services, scale operations, and look to grow revenues for our subscriber base, said Steve Fox, Cable One's vice president of digital services and technology, in a prepared statement. Using Sigma's solutions, we gain operational efficiencies by significantly reducing the manual work and processes associated with service order management and service provisioning for complex VoIP services.

Read More: Cable One Chooses Sigma Systems' OSS VoIP Solution

Managing your migration from a WAN to a MPLS

Multi-Protocol Label Switching (MPLS) offers several advantages, these cover cost, scalability, and reliability issues. MPLS enables better network management and tracking down of malware and unwanted traffic and server / client and application related issues can be pinpointed more efficiently. Before initiating a VPN / MPLS migration, a company needs to look into the bandwidth consumption of the applications, the number of ports that the firewalls will manage, the effect of the transition on the responsiveness of the network, and tools to measure the performance of MPLS. Network Physics offers a solution for better network application management. It offers baseline performance measurement, real-time enterprise-wide visibility of data centers and NOCs, boosts bandwidth by eliminating rogue traffic and removing viruses, facilitates capacity planning and tackling response time issues.

Assessing VoIP quality in an enterprise-wide deployment

Assessing the readiness of a network prior to a deployment can help in minimizing training costs and obtaining better results with the pilot deployments. Vivinet Assessor is one such software that helps in predicting a network's readiness for VoIP. VoIP demands unique network requirements that may entail network upgrades; an assessment report pinpoints the areas that need to be upgraded. The Vivinet Assessor uses a Mean Opinion Score (MOS), which is derived from the G.107E model, in order to provide an accurate feedback on end-user perception of the call quality. 

Linux Meets VoIP

It's like watching a small child grow up right before your eyes.  First it starts walking, the parents make rules for its behavior, and then it graduates to bigger and better things.  Well, I think that this comparison fits quite well with VoIP.  As for graduating you may ask?  Well, VoX Communications has just unveiled that they have created the first ever Linux based server cluster.  The company boasts that the server cluster can hold 10,000 customers, but is hopeful with additional clusters that the number could grow to a several million.  Finally, something really stable we can all sink our teeth into.  According to ZDNet:

"The equipment cost in this initial deployment was less than $100,000, and we expect the equipment required to support each incremental 10,000 subscribers to be less than $100,000," said Paul Riss, CEO of VoX.

Read more at: First-ever Linux-based VoIP server cluster eyed

VoIP & Verizon Strike A Deal

Today is the day that Vonage and Verizon are really working together.  Although they have been having fun testing their 911 system in NYC, they are now moving forward with a "territory-wide" implementation of an emergency system geared for VoIP.  Why territory-wide you may ask?  Mainly because Verizon is only located in 28 states, they are limited to those areas and cannot extend without further assistance.  Although it might be a shortfall at the current time, when their methods are tried and tested, you can expect other phone companies to come forward and offer their assistance to VoIP providers.  Hopefully the system works well for the consumers’ sake, and for the corporate heads that have to square off with the American government.  According to TMCnet:

"Verizon is a responsible steward of the E9-1-1 public trust, through their foresight Vonage is able to implement an E9-1-1 solution that will serve all customers," said Jeffrey A. Citron, CEO of New Jersey-based Vonage.

Read more at: Verizon and Vonage Tackle VoIP E911 Dilemma

Telchemy Expands VoIP Troubleshooter Web Site

VoIP performance management software provider Telchemy has redesigned and expanded their VoIP troubleshooting Web site, available at VoIPTroubleshooter.com.

According to Boardwatch:

Live today, VoIPTroubleshooter.com includes over 60 pages of diagnostic information related to VoIP call quality. Two new features include: an Open Speech Repository, i.e., data base, and an IP Network Impairment Simulator. The Open Speech Repository provides high quality audio recordings of speech files in several different languages that test engineers can use for testing and related applications without any restrictions, except for source credit.

Read more: Telchemy Offers VOIP Help