VoIP Now

Who's Got Muni Wi-Fi?

Municipal Wi-Fi networks are popping all over the world, making connectedness easier for lots of people and pushing the possible ubiquity of dual mode cellular/ Wi-Fi phones (i.e., VoWiFi or VoIP over WiFi). Well, DailyWireless has a great list of the 10 most connected cities in the world, and goes in depth about each city. Note that the first 5 cities or so listed are all in Asia. Most of the rest are in Europe. What's up with North America, then?

Most interesting (to me, anyway): Shoreditch TV, which is a network of 100 public cameras in Shoreditch (east London, UK), broadcasting to the Internet. The idea is to dissuade criminals. Little Brother 2.0? Neighborhood Watch takes on a new meaning. Then again, the UK is considered to be one of the most endemic surveillance societies in the world.

Of course, there are loads of municipal Wi-Fi projects going on in the US and Canada, especially a few big ones in Silicon Valley (42 municipalities over 1500 sq mi).

Where in the World Are... You?

GPS capabilities are supposedly one of the current and near-future hot features of cell phones. It's been predicted, probably even before 2001, that all cell phones will have GPS capabilities, which would be particularly useful for tracking people in emergency situations.

However, for tracking bike and pedestrian traffic, PNAs (Personal Navigation Assistants) are supposedly not ready. That's primarily because the necessary cartographic work for bike and foot traffic hasn't been done for most places in the world, so having a nav system for them is pointless. And for safety reasons, you cannot assume either type of traffic can use regular GPS nav maps.

Still, when and if such maps are recorded, smartphones such as the Apple iPhone or some of the Linux keyless handsets might be ideal platforms for PNAs for cyclists and pedestrians. In fact, I'm predicting a general rise in world tourism over the next two decades (pretty easy prediction to make).

So PNAs for this type of traffic could become a burgeoning market. But having worked in GIS and digital mapping for seven years, I know there's a lot of work to be done to satisfy a market that doesn't yet really exist, and may not exist for many years. Not an easy business decision to make. On the other hand, as Clumsy on their feet says at the end of the article, paper maps never run out of batteries.

Cisco Entering Social Networking?

Cisco, as you probably know, has been a leader in computer networking for a long time. Nuno at 21talks writes about an NY Times article about Cisco buying Tribe.net, a social networking site. As the NYT piece says, it's a cuirious pairing. I can't see why they're getting into this market. But then, I can't understand why Microsoft might be coming out with a Zune phone. (Other than that Steve Ballmer hates Steve Jobs.)

But Nuno thinks it's a good idea, and points out that Cisco previously purchased a social network design firm. Didn't know there were such things. Seems kind of redundant, considering that sites like Ning let you design your own social network for free, and within a few minutes. Who knows. All I've learned is that when a large company buys a web services blog, they have some reason for doing so, even if it's obscure.

Skype Wants Changes To Mobile Network Access

Skype, whose name is synonymous with VoIP for some people, wants cellular networks operations to change, to be more open. In fact, they're demanding that the US FCC make changes to a legal decision from 1968 related to the AT&T network so that it applies to cell networks. That's because mobile operators limit the traffic on their networks, especially data networks.

Read between the lines and you'll probably conclude what I have: that Skype needs this ruling changed to offer full mobile Skype. Of course, they would also become very competitive with mobile operators as a result.

The irony of course is that while Skype has an open developer API (Application Programmer Interface), their networking protocol is closed - as in private. The general idea behind their request is a good one, but it just seems kind of hypocritical when they won't open their protocol - a decision that has caused companies, universities and countries to ban Skype use. And they're couching as a consumer rights issue.

What Internet TV Needs: 7 Suggestions/ Concerns

What's listed here doesn't preclude the possibility that some software or web service already does it. This is my list of ideal IPTV (Internet TV)-related functionality.

1. Mobile TV.
   This is fine, but with wearable, comfortable goggles that project a virtual large screen. Little tiny phone screens won't cut it. The goggles are out there. They just need to be married with smartphones and PDAs. (i.e., maybe through     Bluetooth, since cellular data plans are outrageously priced in some countries.)
   
2. Wireless streaming.
   From my computer to my TV, if I want. (Though my computer screen is still larger than my TV, and I use an external TV capture box, which gives better performance than IPTV.) Apple's tentatively called iTV, for the digital living room, is one example.
   
3. Faster Internet connection speeds.
   Let's face it, Joost might be nice (I'm still waiting for a Babelgum invite), but a faster connection would help, obviously. And what happens, for example, when everyone in my neighborhood on cable Internet starts watching at the same time? At that point, I turn back to regular cable TV, as will others. The success of IPTV hinges on much faster connection speeds.
   
4. More bandwidth.
   My cable Internet provider caps me at 6 Gb/mth. I eat bandwidth for breakfast. I can use a Gigabyte in a single day sometimes. But can I buy more bandwidth? Noooooooo. Instead, if I go over in a given month, they'll warn me twice then cut me off until the next month - something I simply cannot afford to have happen, as a freelance writer. And with Joost's bandwidth consumption, this is important. Which is why I've stopped using it, beyond a few beta tests.
   
5. New compression coding.
   Wavelets theory is an ultra-geeky discipline created by brilliant physicists in the 1970s but has roots in studies done in 1909. It's pure, advanced applied mathematics used to model a lot of phenomena, and a math professor told me that even most PhD's in math or physics don't understand it fully.
      
   Data compression of images and video is one application, and depending on the algorithm used, the space savings are phenomenal. The benefit is that a crunched file would download very quickly. The problem is, that massive crunching requires a fairly significant amount of processing power to uncrunch for viewing. It certainly could not be done, with present home computers, in real-time. That is, you couldn't watch streaming video as it comes in over your Internet connection if the video data has been massively crunched with wavelet compression. The alternative is to not compress and have a faster connection, or more powerful graphics cards.
   
6. Quadcore video boards.
   The whole net neutrality debate was sparked, from what I interpret, when Internet providers felt they had to apply a tiered price structure for connections based on expected usage. Fact is, if we suddenly had the billion or so current Internet users all using VoIP and/or IPTV simultaneously, the current infrastructure couldn't handle it. (Correct me if I'm wrong.)
      
   We all probably want faster connection speeds, and they're coming, but will take time to roll out. What could come sooner is a new set of video compression codecs (last point) coupled with high-power graphics cards sporting their very own quad cores or more. If our graphics cards were powerful enough, and we used super-crunched video formats, we might possibly reduce bandwidth requirements down to a point where every Internet user could potentially watch Internet TV simultaneously. (Of course, it'd be nice to have something similar for VoIP communications: a quadcore sound card.)
      
   
7. More content, more choice.
       Video sharing sites already have a great deal of content choice, though not all of it is necessarily watchable. Soft clients like Joost are young yet, but will need - in my honest opinion - a great variety of content, and a pay-per-view model without advertising. And that requires sign-on from production houses.
    

Pretty much everything I've said here could apply to VoIP quality of service as well.

The Websident of the United States?

I'm happy to report that Senator Barack Obama's campaign team did finally answer my email. I'd griped about that when discussing his very popular social network.

Top Digger Muhammad Saleem says at 901AM.com that Senator Obama is your websident so far. This is based on his social network, my.BarackObama receiving critical acclaim. What's really amazing is that over 4,000 members have created blogs there and 3,000 more have created fund-raising pages. Brilliant. I hope Hillary, Rudy, John E., and others are taking note of this and Senator Obama's Internet TV channel.

I still think there's room for a political streaming video channel, but where, say, bloggers host podcasts/ vodcasts and ask the tough questions most mainstream media won't ask.

Viacom and Joost Exchange Video Valentines

An email in my inbox (as a Joost beta tester) announces a content deal with Viacom. Very cool. While Joost has two problems, bandwith hog and minimal content, as an IPTV (Internet Protocol TV) client, I'm very impressed with the video quality. And while content is still minimal, there still is enough there for many hours of viewing pleasure, including a great deal of variety.

Now while a lot of the Viacom content is from MTV past and present content, which I don't care much about, it might be kind of fun occasionally tuning in to watch old Beavis and Butthead episodes. Uh hunh hunh huh. But the rest of the Viacom content will likely appeal to a much younger crowd, not me.

Now I'm already a TV and IPTV junkie, but what I would shell out money to see is older stuff that is really hard to find - at least in Canada, and possibly in the US, maybe everywhere. For example, I'm a big fan of the American cartoon Freakazoid,and of a cancelled sci-fi-ish show called Early Edition. There are also 1950s short movies of Batman and Superman that I'd love to see and haven't found on DVD. (Granted, I didn't look very hard.)

Put all this sort of content (let me pick) online and allow payment via PayPal (Skype's sister company) and you've got me. Maybe make it a download of the month club sort of deal, for $9.95/mth, lots of content to choose from. And no ads.

Problem is, Joost streaming video content isn't stored on your computer (that I know of), so I'd have to download again and again. Hopefully they'll come up with a solution for that (straight burn to your computer's DVD drive?). But even if not, the Long Tail suggests that Joost (or someone else) could do very well by keeping the price low and offering lots of choice. And when Apple's iTV device comes out, I'm hoping I can stream Joost content straight to the 40" TV I'm hoping I'll buy myself for Xmas this year.

XM Satellite Radio Gets Sirius About Merger

How could I resist a pun like that? Peter Csathy talks about the pending merger of two Satellite Radio operations XM and Sirius. No doubt radio shock jock is wondering if the merger will affect the hundred plus million or so he got. Damn. Maybe I'm in the wrong business. I'm only offending people in the blogosphere.

Now I've been saying (just to myself, mind) since last Fall that they should merge. Service offerings-wise I think that customers would benefit. of course, for some cell phone handsets, you can already get XM Satellite Radio shows for about $15/m. No expensive XM device necessary.

Of course, whatever the merged company is called, they might consider offering some programming via the Internet. That's if their satellites are IP-based.

VoIP Roundup - Mon Feb 19, 2007

The Eye in the Sky: Pushing the IP Communications Envelope
There's been a lot of talk about SEDs - service-enabled devices. SEDs will have their own IP address and are thus pingable across the Internet. Now, imagine that you could query a satellite view a web browser. That's what Iridium is planning: satellites that monitor the Earth, taking pictures. And because they'll be IP-based satellites, Iridium can sell services over the Internet to clients who need to monitor, say, a facility.

Privacy Obligations For VoIP and Telecom Providers
The US FCC is rethinking how it will expect telecoms and VoIP providers to handle CPNI (Customer Propietary Network Information) data - or what amounts to call records and subscriber information. This is as a result of the Hewlett-Packard phone records pretexting scandal and similar cases. Privacy and Security Law Blog has more details on some of the new rules that may be imposed.

Cell Phone Been Bugged?
Despite all the issues of communications -related privacy and security, it's unlikely that most of us have our phones or IP communications bugged. But for whatever reason (jealous spouse, insane employer), if you suspect you do, check out Lauren Weinstein's post How to tell if your cell phone is bugged and the accompanying YouTube video Is your cell phone bugged?

Barack Obama's Social Network?

You might have heard that US presidential hopeful Senator Barack Obama has his own Internet TV channel, thanks to Jeremy Allaire's Brightcove and lots of campaign money. It appears that now has his own social network.

Brilliant way to use IP communications to build your potential presidential profile. (They've taken Peter Csathy's video politicking advice to heart, whether they realize it or not.) Wonder if the Senator has Skype? What about other leaders? It's all well and good, but maybe he should use some of his campaign money to actually answer email. It's been more than two weeks since I sent an email message to his campaign headquarter via his website. I know that as a Canadian, I don't really matter votewise - or at all. But how hard is it to hire someone to set up an automated response, to acknowledge the email? His campaigners might be doing all the new media stuff correctly, but they seem to have email communications handling all wrong. Or maybe men in black helicopters are intercepting my communications. I think I'd better wrap my house in tinfoil.

Joost Bandwidth Issues?

Hmm. Very interesting. Never thought about it until I read about how much bandwidth Joost IPTV might be consuming. It does make sense, consider how high quality the video content I've seen on Joost has been. Quality frames take space and thus increase bandwidth.

This is kind of scary, considering that my cable Internet plan only allows 6 Gb per month, if I'm not mistaken. For someone who tests a lot of software, it's not much, but there are no options for home-based service for me. If I exceed the limit, I get a warning, then I get cut off. Which is disastrous for me, who works all day online.

It'll teach me to read the fine print more carefully on the next application I test. But it does make me wonder how successful Joost is going to be if you can use up 1 Gb in 10 hrs. While it's unlikely I'd watch that enough Joost yet (because of lack of content I'm immediately interested in) to make it an issue, the issue is still there for the future.

I'm not sure that average future Joost user is going to want to be a node in a torrent-style network, and hence bandwidth is a real problem. From a very technical point of view, they could consider using fractal image compression or even wavelets, but those are quite advanced methods that don't necessarily lend themselves to video (fractal compression is a lossy method). Ultimately, it seems to me that Joost is ahead of its time, like Skype was, and that for true enjoyment of IPTV on the larger flat panel TV screens that are coming out, viewers are going to need quad processor computers with souped up graphics processors and Gigabit access speeds.

How VoIP, Presence and IP Conferencing Can Help Your Career

Phoneboy recently asked (on Web Worker Daily) whether telecommuting affects your career, in response to a similar piece at Network World. If someone asked me that in the 90s or even a few years ago, I'd have to say yes from what I've seen of other people. It's primarily because of what he said: executives tend to want/ need to meet their charges. It's been hard to do over traditional telecom and conferencing solutions that the average company can afford.

Now, we're entering into an era where VoIP, IP conferencing and presence applications are setting up the framework where I don't think telecommuting will be as much of a barried to career advancement. A couple of things still need to change: faster computers, faster Internet connections and attitudes. It might be years (maybe a half generation?) before being a full-time telecommuting executive becomes commonplace. It might become a reward of the job. (Part-time telecommuting has been allowed at all levels of staff for years, as I learned at IBM in 2001, even for meetings.) The acceptance may take time but the technology is already here.

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.

VoIP Roundup - Sun Dec 31, 2006

VoIP Comm Now Mainstream?
Ken Camp points out that VoIP is no longer a niche and that it's gone mainstream.

San Fran Wi-Fi Is No-Fi
At least for now, San Francisco is delaying their citywide Wi-Fi network. Again. This is the network that Google is involved in. The issues come from within city council regarding who will own the network.

US Cellular Network Outages Kept Secret
When cell phone networks have service outages in the United States, they aren't announced. In fact, the FCC ordered "wire line" suppliers in 2004 to report them, but in turn removed them from the Freedom of Information Act. [via VoIP and Enum]

The US Presidential Race: Pushing The Viral Video Angle

Senator John Edwards has already announced his plan to run for the 2008 US Presidential race under the Democratic banner. Edwards is the politician who had experienced a recent faux pas dissing Wal-Mart's labor policies on the same morning that one of his campaigners asked them to provide a (free) Playstation gaming console for one of his family members. Wal-Mart of course told him to stand in line like everyone else. Maybe they thought it had YouTube?

At any rate, he seems to have taken Peter Csathy's advice about video campaigning and had someone post an initial campaign clip on YouTube. Now if every politician listened to Peter, imagine how much in campaign savings they would garner. Wouldn't that show some proof of fiscal responsibility?

[sources: 21Talks]

Mobile Comm: Over A Billion Served?

It's agreed upon in many circles that the next billion mobile customers are going to come mostly from developing nations including India, China and a few other Asian countries and very likely a number of African countries (some are already heavily moving into VoIP and looking at phone number portability). But will these customers be served properly? Will they get what they need for their use? In many of these countries, average incomes make it difficult for everyone to have a mobile phone, whether they need one or not.

Imran Ali has a look at some mobile market studies done by various people - including Jan Chipchase of Nokia - and analyzes some of the findings regarding the sociology behind phone sharing, as well as other related issues.

Funding Free Municipal Wi-Fi

You've no doubt heard or read about all the muncipal Wi-Fi projects popping up across the United States and elsewhere. In some cities, there are several subscription options for different access speeds. In others, it's all free. One question that comes to mind is who is going to pay for the costs of this "free" Wi-Fi? In Portland, Oregon, Microsoft is involved in a full-speed wireless network that'll offer free access in return for showing users paid advertising. Dailywireless asks whether this will work.

My own feeling is that in any given city, some people will be more than happy to have free Wi-Fi, even if they have to watch ads. It's why VoIP subscription models like that of Globe7 could work. However, are there enough such people? Way back in the early 90s, I tried free regional long distance calling in return for listening to ads (pre-Internet) and got sick of wait, the crappy music, and the same old ads. If, however, I had to watch, say, movie or TV trailers for access to the Internet, I might be okay on that. What about you?

If there are not enough people supporting the advertisers of such wireless networks, however, what happens to the city? Do taxes go up? Do they force paid subscription on users? And when people say sayonnara, does the city tear down the infrastructure? That costs money, too.

The concept of free wireless access is relatively new. I'm not sure, but I think it was used by some ISPs for dialup Internet access, though I'm not so sure that succeed as I can't think of any examples. Only time (or indepth surveys) will tell if such payment models will work.

Phisher Kings: Teach Someone To Phish?

Someone's psychology, sociology, and/ or electronic anthropology doctoral paper is lurking beneath all this latest research that shows phishers/ spammers/ scammers are using ever sophisticated methods to grab your attention so they can grab something of yours - preferably e-money.

Now I'm not going to get into the psych makeup of phishers; that's not my intent, despite my opinions. But the low cost of the latest communications technology and its ease of implementation makes it ever so much easier for you to at least be a target if not a victim. That means more vigilance in 2007 and beyond, as several experts are saying that the lastest avenues for phishers are vishing and smishing (SMS phishing). VoIP and SMS are, in fact, the latest tech platforms for phishers.

Tech intelligence and social intelligence seem unfortunately mutually exclusive in these cases. Fortunately, about computer-based crime in general, those getting caught are being given stiffer penalties.

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.

5 New + Recent Terms In IP Communications

IP telecommunications is obviously a huge area with many facets, and new ones popping up. While those in the industry and some VoIP bloggers may know the terms, the general public likely does not. I often use Google Trends to compare terms and their relative search volumes. Which is what I've done witih some of the terms below. These are terms to watch, because we'll see them mentioned in the media more often.

1. Spot dialing.
   Brian McConnell came up with this to describe make a call over a Wi-Fi hotspot. I.e., as a replacement to Voice over Wi-Fi (which some people call VoWiFi but is rather techy).
2. TVoIP, teeVoIP.
   Ken Camp wrote a very evocative piece called Ken's magnificent Seven for 2006, about what he thinks have been the hot growth areas for IP Comm. He came up with the term TVoIP to represent user-generated content such as that at YouTube and blip.tv. I've been referring to this as a facet of IPTV, though that's probably incorrect.
3. vVoIP, VVoIP, WoIP.
   This aren't new, per se, as there are references to it back to at least 2004, possibly earlier. But Google Trends says there isn't even enough search data for them to show a comparative graph. But there are three ways to signify this, possibly causing confusion. Should we use any of these or come up with yet another one and hope it'll catch on? Video calling works for me.
4. POVS.
   Garrett Smith came up with POVS, Plain Old VoIP Service to refer to any VoIP service mimicking traditional phone calls. I.e., what some people call pure play, when you use a regular telephone with an ATA (Analog Telephone Adaptor), such as with Vonage and PhoneGnome.
5. FMC, Fixed Mobile Convergence.
   Most searches for FMC are probably for one of two companies with "FMC" in their name. (Google doesn't give hard search volumes.) The longer version doesn't register in Google Trends. But true FMC will push dual-mode cellular/ Wi-Fi calling forwards, though maybe it needs a friendlier name.

What do you think about these terms? A bit technical, in some cases. Got any suggestions for alternates?

Is Wi-Fi Bad For Us?

George Ou of ZDNet writes about a UK woman who claims that Wi-Fi makes her sick. He then lays out a test scenario that he'd like to give her to prove that she can in fact detect when a Wi-Fi access point is present, like she claims. This isn't the first that I've read about something like this. Just about a month back, about someone in the UK - possibly the same woman - claiming they were getting headaches after installing a Wi-Fi router. This woman, Kate Figes, says Wi-Fi leaves her feeling exhausted, nauseous and sleepless.

Ou calls it "EMF junk science" but the WHO (World Health Organization, not the band that causes hearing loss) has an EMF project, due to public concern about health and electromagnetic fields. Figes isn't the only person claiming this. The WHO's brochure on EMF (PDF, 2 pgs) that "[no] major public health risks have emerged from several decades of EMF research but uncertainties remain." Maybe these people are the proverbial canaries in the coalmine of an cumulative illness that takes time to manifest.

I've certainly found myself more fatigued since installing my Wi-Fi router, but that could be for several reasons, including the fact that I always feel fatigued come Oct/Nov, during Daylight Savings Time change (or whatever it's called). As well, since installing my router, I also spend many more hours working on my computer than previously. Sleep is something I do because I have to. And lack of sleep gives me headaches. A few minutes outside in fresh air almost always seems to make a difference.

It's an easy correlation to assume it has something to do with Wi-Fi, but I'm certainly hoping it doesn't. Imagine what'll happen to the fledgling municipal Wi-Fi and Wi-Fi VoIP phone industry if it does.

VoIP Becomes More Free For Schools

The US FCC's E-Rate program means US$2B in federal grants for K-12 schools to Internet, telephone, and hardware costs. While that doesn't mean all schools will opt for VoIP, there are provisions on the application form for VoIP as well as mobile devices such as RIM's BlackBerry and Palm Treo. [via ExtremeVoIP]

A number of schools in the US have started using IP communications for their intercom system, and others for actual VoIP outbound calling. The most important technical issue for schools in this case should be the implementation of E911 emergency calling.

VoIP Roundup - Mon Nov 20/06

California Hospitals Implement Multi-Language VoIP Project
A number of public hospitals in California are implement VoIP technology with live multi-language call support in order to serve the large non-English speaking community AT&T is part of the project. Implementation was completed recently. [via TMCNet]

Phonezoo Ringtones Social Network
Diehard cell phone ringtone lovers now have a social network of their own. Phonezoo lets you create your own ringtones, share them with other members, and discover what they have. You can even rate and discuss each ringtone. [via Roam4free] Color me cynical, but do people have nothing better to do, or am I hopelessly out of touch?

How Now, Apple iPhone?
So if Apple really does come out with a so-called iPhone, how do you think it'll operate? Unlocked with SIM cards? Carriers? Daniel Raffel at O'Reilly Emerging Telephony provides his insights, concluding that a carrier partnership might be more painful to them.

VoIP Roundup - Wed Nov 15/06

Skype Enterprise Features Coming?
Skype execs have hinted at upcoming enterprise and call center features. So maybe this will be how eBay finally monetizes Skype?

Speed Demons
The 100-Gigabit Ethernet (100-GbE) technology is here, being demonstrated by a number of companies and the University of California Santa Cruz. A test run sent a signal from Tampa, Florida to Houston, Texas, and back - a first for a live production network. If I understand this correctly, IP backbones will get this technology fairly soon. And as 100-GbE becomes commonplace, likely in several years time, it should mean some incredible real-time video conferencing ability, superfast downloads of movies, and live video broadcasts, to name just a few benefits.

Legal Issues Surrounding VoIP Enterprise Implementations
TechRepublic details legal issues to be aware of when planning a VoIP implementation. They have real alphabet soup of issues, some of which I've only peripherally aware of: SOX/ Sarbox (Sarbanes-Oxley Act), GLBA, HIPAA, E911.

My Phone Is Red Hot; Your Phone Ain't Diddly Squat

Patrick Barnard of TMCNet asks how much are you willing to pay for fast wireless mobile broadband? Lucent Technologies has done research that shows that both consumers and businesses are willing to pay a premium. The United States and Canada have only recently started rolling out 3G (third-generation) services such as HSPA (HSUPA/ HSDPA - or supa-dupa, as I call it) and EV-DO wireless broadband networks, whereas these are already available in other parts of the world.

But Russell Shaw (who seems to have cloned himself for other blogs) writes at IP Networked (a new GigaOm web property) that he's skeptical of EV-DO, mainly because market analysis shows that many 3G phone users "fail to understand [3G service] benefits." He says that he is actually subscribing to EV-DO service through SprintNextel, which gives him a discounted rate, but that ROI for him specifically might not be worth it.

For me, EV-DO is only a backup, at present. As an onine-based freelancer writer/ technoblogger, I need a "plan B" in case I have any problems with my cable broadband access, or in case I cannot find Wi-Fi network for my laptop if I'm mobile. While it's expensive and I cannot exceed a maximum of 250Mb/month bandwidth. But given that my revenue would disintegrate without it, if I couldn't otherwise access the Internet, it's worth every cent. And I can probably write it off as a legitimate expense against earnings. It's also good for me, since I plan to move next year, and having had lousy response time from cable installers in the past. EV-DO for me is a security blanket, albeit one that could be less tattered and a bit more comfortable - especially in the area of mobile VoIP, which it sucks badly in, at least on the Palm Treo.

Putting Your Best Face Forward

If what Ken Camp is saying in Advances in 3G mobile solutions include facial recognition in video, you might want to make sure that you wake up on the right side of the bed. Imagine: your hair is mess, you're bleary-eyed, and depending on your inclination, your face is either unshaven or unmade. And guess what? Your mobile phone doesn't recognize you and won't let you place a call. Damn biometric machines. Always thinking for themselves and getting it wrong.

Of course, I'm exaggerating. You don't have to worry about video calling etiquette for video-based facial authentication. But there are experiments going on that use facial biometrics to control functions on a mobile phone. This includes more important functionality such as contactless payment, access control, and identification. The biggest problem I see with this, which Ken also points out, is environmental conditions (such as darkness) that might give an inaccurate biometric and thus lock you out. It'll probably take a few years for DoCoMo and others to work these issues out. But if they succeed, we'll certainly live in interesting mobile times.

VoIP Roundup - Mon Nov 06/06

Electrocom VoIP Intercom
First there the VoIP-based home security alarm systems from InnovAlarm, Alarm.com, and others. Now there is Electrocom's VoIP intercom, which they're promoting as a "safety and security solution" for K-12 schools in the US state of Washington. Since data cabling already exists in these schools, installation is minimal. They system works in hands-free mode and allows two-way communication as well as facility-wide paging. [PRWeb via Yahoo News]

Pairing iBlue Mac Mini PBX And Snom VoIP Phones
The iBlue IP PBX from 4s newcom, which is essentially integrated into an Apple Mac Mini, will work with new VoIP phones from snom. Now that's sort of expected considering that 4s newcom is a spinoff of snom. In March of 2005, snom had announced that they were offering a mini IP PBX that could fit into a briefcase.

MyNetFone Satellite VoIP
I've previously commented that satellite VoIP stands little chance against other types of VoIP service. Though I have limited choices for my Internet connection, it actually never occurred to me that there is a market for whom satellite-based Internet, and thus VoIP, would actually be the only option. MyNetFone must have realized this too, when deciding to offer satellite-based VoIP service to rural parts of Australia. ISPhone, by the way, is another Australian satellite-based VoIP service.

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]

The Electronic Global Village Expands

First Singapore announced plans for implementing 2400 hotspots in Northern Singapore. Now Japan plans a giant wireless mesh network over 100 cities, resulting in open broadband access for over 50Mln people. Earlier in the year, Taipei City, Taiwan announced a wireless project to replace PSTN, with an aim of having 200,000 wireless VoIP phones for city employees by year end. And of course there's the countless other cities in the USA and other places installing their own municipal Wi-Fi networks, or just finishing the bidding process.

And with free or inexpensive municipal Wi-Fi, growing in abundance, there's expected to be a burst in sales of plain Wi-Fi or dual-mode Wi-Fi/ cellular phones. If ever there was a time to consider buying stock in wireless and IP-related hardware manufacturers, it's probably now. Just wait until after the US stock market crashes this November due to the mini-tech bubble that's formed.

Embedded VoIP: You've Got Gizmo Project In My LiveJournal

FierceVoIP asks, Will bloggers want to talk to each other, referring to the fact that the LiveJournal weblogging  platform now has Gizmo Project embedded into it. Users of LiveJournal (LJ) can see the online status of their "friends" and communicate either with voice or text chat, or leave a message.

If you want to try Gizmo Project for LJ Talk, you'll need a free LiveJournal account first, which involves a ridiculously hard to read captcha graphic (to prove you are human and not a spambot). Then you'll have to choose between two different types of free account or a paid one. (While I like MovableType, a cousin to LJ, I'm not a big fan of LJ.) Finally, you can download the clients (Win 2000, XP = 11.1 Mb; Mac OS X 10.3.9+ = 18.9 Mb). A microphone and headphones (or speakers) are obviously required.

Once you've downloaded and installed LJ Talk, when you run the client, your "presence" will register on your LiveJournal journal pages, to your LJ friends. (This is based on what I can tell from the LJ pages. Since this is essentially Gizmo Project, I have no plan to install it as well.) Now you need some friends to try it on. Go search for and invite people on LJ.

Will you be my friend? If you are like me and have no friends, you can at least use the client to voice post to your LiveJournal weblog, which is a nice feature. If you already have LJ friends, your contact list will be used to populate LJ Talk. I suspect though can't confirm yet that LJ Talk supports click-to-call, whereby someone can click on a "call me" type of button on your LJ journal web pages to initiate a VoIP call with you in your LJ Talk client.

By the way, you can also use a variety of other compatible clients to chat with someone on LJ Talk.

Enterprise IP Telephony: It Costs How Much?!

While there are a number of reasons that businesses have been slow to adopt Internet telephony, there are companies doing big deals for enterprise VoIP migrations. Williamette Dental in Midwest USA operates 69 offices in the states of Idaho, Oregon and Washinton. They signed a deal with Qwest Communications International Inc. to set up a data networking and VoIP system over three years. At a cost of US$3.8M. The VoIP system will be used to transfer calls from customers to their call center.

The amount took me aback. That seems like a lot for VoIP. That's just over $55K per office. Now I don't claim to know the costs of traditional telephony systems - despite having worked for a large telecom - but $55K per office sounds like an awful lot. Then again, they may need to lay down coaxial, put in IP PBXes, interconnect offices. configure the software and the system, do any necessary training, and possibly manage the services. There's also the cost of call minutes and reportage, although these will likely be less than for PSTN (Public Switched Telephone Network) systems.

But the good news for businesses that cannot afford to spend $55K per office is that there are probably probably plug-and-play IP PBX solutions that are SIP-based (open standard) and which cost less than that. One possible option is an Asterisk solution, maybe even the hybrid Skype-Asterisk solution for businesses, from Pika Technologies. I'll try to expand on business options in future posts.

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.

Virtual GPS

Don't have GPS capability in your mobile phone or PDA? The Navizon Web API from Mexens Technology can pinpoint users by triangulating signals from nearby Wi-Fi access points. This location can then be used as a reference point to show the user additional location-based information such as nearby stores, banks and restaurants.

This functionality could be teamed up with the mobile click-to-call/ pay-per-call that Microsoft and Ingenio are now offering (jointly with Ingenio). It could be a great way to promote local businesses in cities that are offering municipal Wi-Fi. Given that cell phone use is expected to increase all over the world - particularly with dual-mode Wi-Fi/ cellular phones, there is likely to be a market for such services.

[sources: Wireless IQ, Biz Yahoo]

SMS Text Messages In Non-Roman Alphabets

While English might be the de facto language in India, especially for business, and usually the first language taught in most schools there, Hindi is the official language. The script known as Devanagiri, and the language both derive from the dead language Sanskrit, which is a sister language of Latin.

The fact that Hindi is written in a non-Roman alphabet makes it complex to when it comes to computers and cell phones. However, Feedelix Wireless has managed to send the first-ever Hindi SMS message from a subscriber in India to another in San Diego, California. Feedelix's HindiVayuSMS software was used. [via Cellular News]

This of course bodes well for other languages that do not use the Roman alphabet. An alternate method would be for one participant to type in English, and for on-the-fly language translation software to convert to Hindi, even in Devanagiri script. This of course is only a partial solution. Other possibilities are to auto-generate Devanagiri script by translating voice input.

Microsoft's Click-to-Call

Google's doing click-to-call, so why not Microsoft? Ingenio and Microsoft have teamed up to offer a mobile version of click-to-call, which incorporates pay-per-call advertising on Windows Live Search for mobile devices. The pay-per-call advertising market is expected to reach nearly US$4Bln by the end of the decade, for mobile or otherwise. Essentially, advertisers get exposure in Windows Live Search, and if a mobile user clicks their link to find out more information about their products and/or services, then the advertiser pays Microsoft and Ingenio for the call. Example advertisers might be local businesses such as restaurants, hotels, travel agencies. [PR Newswire via Biz Yahoo]

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.

Sightspeed: SMBs and Video Conferencing

Back when I was working on a Master's degree, one class I was in was part of a trial with another nearby university. The course was being jointly taught by two professors, one in each town. The room my class was in was a small amphitheatre with a large screen usually obscured by the overhead projector's screen. The other university had a similar setup. Occasionally, our lecturer would turn on the transmission and we'd see the other professor, who would conduct the rest of the lecture - or vice versa with our professor. This only happened a few times during the semester, and the transmission was over a satellite link. It was video-conferencing of a sort, but very expensive, if I recall.

This was back around 1993-4. I know the room is still there, although I don't know if they still use the satellite link. The campus had access to the "web" back then, which consisted only of email, ftp, gopher, archie, newsgroups, etc., access. The full Internet was only a few months away. Today, there is Wi-Fi across parts of the campus, accessible if you have a student or alumni account. But video-conferencing is not only a lot more accessible these days but far less expensive. Anyone with an internet connection and a video-calling soft phone like Sightspeed can have a video conference.

In fact, several SMBs (small and medium businesses) in the United States are using video-calling functionality. A small teacher certification business in Texas uses Sightspeed's video-conferencing feature to communicate with their students. The founder of a consulting firm also uses Sightspeed to communicate with both clients and colleagues (in another office). Yet another company uses Skype and the Festoon add-on, which bridges both video and voice calls between Skype and Google Talk.

VoIP itself can save a small business. Video conferencing can save a business even more, coupled with the ability to share clickable URLs, documents and desktop applications, the need to travel even locally between offices can be reduced significantly. This not only saves money but valuable time.

Wibree: New Nokia Wireless Connectivity Protocol

Nokia has introduced Wibree, a short-range wireless connectivity protocol, which is complementary to Bluetooth and UWB. It's the culmination of five years of development and is ten times more energy efficient than Bluetooth. Wibree operates in the 2.4Ghz band, has a range of 10 meters, and a maximum speed of 1Mbps (megabit per second). (Bluetooth operates in the UHF above 6 Ghz.)

Instead of being used in cell phones, it would be in watches and possibly enterprise devices such as keyboards, mice and digital pens. Other possibilities are "wearables" such as intelligent jewelry. Nokia already has dula-mode cellular VoIP phones. I'm wondering if Wibree could be used for VoIP "communicator" badges that double as lapel pins or brooches - similar to Vocera's Call Badge.

[via CBR Online, CRN]

The Demise of The Blackberry?

I don't believe this for a minute, given how long BlackBerries have lasted and how much their users love them, but.... IDC has release a report saying that BlackBerries cannot last against push email solutions from Microsoft and Nokia. [via Teleclick.ca]

Here's why I don't believe it. The people who use BlackBerry, as far as I'm aware, are employees. Their employers have all the equipment and servers in place. While Microsoft and Nokia may succeed in signing on new business, existing RIM clients are probably unlikely to want to do a wholesale switchover. Unless there was some overwhelming benefit. And not just cost, I'm thinking.

Another factor? RIM stock. I know former RIM employees who became millionaires and retired. One of the founders of RIM set up an incredible science facility in the Waterloo, Canada area, where RIM is headquartered, with $95M of his own money, if I remember correctly. Then the brightest physics minds of the world were invited for fellowships.

It isn't just Canadians, particularly employees, who bought stock. No doubt loads of Americans, particularly executives, bought stock. With Microsoft stock sucking bollocks for about five years now, execs aren't going to let RIM stock decline just because Microsoft or Nokia comes out with a new competitve product. I'm sure of this. When Microsoft had hot stock, I found this same kind of attitude from executives I worked with. I despised Microsoft's strategies back then, but they defended the big M, even though I presented facts as to why they were not the better choice. No luck at all.

So even if Microsoft and Nokia come out with something better, they have a whole "Crackberry" culture to deal with. And with RIM moving into cool phones like the Blackberry Pearl and plans to get into VoIP, they are expanding their potential customer base. They may not have as deep pockets as Microsoft and Nokia, but they're deep enough to, say, make a few strategic acquisitions. And they are a wildly innovative company with a large "fan" base.

Lights, Camera, Sightspeed

MTV is looking for VJs for their Total Request live show, which often features Sightspeed-driven videos by fans. Sightspeed is arguably the best video-calling VoIP soft phone available. Andy Abramson talks about the MTV casting call and some of the positive aspects of this development, including a new type of social interaction enabled by VoIP.

Earlier this year, Skype promoted a special contest for fans of alt-rock band Coldplay. The two young women who won got to Skype with members of the band for several minutes. Now consider the Japanese TV broadcasting via Skype. So there is great potential for celebrities to connect with fans via IPTV over VoIP IMs, either live or recorded. And with video interaction through Sightspeed, Skype or similar soft phones, citizen video could become the source of unique TV or IPTV content, such as 11 Cameras (which is supposedly a view of the lives of several people via 11 video or webcams viewed through instant messengers).

Free Public iKiosk VoIP Phones

Ginny Granger writes about a network of free public VoIP phones called iKiosks in Aberdeen, Scotland. The iKiosks offer free email, videomail, internet access, and free VoIP phone calls. Calls are based on software similar to Skype. Robert Gordon University is one of the first iKiosk sites.

This is a marked contrast to SJSU (San Jose State University) in California (and other universities) where they had planned to ban the use of Skype on campus but later reversed their decision. Maybe something like iKiosk would be more to the university's liking? Free public phohes is a radical idea. In Australia and other places, public VoIP pay phones are being rolled out.

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.

Skype Ubiquitous On Mobile Phones? Not Just Yet

So it's not just me. According to a CRN article, Skype on mobile phones and PDAs has some technical hurdles to cover before it'll function correctly. Skype CEO and co-founder Niklas Zennstrom specifically mentioned Symbian phones being more difficult than expected. PocketPC-based phones and PDAs do have Skype, but a lot of other mobile communication devices do not. Or when there is a Skype-related mobile solution, such as SoonR, there are still technical glitches, depending on your phone. So I can almost sympathize with Zennstrom.

In fact, Skype isn't really the only VoIP service having this problem of achieving ubiquity on mobile devices. I've had little luck getting anything to work on Palm Treo 650, a very specific phone. Mino Wireless works, and Chris from EQO says that EQO Mobile should work, too. (I'll try it this weekend.) But as for actual mobile Skype, well Niklas Zennstrom says that have nothing to offer yet and no timetable besides.

Too bad. I guess I'll be waiting with bated breath, and keep my outrageously expensive cellular wireless data network plan going in anticipation. Wonder how many additional active users Skype'll gain when they do go mobile? Maybe Skype needs to start acquiring a few companies with some of that US$2.6B they got from eBay last year. In the meantime, though, Skype-specific Wi-Fi and dual-mode phones do seem to work, so you might want to consider one of those for a bit of mobility.

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]

VoIP Roundup - Wed Sep 27/06

EverywhereNet: Might As Well Be On Mars?
Ted Wallingford posted a fascinating article, EverywhereNet is on peoples' minds, in response to Andy Abramson's discussion of something called Open Net. It's new territory for me, and I honestly don't think I can sum it up for you just yet. But if you are interested in reading about ideas for a better national and global IP infrastructure, I suggest you read them both.

Forget iPhone?
Cynthia Brumfield points to David Pogue's NY Times discussion of the rumoured Apple iPhone (not to be confused with the SIP-based iPhone from Teledex). Basically, forget about it.

Jajah Mobile Suite No Threat?
Russell Shaw weighs in with his nine reasons why Jajah Mobile Suite is no threat to cell or VoIP. Common thread: people with existing cellular or VoIP subscriptions are unlikely to switch.

SJSU OK's Skype
Phil Wolff reports at Skype Journal that, reversing an earlier decision to ban Skype on campus, SJSU (San Jose State University) has okayed the popular soft phone for now. The post also has a nice summary of the reasons why Skype would be beneficial to the university. Bravo on both counts.

VoIP Crimes Of Another Stripe?

After the arrest of five foreign nationals in Namibia providing VoIP service without a license, as well as goings on in various Asian and African countries in regards to VoIP, you might be wondering if VoIP is under attack there. Marcelo Rodriguez takes a crtical look [Voxilla] at what Russell Shaw [ZD Net] and Rich Tehrani [TMC Net] are saying.

Rodriguez points out that both Shaw and Tehrani mention "Third World" countries as locales where VoIP seems to be under attack, possibly due to affiliations between the government and the traditional telecoms, but that they leave out the US as being in a similar category. (Examples: Korea and the UAE blocking Skype.) He then goes on to reveal several examples of lobbying, campaign contributions, and all-expense golf vacations.

The Voxilla piece is very revealing and extremely politically charged. I'm going to take my cue to up the voltage. Let's take a few separate scenarios. First scenario, conspiracy: the entire telephony system in North America is fully wiretapped and all calls are monitored either by humans or machines, for whatever political purpose the real men with power wield. Second scenario: the first scenario is crock, but phone calls are a valuable commodity and thus extremely lucrative. Third scenario: a combination of both the first and second scenarios.

Choose your scenario. Either way, VoIP threatens the status quo, and hence spawns acts like CALEA, possibly attacks on Vonage's share price, and debates like neutrality vs tiered Internet service. Everything that is happening politically in telephony satisfies one of those three scenarios. Let's face it: VoiP is a threat no matter how you slice your political pie.

VoIP Roundup - Mon Sep 25/06

Universities Banning Skype
A number of universities have decided to ban Skype, stating that it consumes bandwidth and supposedly is an "illegal" waste of resources. (Illegal? Seriously?) Grid computing apps are also included in the ban at several California universities including University of California Santa Barbara, San Jose State University, and California State University Dominguez Hills. They are not banning Gizmo Project or Wengo. [via Ars Technica] Several countries also ban/ block Skype, including Korea and the UAE (United Arab Emirates).

Telrex CallRex VoIP Call Recording For Cisco
Telrex claims that their CallRex version 3.1 is the  first VoIP call-recording solution to be certified for encrypting Cisco Unified CallManager 5.0 calls. [via Business Wire]

SIP Trunking Makes VoIP Telephony More Flexible
Rich Tehrani reflects on how SIP trunking has made IP telephony more flexible by reducing the amount of proprietary hardware. He points out that not all IP PBXes are connected to SIP trunks; that over half of them (some used by IP-based call centers) are still using PSTN trunk lines.

Could Alarm.com Help Pure Play VoIP?

Yesterday, I reflected briefly on Alarm.com's signing of SunRocket as a partner, who follow Vonage, the first to sign. Like other competitors InnovAlarm and NextAlarm, Alarm.com offers home security alarm services using VoIP for the communications component. Garrett Smith thinks that this sort of additional VoIP-based service might give pure play VoIP providers such as SunRocket and Vonage a bit more competitive edge, by offering their own "triple play" of services, compared to broadband providers.

While I vaguely said something similar, I still feel that cable providers have the edge, especially considering that Alarm.com is target this type of VoIP provider next for partners. PhoneBoy weighs in, basically agreeing, and says any advantage will be minimal because pure plays won't be offering anything unique.

That doesn't mean that there isn't any market for pure play VoIP. Anyone who does not have or want a computer (there are such people) might prefer to opt for a service such as Vonage, were they forced to change from traditional telephony. Or simply want to save on long-distance calls.

VoIP Systems Subject To Security Risks?

Some experts are saying that VoIP in the enterprise represents serious security risks [CIO], making a company vulnerable to vishing (phishing via VoIP) attacks. One anonymous security researcher claims that bank networks will be subject to penetration and the phone lines to hijacking - thus leading to the theft of credit card numbers and bank account data.

Now I'm not a VoIP security expert, but I can make an educated guess, based on my many years of computer experience, that this guy, who goes by the pseudonym "The Grugg", is grossly exaggerating the security issues, potentially to gain some attention. It's absurd to think that banks, who have been dealing with electronic security issues for several decades now, would even think to put their data and VoIP networks on the same lines. Besides telecoms, I've worked at a big mutual fund company. Even they had backup and redundant networks, with firewalled access to account information.

While it's likely true that little technology exists at present to filter out vishing attacks, there's nothing that says a bank's data network has to run on a VoIP network. And just because a bank's telecom system is converted to IP telephony doesn't mean the data network is suddenly at risk. In fact, if someone wanted to mount a vishing attack on a bank, they could do so already using an existing VoIP system (sorry, not going to tell you how). And they wouldn't have any more or less success than if the bank had a VoIP network or not. (On the other hand, a VoIP phone system could potentially be taken offline by a DDoS (Distributed Denial of Service) attack if a load balancing system is not in place.)

Despite what The Grugg (give me a break) is saying, I'm not so sure that bank data networks are at risk. Of course, I could be proven wrong, but let's hope I'm not, as this expert is saying that vishing attacks on banks will probably start later this year. I wonder how he knows this.

Alarm.com Signs Second VoIP Partner

Alarm.com, a home security service that uses VoIP as a means of communication, signed their second partner, SunRocket. Earlier this year, they announced their first partnership with Vonage. [via PC Mag] Vonage recently announced that they would be providing optional VoIP installations courtesy of a third party. Now if Vonage is smart (or maybe Alarm.com?), they'll work out a package deal for customers whereby they can have both their VoIP and Alarm.com's security system installed simultaneously.

While both current partners are "pure play" VoIP providers, Alarm.com is also aiming at partnerships with ISPs next. ISPs, of course, offer "triple play" services: television programming, Internet access, and VoIP. The PC Mag article talks about quad-tier services, which would include cellular phone service, and refers to Alarm.com's service as being like a fifth tier for ISPs.

A similar VoIP-based security alarm service is being offered by InnovAlarm, who will be getting US$10M in venture capital.

Cellular VoIP vs Fixed VoIP

VoIP Central (via EFY Times) indicates that revenues from mobile VoIP will outdistance that from fixed VoIP in Europe and the USA, but especially in the latter, by 2012.

I think that there will be a fair bit of growth in business use of fixed VoIP, especially since communications costs for SMEs [VoIP Central] is expected to be reduce by 20-40%. This cost reduction can be enough to save a small business. As for cellular VoIP, there are a few solutions that work like a charm, but it has farther to go than either true (hard) VoIP (e.g., Vonage, SunRocket) or soft VoIP from a desktop or even laptop. However, SME and SOHO owners who see the benefit of fixed VoIP are likely to add cellular VoIP to their business comm toolkit, and then tell their friends about how much money they save. Hence, eventually cellular VoIP can most definitely outsell fixed VoIP.

Beyond 3G Communications

Web sites have already been throwing around the term 4G (fourth generation) as the successor to 3G mobile communications technology. But a bunch of biggies in the industry had a jam session recently to decide exactly what 4G should have. Silicon.com has a brief look at their ideas.

I'm thinking 4G will include GPS abilities for sure, as location-aware phones [Silicon.com] are expected to be a Euros 622 M business in Europe by 2010. In fact, GPS will be part of 3G phones as well, what with Nokia buying Gate5, a mapping company, and surveys showing that cell phone users want such features. But with Samsung already testing 4G tech, it'll be very exciting to see what else such phones will have - hopefully dual mode VoWiFi and cellular calling,

I suppose, however, that companies like T-Mobile will have to be convinced not to ban VoIP from its cellular wireless network.

VoIP Thief On The Run

Steal VoIP, go to jail. Or if you're Edwin Pena, barely out of his teens, you go on the lam, possibly using your 40-foot speed boat, which was paid for by resold stolen VoIP service. Pena was arrested by Miami police a few months back, along with his buddy hacker. They supposedly stole and resold around 10 M minutes of VoIP service and were facing up to 35 years on a couple of charges. Pena skipped bail and is suspected of heading somewhere from where he can't be extradited. Time to bring in the CSI: Miami crew, though I'm not sure they've covered any telecom crimes to date.

These two guys are obviously bright minds, given the way they engineered their whole set up. Had they thought just a bit further, they could have been doing VoIP security consulting and making good money, instead of doing time. Given the shortage of skilled workers in the IP telecom industry, it's a waste. A good mind is a terrible thing to waste; a good VoIP mind even more so.

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.]

VoIP Roundup - Fri Sep 15/06

President Asks For Warrantless Wiretaps
US president George Bush is asking for warrantless wiretaps, particularly in relation to prisoners held at Guantanamo Bay. [via CNBC TV] Recently, US District Court Judge Anna Diggs Taylor ordered a halt to the wiretapping program, concluding in her report that warrantless wiretapping is unconstitutional. CALEA allows a backdoor for law enforcement agencies to wiretap calls if public security is threaten. However, the wiretapping program in question was secretly signed by President Bush in 2001.

Telus Corp Wins 5-Yr Telecom Contract
The government of the Province of Ontario (Canada) awarded Telus Corp (second-largest Canadian phone company) a five-year, Cdn$140 M contract to manage and supply various network services, including IP communication. [via CNW] Telus recently announced that they were converting to an income trust.

Yahoo Messenger Plugins: Pandaf Sudoku Battle
Not sick of the immensely popular Sudoku number puzzles? The Pandaf Sudoku Battle plugin for Yahoo! Messenger 8 lets you battle against an opponent. I assume you race to finish first. This is of course quite the variation on the puzzle, as it's traditionally a one-player challenge.

Stratus Techologies Acquires Emergent
  Stratus Technologies announced the US$10 M buyout of Emergent Network Solutions [Extreme VoIP], a VoIP infrastructure company.

Free Muni Wi-Fi In Pittsburgh

Pittsburgh officially has free municipal Wi-Fi in the downtown area as of yesterday at noon. The initiative was started by Mayor Bob O'Connor, who passed away due to brain cancer on Sept 1. It took US Wireless Online only two months to build the 60-site network. The FreeConnect service option allows a maximum of two hours per day at no cost. DayConnect offers faster access at US$7.99/day, $14.95/m, or $119.99/yr. VoWiFi (Voice over municipal Wi-Fi) service is planned for the future. [via Pittsburgh Business Times, Business First]

I don't know if that's a record implementation time for municipal Wi-Fi, but it sure sounds like it is. Wow.

What Is Mobile Phone Number Portability?

While others are frustrated with their inability to move their VoIP, phone, or cellular numbers to a new provider, and industry steering committee in South Africa is asking their telecom regulator to delay their MNP mandate [Cellular News].

MNP, or Mobile Number Portability, would require changes in telecom company business practices. Such practices would, in this case, allow customers to port (move) their cell phone numbers with essential ease.

I think that we'll see more and more countries bringing about MNP regulations. In fact, it probably behooves cellular operators and providers to participate, to get started now, if they don't want to see a mass migration towards VoWiFi-only (VoIP over Wi-Fi) phones. While that's not a reality just yet in most cities, the increasing number of municipal Wi-Fi projects (free or otherwise) is going to help.

Personally, I think dual-mode cellular/ VoWiFi phones are going to boom in sales when that happens. So users would get a cellular signal in areas without Wi-Fi Internet access, and VoIP calling otherwise. If number portability is in place in those areas with muni Wi-Fi, then everyone wins.

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.

Mobile Phone Number Portability

Phone number portability is becoming more of an issue for people who tend to move around a lot, obviously. Tom Keating recently talked about his frustrations of moving his stationary phone number to a different VoIP provider. Imagine the problems of trying to move your mobile number. (Everytime I've changed cellular providers, I've had to get a new number.)

In Japan, mobile operators will be allowing cell phone users to keep their phone numbers [VNU Net] when they switch providers. This government-mandated option will not officially take effect until late October, but millions of switchers are expected. Part of the reason for this is that Japan has a very high percentage of mobile phone users (97 M) compared to the population size (127 M).

Portability in this situation is actually beneficial to mobile service providers because of the saturation. Customers have the advantage, but providers also benefit from customers who switch - since the chance of gaining new customers is reduced. Everyone's happy. Hopefully.

Social Networking For Networking + Communication Types

WIP Connector is a website that hooks up business partners focused in the areas of 3G, RFID, Wi-Fi and WiMax communications. It was launched by the Wireless Industry Partnership (WIP) and is designed as a social networking site. [via Silicon]

Full membership costs US$300; a partner membership costs $200 but is currently limited to MX Alliance, Ottawa Wireless Cluster, and WINBC (Wireless Innovation Network of BC) - at least two of which are Canadian. Let's see. A Canadian website (I think) charging US dollars for membership and being promoted on a British website (Silicon). That's IP communications for you: enabling the global village.

Things To Think About When Signing Up For VoIP

I'm linking to Russell Shaw again: he points to Tom Keating's recent frustrating experience trying to move his Vonage phone number to his cable provider. Tom encountered technical problems as well as what he believes are political issues. I'm not entirely clear what he did finally with the phone number, but he did get fed up and dropped his cable provider to go with a more flexible company. Good for you, Tom! Read his article for a synopsis to determine under which scenarios you can move your phone number around between VoIP providers.

While I do 98% of my calling on soft VoIP clients or sometimes my cellphone, I haven't yet signed up for a VoIP phone number. I did, however, get a free call-in number when I signed up for the Hullo VoIP service - which is like a combo of Jajah and Skype. I was actually able to call my computer from my cell phone with it, and as far as I was able to tell, the quality was pretty good. I didn't try it yet, but I'm sure that I could call my Hullo client from Skype this way.

As for cable providers, I have one and only one option because of where I live. But I suppose if I wanted to have VoIP over cable using a regular handset, I could get an adapter and plug my broadband connection through it. Why unnecessarily pay for an additional service I won't need? I'm a very nomadic person and have lost track of how many phone numbers have been registered under my name over 15 years, and in how many cities. I need some sort of global, portable calling number, which only soft VoIP seems able to support at present. Any such number will always work no matter where I am.

What you ultimately choose for your VoIP service will depend on your options and your needs. I work all day at two computers, so I don't need/ want to pick up a phone handset unless necessary. (Haven't owned a landline in about 10 years now.) But for people who want the convenience of a regular phone and the low price of VoIP calls, and who have a broadband connection, a plug'n'play adapter is probably your best best.

If you're prone to being nomadic like myself, check to make sure that you'll be able to transfer your phone number. Ask three different people at the VoIP provider you've selected, and if you don't get consistent answers, run away. Or for convenience, consider a triple-play package from a cable provider. The bigger they are, the more likely they'll be able to pull strings to port the number to your new location.

China's Mobile and Broadband Markets Likely To Exceed India's

Recent talk was that India's mobile phone market would be the largest in the world. But not surprisingly, China might exceed that. The 400 million mobile phones they'll produce this year make up half the world's output and will be used in other countries, but they could just as well be used there, too.

As for India, its come a long way. One East Indian friend joked to me that in India, even the janitor has a cell phone. Which was not meant to be derogatory, but to indicate how far the country has come. Not so long ago, it had daily power outages, but now has the juice to drive cellular networks that include everyone in several financial classes.

However, with the increasing number of middle-class citizens in China, it's more than possible that the Chinese mobile market may exceed India's, where they're focusing on IPTV for some reason. The VoIP market in Asia in general is growing. Though with issues such as VoIP service being illegal in China, I'm not sure if certain types of phones and PDAs are allowed in the country or not.

Even if China and India actually run close numbers for mobile use, broadband use in China is growing at about 80% annually and expected to reach 130 million users by 2010. Part of the increase will be a side effect of hosting the 2008 Summer Olympics in Beijing.

Given the political situation in China, and the fact that VoIP is illegal there, it might be difficult to understand how that government would allow the estimated 80 million users playing online games. When you run a country banning the use of certain words in print or online, it's hard to let any sort of digital interaction go unmonitored. This sort of atmosphere would permeate into a lot of things, including the way events are handled and technologies deployed.

However, dig deep into the history of the Olympics during the time that Juan Antonio Samaranch was top dog of the IOC (International Olympics Committee), and you'll see that the Olympics actually were repeatedly granted to countries and regions where there was political, civil, and/or social unrest,. The net result of hosting the Olympics in those locales actually improved conditions considerably.

Whether or not this happens in China, resulting in more open government policies, remains to be seen. But if it does, China will likely dominate in Internet use whether, whether mobile or stationary, and there will be an explosion of VoIP services and possibly some innovations.

Are Telcos Getting Short Shrift On VoIP?

Canada's CRTC (equivalent of US FCC) ruled last week that they would be regluating VoIP service in Canada, basically reinforcing their decision from May 2005. The big telephone companies in Canada were unhappy with this decision because it prohbits them from offering VoIP services below cost, as a loss-leader for other packages. On the other hand, apparently new VoIP companies can set whatever price they like. Furthermore, telecoms have to file a tariff for VoIP services, whereas cable companies do not. I'd assume soft VoIP providers wouldn't have to, either.

The odd thing is, Canada has a law in place since at least the 1970s, if not earlier, which prohibits any business from charging different prices to different clients for the same service. It appears, however, that the inverse is not true. While I have no love for telecoms, and have worked for some, on the surface, this decision would seem some what unfair to the telecoms. On the other hand, it gives VoIP startups a chance to compete against otherwise incumbent companies with deep pockets.

Where I don't think the ruling is fair is that cable companies get a leg up. If you know Canadian cable industry history, you know that in many areas, cablecos owned regional monopolies on service. When the giant Rogers Cable started buying out smaller regional cable providers, in at least the province of Ontario, in the late 1970s and through the 1980s, they became more powerful and allegedly incredibly uncaring about customers. At least, that's what I hear from Canadian friends, acquaintances, and insiders. Rogers Cable also owns mobile phone services that are competitive with Bell Canada's Bell Mobility division. So why Rogers Cellular, for example, should get an advantage in VoIP services over Bell Mobility, I'm not sure. They are probably about par in their power and resources.

In this situation, both should be shackled equally. Dare I say it: are the telcos getting the bum's rush? Gazing into my crystal ball, the worst case scenario shows that traditional telcos will crash and burn in the next 10 to 15 years, while cable providers will enjoy the richesse of exponentially increasing triple-play subscribers. But then again, my crystal ball is bit spotty.

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.

VoIP Call Quality Now Better Than PSTN?

ITWire has a story which quotes a testing company named Minacom. Minacom is claiming that VoIP phone service "now sounds better and connects faster" than PSTN phone service. This is based on data they collected over twelve months, and only applies to VoIP services offered by cable providers and telcos. The test uses a standard measure called an MOS (Mean Opinion Score). Minacom's test contradicts Brix Networks' recent report saying that quality is declining. However Brix measured opinion on soft VoIP and pc2pc only calls.

Having tried only soft VoIP services, I can't comment on Minacom's findings, except to say that I can see how phone2phone VoIP calls, using a plug'n'play adapter and a broandband internet connection, would be fairly high quality. As for soft VoIP, it's definitely not true. Not in my experience, anyway. Basically, the more software of any type that you have running on your computer, the lower your call quality is going to be.

As laptops tend to have less RAM than desktops, they are the worst for call quality. That's true even if one party in the conversation has a powerful desktop, as I recently found out when calling a friend on his laptop. My laptop with 512 Mb didn't fare much better, unless I pretty much closed all programs. Which is why I switched to making most of my calls on my desktop. High soft VoIP call quality requires optimum computing power.

My experience with my desktop (1 Gb RAM, dual processor) is that pc2pc calls are almost as high quality as regular phone2phone. (As I've said, I haven't tried a VoIP adapter or VoIP phones.) It's when there's a mix of pc and phone in a VoIP call that quality seems to go down. However, according to a couple of people that I've called on both Skype and Hullo, Hullo calls were almost as if I were calling from a regular phone.

So quality from soft VoIP services seems to be increasing, but I think VoIP as a whole has a ways to go yet. Better quality VoIP phones and faster connection speeds would make a difference. We might even find faster microprocessors in VoIP phones, or special VoIP-dedicated chips in the next generation of computers, just like graphics cards were eventually dedicated to computer screen management. A dedicated VoIP computer chip, either in computers or phones, would go a long way towards improving call quality. (If there are VoIP-dedicated chips, I'm not aware of them. Let me know.)

Google Talk Hearts Skype?

Google's Talkabout weblog, the official blog for the Google Talk IM client, has a posting about the announcement between Google and eBay (Skype's parent company). There's also a little blurb about exploring "interoperability between Google Talk and Skype". Yeah! Google Talk is based on an open standard, whereas Skype is not. This should get very interesting, with all these IM client pairings.

Russell Shaw at ZDNet, however, thinks Google's not interested in the full potential of Internet telephony for Google Talk. His reasoning, from reading between the lines of the deal, is that Google is saying that they have "relatively modest plans for Google Talk." I have to agree that Google has yet to make Talk a full-blown VoIP IM client. It can't even call out to regular phones without the help of software like Vozin Communication's Talqer. As for whether Russell's right, we'll have to see. Though I think that the higher ups at Google often use the "mystical warrior" philosophy to mislead us on their real intentions :)

VoIPcasting: Recording VoIP and Podcasting

If you're running Skype and want to record your conversations, VoIP-Sol lists 15 voip recording applications (10 for Windows, 5 for Mac) specifically for Skype. If you are using something other than Skype, there's our posts: Recording Your VoIP Calls and How To Record VoIP Calls - Reader Q+A.

What you do with your recordings is your business, but if you plan to podcast them online and have or plan to use Asterisk IP PBX, here's Nerd Vittles' lowdown for a podcast studio using your phone and a free podcast hosting service called Gabcast. Gabcast lets you record podcasts from a phone or using VoIP.

You can actually use Gabcast from any soft VoIP client that has pc2phone capabilities, so you don't really need a sophisticated setup to make a podcast. I used a cheap microphone and Skype. As long as you follow the rules for good VoIP quality, your VoIPcasts will be of reasonably good quality as well. If you're opting for a very professional production, there are all kinds of audio equipment you could look at, and which I might discuss in the future, if readers are interested.

Ubiquitous Streaming Video On Your Cell Phone

So, you're walking by a billboard for a new TV show that looks interesting. You want to write down the details, but don't have pen and paper handy. Your head is too full of other things to remember unaided. So what do you do? Pull out your Bluetooth-enabled cell phone or PDA, point it at the billboard, and download a 30-second video clip that has all the information you need. No fumbling for a pen. [via Telecommunications Industry News]

This is a real scenario that the United States' CBS television network has created in some New York City train stations. You can stand up to 36 feet away from these special billboards and download clips, provided you have a Bluetooth-enabled smartphone or PDA. Telly junkies like me will no doubt be happy with such uses of streaming video. I can't count the number of shows I've missed over the years because I dislike watching a new series from the middle, due to missing the first few episodes. I'd rather wait 3-5 years, when it goes into re-run syndication, to watch the series in sequence. [I admit to being a TV junkie, but I do write about media, so it's kind of a necessity.]

This is different, of course, than IPTV (Internet Protocol TV), mobile TV, video VoIP, or VoIP interaction with TV characters. What I'd really like to see, though, is if I could download TV show information from these billboards, then transfer it from my mobile phone to my IPTV setup to let me pre-program my software to record the show to my hard drive. Now that would be a video junkie's dream come true. How long do you think it'll be before someone comes up with this sort of thing, if it's not already available?

VoSKY Marries Skype and IP PBXes

Who said Skype isn't ready for enterprise? VoSKY thinks otherwise, and to prove it, is offering an actual Skype-certified solution that lets you use Skype with a PBX. Aimed at SMBs (Small and Medium Businesses) of 10-300 employees, the device bridges between a PBX and a Windows XP computer. [IW Distribution/ VoSKy via Asterisk VoIP News]

The IW Distribution press release actually claims VoSKY's device is "the World's First Skype Solution for Business". IW is only promoting the product in the Australian and New Zealand markets, but no doubt other distributors will carry it in other parts of the world, if they aren't already. (I'll do some digging and find out.) Though I'm not surprised, since the SMB and consumer VoIP market in Australia is expected to build to 6 million by 2011.

I wish I could get my hands on hardware like this, to try out. It's likely the beginning of a series of offerings that leverage the cheap pc2pc and pc2phone calls of soft VoIP clients, like Skype and Gizmo Project, for use in even more sophisticated CRM VoIP applications. Businesses need more VoIP products like this.

VoIP Roundup - Tues Aug 22/06

SMBs Becoming More Aware Of VoIP
According to a Q1 survey by Savatar, around 30 percent of SMB (Small and Medium Business) companies are either familar with VoIP or already converting/ converted. [via TMC Net] This of course spells good news for VoIP providers, system integrators and hardware makers.

African VoIP Developments
Kenya's Information Minister, Mutahi Kagwe, thinks that using VoIP could reduce the Kenyan government's phone bill by up to 70%. [via Capital FM] Given that various government ministries owe Telekom Kenya several billion shillings, VoIP thus seems an appropriate solution. They might take a cue from the Taipei City government and consider implementing a muncipal Wi-Fi network over which they could conduct VoIP calls.

In Nigeria, Dr. Ernest Ndukwe says that VoIP is "the engne of telephony in developing countries". The EVC (Executive Vice Chairman) of the Nigerian Communications Commission was speaking at a VoIP conference in Lagos. [via allAfrica]

Australian TAFE Colleges Going VoIP
A group of Australian TAFE (Technical and Further Education) colleges are switching their telephony system to VoIP. Approximately 4,000 VoIP handsets will be part of the migration. Some of the colleges already have VoIP in place, while others are still being converted. [via ZD Net Australia] Cisco is the project vendor.

Enterprise: Choosing Between Hybrid and Pure IP VoIP Systems

Telephony has been moving from PSTN/ POTS systems to hybrid IP-PBXes, as far as enterprise VoIP systems go. But some people expect that hybrid VoIP systems, which support TDM and IP calls, will be outdated in just a few years, supplanted by pure IP that is well integrated with data applications. [via Datamation]

Thus, companies who are currently planning a switch to VoIP need to consider what sort of system that they want to go with. If this is the situation you find yourself in, ask yourself how you to plan to use VoIP. If your business cannot benefit from integrated data applications, then a hybrid system is probably sufficient. But if you want to be able to build, say, a sophisticated CRM (Customer Relationship Management) system, a pure IP system is the direction you should consider.

New DSL Service Charges For Verizon Customers

Verizon has decided to charge their DSL high-speed Internet customers an extra US$1.20/mth (for access speeds up to 768 kbps (kilobits per second) or $2.70/m (for faster speeds). This will probably come as a surprise to customers who were expecting lower monthly bills based on a FCC decision last year to deregulate DSL (Digital Subscriber Line) service. [via CRM Buyer]

The old government fee for the USF (Universal Service Fund) is being phased out. So Verizon and no doubt other cable providers are taking advantage of it by imposing their own fee. Of course, they're claiming that this new fee has nothing to do with not having to pay the USF fee anymore. Great to know that they're thinking of us.

So what gives? Isn't their monthly service fee enough? Are they really not making any money? (Cable providers are.) Could this new fee be due to expectations that free VoIP over DSL phone calls will clog up their lines? Hmmm. Read between the lines in the CRM Buyer article, and that's what it seems like.

It'll be interesting to see who the next DSL provider is that applies a similar charge to customers' bills, and whether there will be a shift to cable services.

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.

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.

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.

Cable VoIP vs Pure Play

Apparently cable VoIP is giving providers such as Vonage a run for their money. That's because for a few dollars more per month, customers are getting cable (data + video) plus telephony, as well as other features that pure play VoIP cannot provide. [via CED Magazine]

This IMS (IP Multimedia Subsystem) architecture promises far more than pure play VoIP. One of the most important expected features, to some people, will be a global phone number, which can be used anywhere and can be called from anywhere.

While Vonage is still in the lead in volume, it's probably due to the extra features why cable companies are leading in new VoIP subscribers, and why one company, Time Warner, isn't far behind in total subscribers.

On the other hand, I know people who do not watch TV or use the Internet, but do have a need for a telephone, without any "global number" feature. There's always a market for basic telephony. It just may no longer be worth the amount of monthly advertising that companies like Vonage are said to spend (US$20M).

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]

VoIP Roundup - Wed Aug 09/06

Jeff Pulver (Pulvermedia) and Paul Kaputska have just launched Vonosphere, a website dedicated to voice- and video-on-net news. Jeff is a very proactive person who writes letters to politicians regarding issues such as net neutrality. Congrats to both of you on the new site. [via IP Inferno]

The pricey (US$350) new Mylo from Sony, like the HyunWon Boxon, is a combo consumer electronics gadget. But it actually has VoIP, in the form of Skype. [Sony via Engadget, The VoIP Weblog]

Vonage has come up with a way to offer real e-911 access to their customers. The service ties the caller's phone number with a phsyical address. [via Xchange Mag]

Cindy Waxer of TMC Net says that VoIP job opportunities abound. So, she says, forget about become a doctor or lawyer. Apparently,  the second-fastest growing occupation is in the area of network systems and data communication analysis, right through to 2014. Job numbers will increase by 55%.

VoIP in the enterprise to date relies on WANs (Wide-Area Networks), but deployment's been a headache for some network managers. That's because VoIP is a demanding application in terms of network usage and traffic patterns. [via TMC Net] An alternative is to deploy VoIP over MPLS (Multi-Protocol Label Switching) networks.

Meebo Meet My VoIPpleganger

A couple of early Friday mornings ago, something strange happened to me, just before I went to bed. Skype messaged me saying someone wanted to add me to their buddy list. Okay, that's not strange, but the person had the same initials and last name as me. Had I pressed some strange combo of ALT-keys? No, this was a real person, and they lived half-way across the world. Some sort of VoIP doppleganger - a VoIPpleganger, maybe?

Then, after a few hours of sleep, I'd barely un-hibernated my laptop when someone from the Ukraine IMed me via the Gaim text IM client, on my AIM/ICQ account. Gaim was running for one business contact, Google Talk for another, and Skype was running for the free SkypeOut within Canada and the US. Gizmo Project wasn't running, and I hadn't installed Yahoo! Messenger 8 yet. And I'd just shut down the Trillian IM client because Gaim could manage the same accounts.

The moral of the story? Arrrrrrggggghhhhh. My laptop's RAM is always maxxed out because of all these damned text/ VoIP IM clients running for different biz contacts. Are you going crazy trying, too, trying to manage a handful of software clients?

Fortunately, a number of companies are making an effort to either be compatible with other IM clients, or at least be a bridge. We've already covered a few. For example, the newest Yahoo and MSN Messengers are compatible, and Festoon Unity tries to bridge Skype + Google Talk.

Well here's another one, which bridges four different IMs. And you don't have to download anything: Meebo. Meebo, which is AJAX-powered, works in your web browser, sort of like a virtual desktop. It bridges AIM/ICQ, Jabber/ Google Talk, Yahoo! Messenger, and MSN Messenger IM accounts. You get little dialog windows that you can move around, and all of your contacts from all of your above-mentioned accounts sit in one handy little window, with different icons to differentiate their source IM.

Meebo Me is a separate web-based service which lets you place a chat box/ shout box on a website/ weblog, and probably aimed at the various social networks. I haven't tried it because I already have an abundance of shoutboxes on my sites.

Meebo itself doesn't handle VoIP, nor Skype or Gizmo Project clients to my knowledge. But maybe all that's coming. Although it is convenient, easy-to-use, and a good start to IM network compatibility. Enough said; go give it a try.

Is SIP The Building Block For IP Telephony Features?

Many people think that the SIP-based technology will be the core building block for future enterprise IP telephony networks, especially for call control protocols and offering features for VoIP that are already present in existing telephony systems.

SIP, or Session Initiation Protocol, is an IETF proposed standard to manage online multimedia sessions that include video, voice, IM, and more. As such, there is expected to be strong interest in the SIP market in the near future. [via CNS Magazine] Light Reading has published a new report studying the SIP market  (US$900).

VoIP Roundup - Tue Aug 08/06

Looking for work in the VoIP field? 2it Consulting is looking for a Pre-sales Engineer with Cisco VoIP/ IPT (IP Telephony) experience for one of their clients in the Sydney, Australia area. [via IT Wire]

Jajah has added Australia and New Zealand to its list of free-call countries that can have free phone-to-phone calls using Jajah's VoIP bridge. [via m-net]

The Philippines government has an interest in VoIP and wants to build intranets for its use. Several government agencies are said to be buying switches for installation. Once the VoIP intranets are built, the next step will be to hook into commercial telephony networks in the Philippines, but not until they offer VoIP services as well. [via Inq7] This is an interesting approach, and one I assume the VoIP-over-municipal-WiFi project in Taiwan is considering as well.

Apparently, Skype will have an official version available for MS-Windows Smartphone 2003-based mobile smartphones/ PDAs. (Note: there is already a Skype client for Windows Pocket PC-based devices.) The bonus for owners of dual-mode phones is that they'll have a choice of Skype over either Wi-Fi or 3G - a sort of DIY converged service. [via Red Herring] It'll be interesting to see how fast they'll come out with a Mobile Linux version, once Mobile Linux for PDAs actually exists.

First there were Skype-certified Wi-Fi phones, now Alpha Networks is offering Google Talk-enabled Wi-Fi phones. Google's GMail will also be supported. [via Asterisk VoIP News]

Keeping Secrets In The Open Using VoIP

Hackers-cum-researchers performed an interesting security-testing experiment earlier this year using VoIP phone numbers and Internet social networks. They presented their findings recently at Defcon.

Their primary plan was to determine if secret signals could be passed right out in the open, from enemy agencies to their agents. They theorized that the use of social networks to transmit carrier messages might increase the noise ratio so that it would be harder for "unauthorized parties" to decode the secret but publicly-transmitted messages.

This is in fact a technique already used covertly by intelligence agencies. However, they use shortwave numbers stations, and all governments have denied such operations. The general technique is to broadcast streams of seemingly nonsensical numbers or words, often in a female or child's voice. Of course, the stream represents a code, and only a few parties have the cipher to decode it.

Strom Carlson, a security researcher, and the hackers collective Project Evil teamed up to see if someone could do the same thing using the Internet, particularly using any of the abundant social networks out there. What they did was set up their own numbers stations. But instead of using shortwave transmissions, they used VoIP phone numbers and recordings. If you called such a number, you would hear a stream of code words. They advertised the existence of the VoIP numbers stations using Craigslist pages, using fake messages, to see if anyone would participate.

In short, they were successful getting others with a cryptographic interest to participate and decode messages using a one-time key. They figure enemy forces could be too. This is something proponents of CALEA may want to take note of: if hostile parties want to use VoIP, they are not necessarily going to use unencoded messages. (On the other hand, this experiment by Carlson might just give CALEA proponents more fodder.)

CALEA stands for Communications Assistance for Law Enforcement Act, and, in short, gives any Law Enforcement agency the right to wiretap communications networks, including the Internet and VoIP, in special circumstances. Although to date, it's not on the agenda to tap soft VoIP calls using clients such as GoogleTalk and Skype.

Of course, there are those people that believe that email spam is being used as numbers stations for intelligence communications. Although who is behind it is hard to say. (I particularly notice some interesting word patterns in the spam in my university alumni email account.) Public key cryptography concepts date back centuries, and the Internet is a perfect distribution vehicle. Just never thought VoIP could be used as a supplementary broadcasting outlet.

Additional sources: Slashdot, Homeland Stupidity, Defcon.

VoIP Roundup - Mon Aug 07/06

According to a recent IDC report, Microsoft views VoIP as a very profitable revenue opportunity, and their iniatives will be disruptive for the next few years. Part of their plans include challenging PBX and IP PBX vendors. [via Businesswire]

As businesses and individuals move towards a digital media convergence, in terms of network infrastructure, for voice, data, and video networks, security is going to become more of an issue. Security Park recommends that vendors wanting to enter the VoIP security space should work closely with end-user focus groups. [via Security Park] They have a VoIP security special report (US$1295), in association with Data Monitor, which addresses some of the issues.

Zeus Kerravala, VP of Enabling Technologies, Yankee Group, spoke at TMC's VoIP Developer show, stating that the "low hanging fruit" of the VoIP market "consists of softphones, call centers, and the convergence of VoIP and mobile devices." As well, he suggested that companies focus on ROI (Return on Investment), not TCO (Total Cost of Ownership). I'm thinking that eBay already thought of that when the spent US$2+ billion buying Skype.

VoIP traffic volume on telecom networks is expected to double during the next 12 months. As a result, call quality may get worse. The solution may be new SIP-based services enabled by IMS (Internet Multimedia Subsystem) upgrades to telecom networks. [via VNU Net]

The Prairie Island Indian Community in Minnesota, USA, is using VoIP for their communications. The solution from IPcelerate will also include a rapid emergency notification system that alerts all 150 community households. [via TMC Net] Glad to see that someone solved the emergency calling problem.

With video-conferencing become a standard feature in the new generation of VoIP/ WoIP soft clients, businesses are asking questions about how and what hardware and software to setup, as well as issues of conferencing etiquette. VoIP.com is offering some guidance in that regard. [via PR Web]

VoIP Call Quality To Landlines Really Does Suck

I've increasingly been taking advantage of Skype's free calling to landlines within Canada and the US lately. To date, I've probably made calls to six or seven people at four different phone numbers. It appears that VoIP calls made to my Internet hosting provider's support line are of the worst quality. At least on their end.

With the exception of one call, I hear the person I'm calling (on a landline) clear and crisp. But last night, while trying to resolve some domain name issues, the hosting company rep repeatedly had to tell me that she was picking up only every other word I said. The conversation ended up taking twice as long as I'd hoped. For Skype calls to other people, though, the callee stated that they thought I sounded distant or maybe in some sort of booth. Quality wasn't great, they said, but it was passable.

On the other hand, a Skype-to-Skype VoIP call with someone half-way across the world was clear as a bell, with a single audio artefact - a slight buzz for a millisecond - and a slightly reduced volume. The other party literally sounded like he was in the same room as me, hence barely a noticeable delay. The person's voice, however, did drop in volume a few times. Which might have been what had happened with my voice when I called my hosting company.

This all contradicts what I said previously about call quality, supporting Brix Networks findings. I think there are a number of factors to consider when determining what kind of VoIP quality you'll experience. There are ways to improve call quality, but if one party is using a landline and the other a computer, quality may be poor.

My observation so far is that if you want to involve VoIP, pc2pc seems to have the best call quality, provided you have a broadband Internet connection. Phone2phone with a VoIP bridge usually does as well. (I tried with Jajah, which offers free calling between registered users. However, I only called myself, with a phone in each ear, so that's not a true indicator.)

How To Make Free VoIP Calls - Reader Q+A

It's not suprising that a lot of readers of this site ask, via the comments, how they can make a free VoIP call from a specific country to another, where the callee has a phone but no computer. So I thought it'd be worthwhile giving a summary of some of the services that have come out this year, in terms of categories rather than specific software.

First, let me answer the question(s) as simply as possible. There are some countries that are less likely to have free calling between PC and PSTN/ mobile phones - not that I've seen. Two of those countries are India and China. Maybe it's because they're the two most populous countries in the world, and few companies want to give up the potential market share.

The only exception I've run across is Jajah (see below), which is currently offering free landline and mobile calling to/from China. But if you run their trial, you can also make a 5 minute call to/from India as well. If I find any other VoIP services that allow free landline calls to/from India or China, I'll write about it on this site.

On the other hand, if both the caller and callee have a computer with a non-dialup Internet connection, you can make all the free VoIP calls you want, between any two countries, with pretty much any softVoIP client. For example, Skype, Google Talk, MSN Live Messenger, Yahoo! Messenger, Gizmo Project, etc.

If you want to call or receive calls on a regular phone via a VoIP network, there are SIP-based adapters (hardware). You'll still need a broadband Internet connection, but won't need a computer. But how many people have a broadband connection and no computer? Not many, I'm guessing.

For many countries, there are a few options for free PC-to-PSTN (PC-to-phone, PC2phone) calls, occasionally including mobile. Some are time-limited promos, some are permanent offers. Here are just a couple of options. (I'll not cover everything here.)

(1) Skype just finished a July promo for free pc2phone calls from Canada and the US to Mexico, Japan, and the UK. I'm guessing they'll have other country offers later this year. They also have free pc2phone calls within Canada and the US until Dec 31/06. But if you don't live in either Canada or the US, you'll need to pay for their inexpensive SkypeOut service, which lets you call pc2phone to many countries.

(2) Gizmo Project has a permanent offer that let's you call pc2phone between 60 countries (but not India and China). However, both caller and callee need to register as Gizmo Project users. (This might mean having to download and install the software as well. So if you don't have a computer, you may have to ask a friend. Keep in mind that you are allowed up to, I believe, three phone numbers per registrant, so your friend may not want to help you :)

(3) Jajah allows PC2phone calls free for up 30 minutes. It's unclear exactly which countries are allowed and which are not, as I've read different things. They do have a 5-minute trial call, and their list of countries includes India and China. But when I read their web pages, I see only China included in the 30-minute free calls, between registered users. You should note that Jajah allows you to make phone2phone calls, not pc2phone calls.

This is just a sampling of some of the nice VoIP plans currently available. If you know of others, please feel free to mention them in the comments section. I will try to put together a comprehensive free-VoIP guide, before Christmas time, that points to articles both here and on other websites.

VoIP Roundup - Wed Aug 2/06

MediaRing in Singapore will be offering "prefix-3" VoIP phone numbers. These numbers can receive calls from both PSTN and mobile phones as well. [via ChannelNews Asia]

Verizon had just reported a loss of US$500 million between Q1 and Q2 of 2006. A story in today's New York Times confirms this. Verizon is the USA's No. 2 local phone carrier, just behind AT&T. Qwest, the fourth largest carrier, also reported losses due to VoIP/ Internet telephony compared to a year ago. Both companies stated that increased sales of broadband and wireless services dampened the losses slightly. [Aside: As I was about to post this roundup, CNBC TV showed a video segment about New York's over-taxed electric grid, and how Verizon is helping out with hydrogen cells.]

Now here's a company who understands unlimited Internet usage from a cell phone. The UK's T-Mobile is offering rates of less than a penny per kilobyte, with a maximum charge of 1 GBP (Great British Pound) per day. It's all free after that, for the rest of a day. They're also not limiting what sites you visit, but proof of age is required for access to adult sites. Unfortunately, this service is only available for two phones: the Motorola v3 RAZR and the Nokia 6131, with other handsets promised soon. [via The Register]

Infonetics Research, in their VoIP Services report, says that VoIP service revenue has doubled between 2004 and 2005 in North America, Europe and the Asia Pacific. In these regions, from 2005-2009, It's expected that US$120 billion will be spent on VoIP services. [via Infonetics]

VoIP Inc. has just launched their VoiceOne Lab Development website as a showcase for their new VoIP technology and projects. [via New Telephony] Interestingly, the project page has something called the gTalk Mobile Client. Should be interesting to see who has the rights to that name, as some people use GTalk to refer to Google Talk's IM client.

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.

VoIP Roundup #4

Skype will be getting SMS text messaging services courtesy of Mobile 365. The latter company already delivers 2 billion messages monthly. [via Biz Journals] Skype had already added a free SMS service in early 2005.

The Inquirer (British) thinks that Microsoft's real threat is Skype.

With all the inexpensive means of publishing content, citizen journalism is on the rise. People are recording war footage in the Middle East with their cell phones and posting the content to websites, including YouTube.com, as a way to share what the "camera person" is experiencing. Some even write a description, to express all the feelings. [via SF Gate]

New Zealand's Woosh wireless has broadcast rights from Sky TV to provide Internet TV using its WiMax network. [via NZ Herald] IPTV (Internet Protocol TV) is the next frontier in multimedia content over the Internet, with tests being conducted worldwide, including the US, India, China, and elsewhere, supposedly causing fear in cable TV companies.

Market Clarity, a telecom research firm in Australia, has a free online directory listing VoIP providers in that country. [via IT Wire]

Batelco in Bahrain has expanded VoIP calling from five destinations to over 200. They've also reduced rates for their international VoIP-based calling cards. These cards are valid for calls from PCs, PSTN lines, and cell phones. [via Trade Arabia]

Unlimited Cell Phone Data Plan? - Mmm, Not So Much

Nate Anderson has a great overview of how unlimited "unlimited" 1xEV-DO (=EVDO = Evolution-Data Optimized) wireless data plans on cellular phones really are. He mentions Verizon, but what he says holds true with a lot of providers.

EVDO is the wireless data network that some CDMA-based cell phones use, in a number of countries in Europe, Asia-Pacific, and Canada and the US. (The Wikipedia link above has a fairly comprehensive list of carriers, phones and laptop data cards which use EVDO.)

If you have a smartphone or PDA that uses CDMA, the wireless data plan available is likely to be EVDO-based. If you've purchased, or are thinking of purchasing, an "unlimited" monthly data/ wireless plan, check your provider's fine print. More than likely, there's a bit of text that says you cannot use the service for VoIP, streaming music or video, and several other purposes.

When I called my service provider recently to upgrade to the "unlimited" plan, he told me that not only was that plan grandfathered, it had only ever been available on the laptop data card, not my Palm Treo 650. I insisted that the sales rep had said otherwise, but the rep wouldn't budge. So I ended up witha plan offering only 250 Mb/mth bandwidth, for something like $100/m. Ouch.

Consider that one day, when I had trouble with my regular Internet connection, I used my Palm Treo 650's EVDO connection, via a Bluetooth USB adapter (different from a Bluetooth headset) as a modem, from my laptop. In a regular half-day's web browsing for researching my daily articles, I used nearly 90 Megabytes. In a half day. No streaming music or video. Just one test of VoIP, because I was writing about a service.

Note: CDMA phones cannot use the phone and the wireless connection simultaneously, which usually means not being able to use most VoIP software directly on the device.

Obviously cell phone data plans are not for power web surfers like myself, but it sure would be nice to have options for a bigger bandwidth plan at a better.

VoIP Roundup #2

TechCrunch reports that SightSpeed 5.0 launched. Yet when I checked the Sightspeed site (10 PM Pacific time), there's a message saying you should return at 9 PM Pacific time. The new version apparently includes place-shifted TV, a new video codec, PSTN out- and in-calling, and more. The beta was available a couple of months ago. Sightspeed is a competitor to Skype, but went one step further by incorporating native call recording as well as video calling and free voice and videomail. [Update: 5.0 is now available for both Mac OS X and Win XP. Unlike Skype, Sightspeed is keeping version numbers for both platforms in line.]

GigaOm points to a post on Andy Abramson's VoIPWatch about a new deal between SixApart and Gizmo Project. The new service would let LiveJournal webloggers VoIP and text IM site visitors. This is in addition to the recently announced Jabber-based text IM that LiveJournal would be adding. These kinds of integration of web services are going to appear a lot more often, as innovative companies like SixApart see the value to the end users.

Benjamin Higginbotham presents a compelling argument for why Skype has not won the VoIP battle yet, saying that while it's great in the C2C (consumer- to- consumer) market, it falls down in the B2B (business- to- business) and B2C markets. Skype did announce late last year, and again recently, that they were going after the enterprise VoIP market. That is despite saying their software was not enterprise grade. Nevertheless, I think I have to agree with Benjamin, as Skype (and most other softVoIP clients) support neither VoiceXML nor CCXML, which would really make a difference for businesses.

VoIP Roundup #1

This is a roundup of recent VoIP-related news from various sources.

Skype has just released V1.5 for the Mac OS X platform. New features include a new interface and ability to import addresses. Video support requires a plugin, and call recording is still non-native. [via Pocket Lint, Tech Crunch] Unfortunately, great VoIP recording tools like HotRecorder only run on Windows, at present.

Not sure if they'll be as hot as other cute monster novelties, but Verballs double as a USB-enabled hands-free Skype phones. [via Engadget] Apparently they wave their arms and move their lips. Scary.

The previously announced WiFi Skype phone from SMC is now available for US$199. It'll work over any WiFi connection that does not require browser authentication. [via Market News] SMC is one of four companies that recently announced WiFi Skype phones. Others companies include Belkin, Netgear, and Edge-Core.

A report by Telephia says that pure-play VoIP subscriptions are up but network call quality still needs improvement.

Newsday reports that there are over 1100 providers in the US that offer Internet-based phone services, but other than that, Internet phone service could be the future. Which is what I said the other day - all the more reason for softVoIP network compatibility.

For Better Or For Worse - Is VoIP Quality Decreasing?

Brix Networks, a company who makes monitoring tools to test VoIP networks, says that data collected on their TestYourVoIP website indicates that users are rating 20 percent of nearly a million calls tested as being of poor quality. This data spans 18 months.

My own experience is not substantial, but I say quality is getting better, at least in newer softVoIP clients. Over a year ago, I VoIPed a friend using MSN Messenger, which he was also using. The call quality was terrible. Since then, I've either used or briefly tested Jajah, Skype, and Talqer, all on the same laptop, headphones, and cheapo $1 microphone. Talqer had the best call quality. And I'm using a wireless connection. Direct broadband connections would probably offer the best quality.

It is of course to Brix' advantage to publish such disappointing findings. And if I've understood the BusinessWire press release correctly, the TestYourVoIP service is really measuring broadband quality, not actual calls. So the data might in fact be misleading, considering that there are a lot of other factors to consider in VoIP call quality testing.

What's your experience? Are you finding better quality? If you want to test your VoIP, try Brix Network's Google Gadget, which requires you to have Google Desktop Version 4 or higher.

Sources: ComputerWorld, BusinessWire [via FierceVoIP]

Microsoft's Showing Strong Interest In VoIP

Microsoft is showing its intense interest in VoIP lately by partnering up with not just Nortel but Yahoo as well.

Microsoft and Nortel are working on a new project focusing on unified communications, which the MS press release says includes e-mail, IM (Instant Messaging), telephony, and multimedia conferencing. Take the latter to mean WoIP - Video as well as Voice over IP.

Nortel has proven itself to be an innovator in telephony hardware and software in the past. This is also a great step forward for Microsoft in the VoIP market. Can they do the unthinkable and make this unified communications thing open? You never know.

They're at least trying to unify MSN Messenger IM with Yahoo! Messenger - both of which now have VoIP capabilities - in a new alliance. Between the two IMs, that's about 350 million users. Now what's the chances that they'll switch to SIP, thus making themselves compatible with true VoIP IMs such as Gizmo Project and Sightspeed? [Note: despite an earlier post about finding VoIP plans, Sightspeed is in fact SIP-based.]

Additional sources: Microsoft [via FierceVoIP].

Improving VoIP Audio Quality

The consumer market for VoIP grew by over 250 percent in 2005. This refers to people who actually subscribed to a VoIP service, which amounts to over 3 million people. That number is expected to nearly triple in 2006, and be nearly ten times in 2009. [C|Net News] Call audio quality is going to be an issue sooner or later, if it has not become one already.

Besides hardware-based VoIP, many more people are using soft clients such as Skype, including some PDA users, without any subscription plan. Some may even be using the voice capabilities of IM (Instant Messaging) clients such as Google Talk or MSN Messenger. For Skype alone, there are an estimated 100 million users worldwide.

Up till now, people may be putting up with poor call quality, simple because for soft client users, VoIP calls are very cheap or even free. My own experience suggests poor audio quality is fairly common. If you're a VoIP soft client user, here are a few things to consider, to improve your audio quality:

(1) Don't use a $1 microphone if you intend to record VoIP calls, particularly for podcasts. You mic doesn't have to be expensive either. You're not recording vocal tracks for a music CD. For standard calls that will not be rebroadcast, you can probably use a sub-$30 mic or headphone + mic headset combo. (My $1 mic works just fine, if my laptop RAM is free.)

(2) Make sure the RAM on your computer isn't maxed out. For my daily work, my RAM is constantly topped out and it affects my audio (and especially my video when I use a WoIP soft client such as Sightspeed). If you notice poor audio quality, you could try closing some other programs on your computer. Sometimes it's the program itself. I noticed that the free Babble.net client is unfortunately a memory hog.

(3) Expect poor audio quality if you have a slow Internet connection. If you're using a Wi-Fi setup, it might be a matter of positioning. Try moving around.

If any of the above problems arise, you'll probably get audio artefacts including warbling, echo, or buzz. Electrical interference can cause your microphone to generate hum as well.

As for the audio quality of calls over VoIP hardware or networks, that's something manufacturers and providers have to work on. As the C|Net article says, PSTN phones use dedicated networks, thus providing high quality calls. Early VoIP adopters are putting up with issues of poor audio quality and reliability. However, as VoIP usage spreads, newer customers are less likely to put up with poor service. Someone also has to come up with a reliable e911 emergency calling solution.

Who's Making Money In VoIP?

Om Malik paints an interesting VoIP application scenario with an in-beta service called Jangl. Jangl deals with issues of VoIP presence as well as security using a unique bidirectional phone number that allows two parties to communicate without revealing their own phone numbers.

The example Om gives is one applying to the dating scene. Considering that there are online dating services like Verbdate, which integrate Skype's VoIP software to allow people to talk to each other for free, there's obviously a market for VoIP dating applications. But with the way the world is, partial anonymity, or at least security from having to reveal personal info, is a boon. Jangl does this by assigning a phone number between two parties, which acts as a bridge, regardless of the numbers they are using to make the call.

Jangl joins the growing list of new companies that are opting for private financing instead of going IPO. However, history shows that when a VC firm pumps in money, they want to not only make their money back but get a return on it. That's kind of why VCs exist, right?

Jangl is only in beta, so it's yet to be seen whether they can make money from their model. Still, with all the free VoIP options available these days, the ones taking advantage of the Long Tail phenomena will be most likely to succeed. That is, give away parts of your service/ product offerings for free, and hope that a large number of people will use your paid services once in a while. Or if you're lucky, frequently. (Which is why I think that Skype's plan to enter the enterprise VoIP market is a bad idea, besides the fact that their Skype is not enterprise grade.)

But since so much VoIP service is free, it's the companies that give good value-added services at a reasonable price who are candidates for business success.

Here's my jaded prediction: new tech (and web 2.0) companies will continue to stay private for the next year or two, followed by a large number of IPO offerings in the very late 00s, capped by a market crash in 2010. History (i.e., market data and news archives) shows a recession at the beginning of every decade since at least 1970, which inevitably means tech stocks crash and burn. Only those VoIP companies who develop a stable footing in the next year or two are likely to survive, and that means having private funding, instead of suffering the vagaries of the stock market and suddenly finding that cash is low.

Finding The Best VoIP Plans

With all the choices for VoIP service now available, it's probably confusing for newbies to figure out what service is best for them. The questions you have to ask yourself, in finding a good rate and service, includes how you'll use VoIP.

Do you plan to call from your computer to another person's computer (PC-to-PC), to their phone (PC-to-PSTN), from your phone to their computer (PSTN-to-PC), or from phone to phone (PSTN-to-PSTN)? (Note: PSTN = Public Switched Telephone Network.)

The cheapest choice, obviously, is free, which PC-to-PC calls tend to be. Your choices at present include Skype, Sightspeed, and Gizmo Project. The latter uses the SIP standard, which means that users of other VoIP SIP-based soft clients can talk to each other across their networks. Clients such as Skype and Sightspeed cannot do that because they use proprietary systems. There are many more choices than those three, but they are the common ones.

For PC-to-PSTN calls, there are services like Babble.net, who have 3-month promo of up to 30 minutes free for calls to certain countries. Skype has this for Canadians and Americans until the end of 2006. For PSTN-to-PC calls, the target person needs to have something like SkypeIn service, which essentially provides a worldwide phone number.

Then there's RebTel, who've made international PSTN-to-PSTN calls very affordable, especially for mobile-to-mobile phone calls. Jajah also offers a combination of free call options, including PSTN-to-PSTN, and is based on the SIP open standard. Possibly to compete with providers such as Babble and RebTel, Skype is currently offering free weekends in July for calls to certain countries.

You can also use the free VoIP capabilities of some IM (Instant Messaging) soft clients such as Google Talk, MSN Messenger, etc., but they are strictly PC-to-PC and are client-specific.

These are just some options available right now. Keep in mind that most of them currently have poor to no support for emergency calling.

additional sources: Times Onilne UK.

CORRECTION: Despite my incorrect comment above, I have previously correctly stated that Sightspeed is SIP-based. Thanks to Peter Csathy and Andy Abramson for pointing out the error.

Could RFID Transponders Be Used For VoIP e911 Caller Verification?

If you've been following our sister publication, RFID Gazette, you'll know about a new type of RFID (Radio Frequency IDentification) tag called RuBee with IP addressing capabilities. This means that if an SED (Service-Enabled Device) is equipped with a RuBee tag, it could be accessed via the Internet. This could be a potential solution for the VoIP e911 problem.

SEDs could be any networked device, including a digital camera, digital toaster, digital lighting system, etc., which can communicate with each other, based on predefined roles. What this also means is that if a transponder with a RuBee tag were devised, it could potentially be used as a means of directing and responding to VoIP-based e911 emergency calls.

Now this is pure speculation, and there are still a lot of technical issues that have to be solved. For example, RuBee-based transponders would have to be stationary and thus not attached to VoIP phones or to any mobile device such as a laptop or PDA (Personal Digital Assistant). However, any device used to make a VoIP call would have a RuBee tag to transmit emergency status to the nearest transponder. The transponder would be designed to route the call appropriately.

This means that each transponder would have to be geocoded, possibly in sync to a postal/ zip code grid. Alternately, in cities with Municipal Wi-Fi, e911 transponders could be integrated wherever signal boosters are installed. That at least helps narrow down where a call is coming from, even from a VoWiFi phone. Finally, household or neighborhood transponders could also be made available, for those interested. Of course, in the latter case, you wouldn't want the transponder accessible to just anyone over the Internet. Just speculating.

Large Enterprise VoIP Projects Catching On

More companies are realizing the value of VoIP in the enterprise, striking up ever larger deals. Rolls-Royce, the distinguished carmaker, is incorporating VoIP into their operations. Canada's Nortel, a long-time veteran in telephony equipment, was given the $20 mln deal, which spans seven years.

Rolls-Royce's network of users spans countries in Europe and North America, adding up to over 26,000 users. This is one the largest enterprise VoIP projects to date. However, consumer goods manufacturer Kimberly-Clark will be implementing VoIP for over 200 sites consisting of over 57,000 employees.

Also, if you use the term enterprise loosely, Taipei City government in Taiwan recently started rolling out their VoIP network over Municipal Wi-Fi project, aiming at 200,000 wireless VoIP phones by the end of 2006. The initial project covers only administrative offices and public schools - a sort of enterprise, albeit government.

With so many users on one subnet, there are issues of audio file storage for voice mail, etc, just as there would be for PSTN systems. However, VoIP being relatively new, and being accessible via soft clients such as Skype and Sightspeed, there may be a tendency for employees to replace modes like IM (Instant Messaging) with VoIP, thus potentially making audio file storage a more critical issue sooner.

It's interesting that the Nortel project is expected to last seven years, although no reasons were given for that length of time. I'd like to how long other large corporations give themselves for similiar projects. Small enterprises, however, could probably roll out projects in short time period, provided they plan appropriately.

[via VoIPendium, Silicon.com, NewsFactor]

PSTN Phones Pull A Monty Python: We're Not Dead Yet

You may find it hard to believe, but I actually know quite a few people, mostly over 50 or 60, who neither have a computer nor want one. In fact, some of these people have never had an answering machine and have no intention of ever getting one. Think they have a cell phone? Some don't. What are the chances, then, that these people are going to rush out and buy VoIP handsets? Nil, of course. For them, VoIP has to be transparent.

Converged Wi-Fi/ cellular handsets might be the next big thing, VoIP hardware-wise, but good old PSTN (Public Switched Telephone Network)  phones might not be dead just yet. Making VoIP simple for consumers will go a long way towards wider acceptance of VoIP, especially from customers reluctant to make unnecessary hardware purchases.

While VoWiFi (VoIP over Wi-Fi) may be relatively simple to set up for those comfortable with technology, and VoIP over Municipal Wi-Fi very easy to use, esecially if you have something like DLink's new VoWiFi phone, there'll be some resistance.

What could be easier, then, than VoIP that incorporates PSTN-to-PSTN connections via  VoIP gateway in the network. From the customer point of view, it's transparent and requires no new hardware or handsets. Here's a diagram showing how such connections work, and also support PSTN-to-VoIP, VoIP-to-PSTN, and VoIP-to-VoIP.

VoIP Over Municipal Wi-Fi

Telecom companies these days have VoIP to contend with from not one but two fronts. VoIP is already disrupting both landline and cell phone revenues, causing telcos to reduce their prices.

Now, with numerous cities and even countries pushing for either paid or free Municipal Wi-Fi, telcos also have to contend with the potential loss of revenues from their Internet Service Provider divisions. This isn't just an American problem, it's widespread, worldwide issue, even reaching the Pacific islands, which include Fiji, Micronesia, etc.

What's more, telcos now have to deal with the loss of revenues that will result from the use of VoIP over Muni WiFi. In fact, several cities are pushing for wireless VoIP services, including Taipei, Taiwan. American telcos could learn a thing or two from the Taipei WiFly/ EasyCall project. It's the collaboration of the city government and the Taipei Computer Association (TCA), and is overseen by several ITSPs (Internet Telephony Service Providers).

Traditional telcos should be thinking about modifying their offerings to become ITSPs and even collaborating with or buying out existing VoIP providers. Especially if other cities start thinking like the Taipei government, who are using wireless VoIP to replace their PSTN (Public Switched Telephone Network) for administrative offices and public schools. Their aim is to have 200,000 wireless VoIP phones by year's end.

Obviously, if Taipei pulls this off, and figure out how to handle e911 calls, especially when there are school children at risk, other cities are going to follow suit. And if telcos don't find a way to participate, there's going to be some explaining to do to shareholders. In fact, I'm a strong believer that telcos could play an important role in solving e911 technical issues. If you can't beat ITSPs, join'em.

VoIP Deployment Takes Off In Asia and Africa

While VoIP is a technology that is beneficial to people worldwide, it's deployment appears to be gaining the greatest growth in Asian, Middle Eastern, and North African countries.

These are countries typically associated with average incomes that are much lower than those of European or North American countries. With the reduced cost of telephony provided by VoIP systems, it's not surprising that several countries in these regions have made deployment of IP telephony a priority.

In fact, Avaya's survey of decision makers in the Middle East and North African countries shows that over 70% have plans to deploy IP telephony. Some already have converged voice and data networks.

The financial benefits are realized not only in initial implementation of IP telephony, but also in the ease and cost of being able to later add VoIP applications. The drawback for telephony solutions providers based in English-speaking countries is that there is a need for other-language solutions, which could conceivably increase project costs.

While VoIP deployment in Africa and the Middle East is strong, it is growing rapidly in Asia-Pacific countries in particular, surpassing most other regions in the world, with a 60+% increase in VoIP equipment purchases.

A primary cost benefit of deploying IP Telephony comes from regions where telephone lines or cellular networks are currently at a minimum. Instead of the cost of having to install separate voice and data networks, the convergence afforded by IP telephony results in an overall savings.

Sources: VoIP News; Avaya - South Africa, Middle East + North Africa; Mena Report [via VoIPendium]

Ubiquitous VoIP Communication and SED Service-Enabled Devices

Imagine for a second an "Internet of devices". These are devices that can communicate with each other over the Internet Protocol. IPv6, that is. IPv6, the next version of the Internet Protocol, has been touted by some as the next version of the Internet (or maybe Web 3.0, depending on how you look at it.) IPv6 will allow SEDs, Service-Enabled Devices, to be interconnected, thus allowing consumer electronics gadgets and appliances to talk to each other.

SEDs know their role and how they need to interact with specific other devices. Besides computers, the list of future SEDs is reputed to include digital cameras, stereo systems, toasters, stoves, refrigerators, lawn sprinklers, your lighting system... Well, you get the idea.

An example of an SED-to-SED interaction might be as follows. At the press of a button, a digital camera sends a packet of photos, via WiFi, to a laptop running software that then publishes the pics to a pre-configured website gallery. The laptop, of course, can be connected to the Internet either directly through a local network, or even via a Cellular WiFi data card.

Another example would be to log on to your home's hypothetical web server and turn on your lawn sprinkler or even the air conditioning. (In this situation, you would be overriding scheduled activities.) Sure, this sounds like something that media mentioned, oh, in the 1980s, 1990s, and so on. But it's now possible.

Where's the VoIP in all that, you're asking. Well, what if you could do the same as described above, but by just speaking a few commands into a VoIP-enabled SED, which would then communicate to your sprinkler or whatever?

There's a very real possibility that VoIP capability will become ubiquitous. Thus, all kinds of consumer electronics will be SEDs and VoIP-enabled. You'll be able to talk to your gadgets, order them around a little with a few voice commands. When that happens, you'll also be able to give authorization for various transactions  [VoIP Lowdown] via any VoIP-enabled device in an office, home, store, or wherever. Alternately, gadgets will be able to VoIP you.

Of course, we'll have to work on voice-recognition software as well as language translation, but there's a lot of work already going on in those areas as well audio-search, audio-to-text, and text-to-audio.

We're not quite yet at a state of communications as on TV's Star Trek Enterprise, but we're not far off. When that happens, however, you just might want to be careful what you say out loud.

NetAlly Works with Avaya, Cisco and Mitel

NetAlly Lifecycle Manager of Viola Network has now added up with Avaya's Communication manager, Mitel's Enterprise Manager and Cisco's CallManager and voice gateways. NetAlly helps for the successful implementation of VoIP with a minimum investment of time and resources. It is a software suite for managing VoIP assessment, deployment, network monitoring and network optimization. The VoIP assessment is significant to determine whether the network is reliable for VoIP deployment or not. It operates in passive monitoring mode to collect data on VoIP phones, monitoring call records and quality statistics.

Via: [IP Telephony]

BellSouth deploys GeoProbe VoIP monitoring system

BellSouth Corporation will install Tektronix's GeoProbe VoIP monitoring system from its Unified Assurance suite throughout its IP network. This system will easily trace out  the networking problems of BellSouth affecting the VoIP service by assembling all signaling data relating to VoIP traffic.

Doug Dickerson, Vice President at Tektronix quoted,

While VoIP offers a solid case for reducing service provider operating costs and increasing service revenues, these advantages could be offset by the higher degree of complexity in monitoring service quality if not managed properly.

News: [VoIP Central]

Tollgrade, Minacom agree for VoIP testing initiative

Tollgrade Communications, Inc. and Montreal-based VoIP test vendor Minacom joined forces to deliver integrated testing products for VoIP over cable networks.

As per the agreement, the companies will install products on Tollgrade's newly introduced End-of-Line (EOL) probes and its DOCSIS-based Status Monitoring Cheetah Transponders.

Test calls apply Minacom's VoIP testing technology to measure over 40 service quality measurements, which include speech power, speech quality (MOS), distortion, loss, clipping and noise.

In addition to these, the VoIP technologies of Minacom measures call connectivity metrics like Post Dial Delay and dual-tone multi-frequency transparency.

Via: [CNS Magazine]

Cosmoline to Expand VoIP Network

Cosmoline is using the services of Veraz Networks for expanding its services with Veraz’s ControlSwitch softswitch. This step would enable Cosmoline to gradually replace eight of its legacy switch with a single Veraz Control Switch. Veraz is planning to deploy 20,000 VoIP  ports and  expanding Cosmoline’s VoIP network which carries services over fifty million usage minutes monthly.

It had earlier deployed the Veraz Long Distance Network Compression system using Veraz’s I-Gate 4000 media gateways which can be used for VoIP trunking and switched VoIP services.

Via tmcnet

Brazil Gets GlobeTel’s VoIP Platform

GlobeTel Communications has come up with a VoIP based communications network for Brazilian iLigue.com.br. This portal provides Brazilian telecom customers calling services locally and around the world. VozBrasil.com was launched two weeks ago and iLigue is being rolled out along with GlobeTel’s Brazilian partners. This network enables Brazilian residents to obtain a telephone number in Brazil, states or any other country without the requirement for either broadband internet access or special equipment.

Dan Erdberg, Division president, GlobeTel’s VoIP Division stated:

We've developed a system specifically for Brazil that not only allows our customers to have local phone numbers and service in Brazil, with the ability to call within their state or anywhere in the country at a very low cost, but it also gives them the ability to have a US-based phone number that can be forwarded to their Brazil-based cell phone, home phone or office phone.

Via ipcommunications

DigiLinea expands regional network

Good news for the Americans whose friends, relatives and well wishers are living in Latin America. Because DigiLinea has some special arrangements for them. The leading VoIP service provider has planned to expand its regional networks. Now the US Hispanic community can communicate with their friends and relatives at Latin America at a lower cost. It is the better opportunity for the people to keep touch to their relatives living in a distant place.

DigiLinea has acquired the legal right to provide telephonic services to its customers. It has acquired telephonic licenses in Latin America. At the same time, the company has distributed the licenses to ensure services and local telephone operators in Gutemala. A good number of Guatemalan immigrants are living in the US. With DigiLinea planning to provide cheap VoIP services, the customers are likely to develop a sense of attachment towards it.

Via: [TMC Net]

Bob Jane T-Mart Migrates to Nortel’s IP Telephony

Bob Jane T-Martwhich which is regarded as one of the Australia’s largest independent tyre and battery retailers is migrating its national communication network to IP telephony using Nortel solutions. The overhaul would be seeing more than thirty company owned stores and sites upgraded with Nortel IP telephony and digital handsets with similar packages offered to the company’s 120 plus franchise stores around the country.

According to Edward Hore, IT manager, Bob Jane T-Mart:

Towards the end of last year we realized our existing phone system was becoming far too expensive and difficult to manage, so we started looking for a self-maintaining system with the aim of significantly cutting our call costs in the process.

Via ComputerWorld

VOIP configured for K-12 schools

Rauland-Borg has introduced the Telecenter VI communications systems for K-12 schools. Rauland-Borg has designed the Telecenter VI in a manner that combines phone, emergency communications and paging/intercom on a single Voice over IP platform. Besides the regular VoIP features , the Telecenter VI  has the much-needed school features – intercom, paging, emergency tones, and class change bells. 

The Telecenter VI is big on critical life safety features. It enables 911 call alerts to key administrative staff, emergency all-page from any phone, remote door lock/access control, facility-wide lock down/take cover tones, and telephone trunk seizure. This ensures that that 911 calls always go through. Telecenter VI also offers capabilities for enhanced parent-teacher communications, which includes voicemail, homework helpline, and an audio bulletin board.

Via PRWeb

The largest U.S. university VoIP deployment

Educational institutions have enthusiastically adopted VoIP. Just the other day, we reported about a remote Indian settlement that used VoIP to get in touch.

Boise State University has installed perhaps the largest Voice over Internet Protocol communications in any college in the United States.

The University’s new VoIP-based system handles more than 14,000 phone numbers and 4,000 handsets. Time Warner Telecom provided the IP trunk connections. The cable company is providing 20 megabit per second speed link

Via UPI

School District using VOIP

Earlier, we reported about a city administration using VoIP services. The Deer Valley Unified School District (DVUSD) in Phoenix, Arizona is spread across 400 square miles and comprises 37 schools. The school district has recently deployed VoIP-based communication system.

The VoIP system uses Nortel's CS 1000, and Tenor VoIP switches from Quintum Technologies. Not surprisingly, the school district has reduced its communication costs vis-à-vis regular landline system.

Moreover, DVUSD’s also wanted to integrate a wide range of analog systems into their new VoIP-based network. It was able to do that using Nortel's solution.

Via Yahoo Finance

Making Unified Communication a reality

According to the latest research report from Light Reading , organizations must solve all sorts of interdepartmental and technology issues before they set out to harness all benefits of IP-based communications.

The report rightly says,

Rather than simply using the same network resources, telecom and IT must be willing to share access to applications and databases.

It goes on to suggest,

Solutions must incorporate existing components from a variety of cross- industry players, which means standards such as SIP [Session Initiation Protocol] and Simple [SIP for IM and Presence Leveraging Extensions] become extremely critical.

Via Sys.Con

Soundwin's Eight-port VoIP gateway

Soundwin Network Inc., a Taiwan-based company, introduces an eight-port high-density gateway, the S804.

he S804 VoIP gateway supports four FXS and four FXO ports that connect to IP and PSTN networks. It also has the NAT router function so that you can access the Internet using a single IP address.

The S804 VoIP gateway complies with H.323 and SIP protocols. It is built upon a Texas Instrument (TI) chipset. It comes equipped with T.30/T.38 support and G.711/G.723/G.729 voice codec.

The S804 VoIP gateway also features echo cancellation, polarity and NAT traversal, smart QoS and voice channel display. Its other main features are direct dialing mode, silence compression, watches dog function and outbound proxy function.

Via Global Sources

VOIP in Russia, courtesy Nortel

Telecom giant Nortel has provided the technology for an IMS-based, next-generation network developed in the city of Novokuznetsk.by JSC Sibirtelecom, a telecommunications service provider in Siberia.

The new network in Novokuznetsk meets the requirements of modern NGN communications standards. It is capable of offering advanced multimedia services such as Internet telephone calls, Internet mobility and video telephony.

It is also true that the Russian Wireless market is a tough nut to crack. One hopes that this innovation in Novokuznetsk is a success.

Via TMCnet

University of Queensland uses Open Source VOIP

University of Queensland will use an Asterisk-based VoIP system over its multi-campus fibre network. Next on the University's agenda is use VOIP on its brand new Wireless Network.

As of now, the University has successfully integrated the Asterisk server with its traditional DM PCS. It is now looking into the presence capabilities of the system.

If things go well, the university plans to use this new VOIP system for its 5500 staff and 35,000 students. It is going about this a phase-wise fashion.  At present, only about 10 people are using the Asterisk system. Next, the University will begin a pilot project in one of the residential colleges, bringing VoIP to students' dorms – involving around 200 users.

Via VoIP News

NT government moves ahead with VoIP plans

The Northern Territory government is targeting user support and governance which is critical to the success of a looming state wide VoIP. Within this month, the last step of a pilot project which includes deployment at the Department of Corporate and Information Services in Palm Court in Darwin would be completed. The pilot project which was introduced in October has reached a limited number of sites which includes the Department of Justice and Department of Health and Community Services.

The project would see most government sites which include around 12,000 public servants using a VoIP solution from Telstra.

via [ZDnet]

Sentito Networks' ONX solution wins awards and lots of praise

The Sentito ONX solution - featuring an Intelligent Voice Gateway (IVG), Proxy7 Signaling Gateway, and PreVision Network Manager – has won a "Best of Show" Award at Technology Marketing Corporation (TMC(R))'s INTERNET TELEPHONY Conference and EXPO East 2006.

Sentito's solution was chosen for 'Innovation and Quality'.

The Best of Show awards are given to companies unveiling the most impressive new products or releases at the show.

You can find the complete list of the winners on the TMC Web site, www.tmcnet.com. The list will be published in the March 2006 issue of INTERNET TELEPHONY(R) magazine.

Via [BusinessWire]

The coming age of Voice over wireless local area networks (VoWLAN)

Voice over wireless local area networks (VoWLAN) combines voice over IP (VoIP) and wireless networking.

Analysts say that the lure of VoWLAN is due to the fear of high Wirelesss phone costs.
There's a lot of pent-up demand for VoWLAN, and particularly for dual-mode cellular and wireless VoIP phones.

Moreover, VoWLAN also frees up the office workers who are no more bound by location inside the office. VoWLAN allows for freer movements.

Anytime, anywhere conference, anyone?

Read More

Technical Services about to complete VoIP transition on campus

Students would now be able to enjoy features such as caller identification on their dorm room phones when Technical Services completes the last phase of campus wide switch over to VoIP. The dorms in the Choater Cluster have already been converted to VoIP before Fall term and the remaining were converted over the winter break. The service would soon be available in all the dorms. The last phase of the project involves converting the remaining off campus buildings, safety and security communication system and blue light phones which would be done by June 30th.

via [The Dartmouth]

UIP 165P: The home wireless station

Uniden America in partnership with 8x8 announced a mobile station, Uniden UIP 165P, which can link up to 10 or regular phones to a mobile station. This can easily mesh the whole house or a small office for the Net Telephony.  The company is confident; this could help VoIP Technology to move closer to the regular PSTN customers. The wireless station is available for $130 and right now has capability to handle 10 phones with distinctive rings.

The system is compatible with 8x8’s with 8x8's Packet8 Internet Phone Service, which offers unlimited calling in North America for $20 per month.

via [UPI]

TotalRoam® Mobile VPN software v4.1

Padcom Inc, has released TotalRoam® Mobile VPN software v4.1. It will offer enhanced support for VoIP applications and will enable streamlined TotalRoam installation and configuration. The software will also allow over-the-air administration for remotely upgrading TotalRoam clients and QoS values will be assigned to data transport packets.

The mobile VPN implemented by TotalRoam provides a secure and continuous connection. It allows users to roam between different network infrastructures while using only one connection. The software, thus, provides maximum network connectivity and allows mobile workers to access critical business applications.

SkyTalk VoIP network in Miami

SkyTalk Communications has finished the deployment of a full VoIP infrastructure situated at a secure location in Miami. Its sister concern Payless Telecom Solutions, Inc. will provide IPTV and VoIP services to its customers via the Skytalk backbone.

The future of P2P

P2P as an alternative for enterprise telephony gained support with the acquisition of NimCat Networks by Avaya. The acquisition was worth $ 46 million. NimCat is based in Ottawa and its NimX software can run on any standard IP phone.

The software allows users to avail PBX functionalities such as voice mail, auto attendant, call waiting, etc without having to deploy a dedicated telephony server. The plug and play system features self-configuring phones that can automatically assign extensions. voipplanet.com reports:

"Avaya looked at [NimCat] as a viable competitor at the lowest end," Dzubeck says. "And it could also link easily with the high-end systems for the big play into the branch office [market]." The price for NimCat was also right, he adds.

Read More: NimCat and Avaya: Declawing the cat?

VoIP offerings from Sonus

Sonus Networks Inc. provides VoIP infrastructure systems to traditional carriers, wireless network operators, cable providers, and ISPs. Sonus has supplied its equipment to customers that include AT&T, Bell South, Cingular Wireless, Qwest, Level 3, etc. Since its birth in 1997, Sonus has concentrated on developing a scalable and distributed network architecture.

This focus has resulted in Sonus gaining eminence as a carrier-class equipment provider with the advent of IMS. Sonus products support wireless technologies such as CDMA, 2GSM, 3GSM, and WiFi and wireline technologies such as xDSL and cable. The Sonus architecture has distributed intelligence, centralized subscriber database and routing, call signaling, and common applications.

The four major functions as defined by the IMS are Border and Media Control, Session Control, Application Layer, and Management and Charging. voipplanet.com reports:

The Application Layer products include the ASX Feature Server, which provides traditional, Class 5 subscriber features, such as call waiting, call forwarding, caller ID, and so on; and the OSPA Application Servers, which are provided by third parties.

Read More: Vendor Network Architectures—Part V: Sonus Networks, Inc.

Problems with peering

Although peering deals between service providers allow them to circumvent PSTN, they can also lead to problems. When Level 3 Communications ended its peering deal with Cogent Communications in October 2005, it led to a major outage for a section of customers of both providers.

The reason for the peering arrangement falling through was that Level 3 believed that Cogent was using the peering network to direct an excessive amount of traffic and was not reciprocating by receiving an equal amount. The resulting outage has focused attention for the first time on the need for a body to oversee peering arrangements. redherring.com reports:

Usually peering arrangements allow the peers to avoid costly connections with the public network. Traffic on the peered network usually flows without cost to the peers.

Read More: Cable Talks VoIP Peering

Backup for an IP network breakdown

Network failures in an IP environment can lead to a communication breakdown between the headoffice and the branch offices. Even though installing an IP PBX in each branch office can help companies to forestall such a problem, the cost of this exercise would defeat the very purpose of implementing IP telephony.

Multi-Tech(R) Systems has launched a SIP Voice over IP (VoIP) server that provides elementary IP telephony service to remote locations in case of a WAN breakdown. The MultiVOIP SS survivability server can also link to analog phones and fax machines. news.thomasnet.com reports:

Shipping immediately, the MVP210-SS two-port model sells for an MSRP of US$999, the four-port MVP410-SS for US$1599 and the eight-port MVP810-SS for US$2599.

Read More: SIP VoIP Server helps IP telephony survive network failure

Commtel Network to resell Nokia DSL

Commtel Network, which is a value added reseller of network solutions by Nokia, will be supplying Nokia broadband DSL solutions to Telecom Cook Islands Ltd in the Pacific. The DSL system should start working by March 2006. It will represet a first for Nokia in this market.

The Nokia D500 IP DSLAM (Digital Subscriber Line Access Multiplexer) is the only IP DSLAM solution in the industry that provides integrated IP functionality and has been selected for making quick Internet access available in this market. prnewswire.com reports:

The Nokia D500 is a complete IP-based multiservice access node platform for providing both quality of service and secure authentication.

Read More: Nokia and CommTel Expand Broadband in the Pacific

Broadband users increase in Britain

In May 2005, the number of broadband users in Britain went past the number of dial-up connections for the first time. Of the more than 8.8 million broadband connections in Britain, 2.8 million use cable. The number of broadband users in Britain is expected to cross 10 million by the first half of 2006. tmcnet.com reports:

Expanding broadband internet provision is a central part of the rationale behind the deal (as well as squeezing out costs in a fiercely competitive market).

Read More: It's survival of the fastest as Broadband Britain

Wi-Fi networks in US cities

Municipal Wi-Fi networks are under consideration in various cities in the US. According to the report titled "2006 Muni Broadband Outlook" by Light Reading Insider, cities such as Anaheim, Calif.; Atlanta; Austin, Texas; Boston; Buffalo, N.Y.; Chicago; Denver; Kansas City, Mo.; and Lenexa, Kan. are planning to deploy Wi-Fi networks.

The main reason in favor of Wi-Fi deployments is that it allows greater capacity per user since the access points are deployed over small areas and the spectrum can be utilized several times. smallbizpipeline.com reports:

Because of the simpler CSMA protocols and smaller networks, Wi-Fi also has inherently lower latency, less than 10 msec, whereas 3G networks are struggling to get down to 100 msec.

Read More: The Case For (And Against) Muni Wi-Fi

Vonage America supports i2

The Interim VoIP Architecture for Enhanced 9-1-1 Services or i2 is receiving support from Vonage America. The i2 standard is for an architecture that will enable IP telephony service providers to make E911 services available by using the existing infrastructure. This arrangement will continue till an all-IP E911 system is available throughout the country. voip-magazine.com reports:

Last month, NENA announced the release of a reference guide to help VoIP service providers in planning and communication with regards to the FCC's E911 mandate.

Read More: Vonage Supports NENA's E911 Plan

Growth of broadband DSL

According to the analyst Point Topic, broadband access via DSL acquired 40 million customers globally over a period of one year till September 30, 2005. This adds up to a total of 125 million broadband DSL customers across the globe. voip-magazine.com reports:

This report confirms an Insight Research study, which states DSL growth as well as rapid growth of VoIP and wireless revenue are pushing the telecommunication industry's revenue to $1.2 trillion in 2006.

Read More: Report: 40 Million New DSL Subscribers in Year

Considerations while deploying an IP network

Three important considerations for a successful IP telephony deployment include network architecture, business processes, and an understanding of next generation applications. The issue that is top most in the minds of business managers is regarding the ability of the IP communications network to co-exist with the existing infrastructure.

A successfully deployed converged network can radically alter the manner in which staff communicates and impact employee productivity and the scope of business. Businesses need to plan beforehand for changes in business processes and foster acceptance and adoption of IP-based communications. voip-magazine.com reports:

A careful and comprehensive plan that aligns business process with technology architecture should be an executive’s first step before the actual implementation begins.

Read More: Business Processes for a Converged Enterprise: An Introduction

VoIP peering for CableLabs

The RFI released by CableLabs has met with a good response. Companies such as VeriSign Inc, Stealth Communications, NeuStar Inc. etc have answered to the RFI.

CableLabs intends to develop a VoIP peering fabric that would allow VoIP subscribers of the member MSO’s to communicate directly without their calls being routed through a PSTN network. This could save the MSO’s thousands of dollars every month in interconnection fees to the PSTN companies.

BreezeACCESS VL by Alvarion Ltd.

Alvarion Ltd. has provided Rioplex Wireless with BreezeACCESS VL for increasing its network. Rioplex is based in Texas and has completed the deployment of a 5,000-square-mile broadband network. The company services residential customers and SMBs and uses licensed frequencies. The company can now offer VoIP and data services to its customers including telecommuters with the help of the BreezeACCESS VL system. Speeds can go up to 25 mbps for every user.

Improved performace of IP PBXs

When legacy PBX systems were the norm, users had to pay for proprietary operating systems that used proprietary hardware and software. The upshot of legacy PBXs has always been the reliability and availability. The advent of VoIP was greeted with skepticism regarding the ability of IP telephony to provide reliability on par with legacy systems. networkworld.com reports:

Network managers immediately need to understand the critical elements that comprise modular open PBX systems and begin understanding the finer points of each.

Read More: VoIP: more survivable than legacy PBX

BIGData from US LEC Corp.

US LEC Corp. had added BIGData to its Dynamic T VoIP service. BIGData is a high-bandwidth Internet and networking service option. The company serves enterprises in Eastern United States. Dynamic T BIGData allows speeds of up to 45 Mbps and is ideal for businesses that require large bandwidths. The bandwidths are available incrementally; this allows businesses to scale as per their requirements. prnewswire.com reports:

"While Ethernet Local Loop provides a desirable solution for some high-bandwidth requirements, US LEC can now also offer businesses a more scalable Internet solution that adds an advanced, feature-rich voice solution and optimizes the bandwidth investment."

Read More: US LEC Boosts Dynamic T(SM) VoIP Service

Adtran Total Access

The Adtran Total Access 904 and 908 are 4 and 8 port gateways that offer the functionalities of an SIP gateway, an IP router, a firewall, remote survivability, etc. Carriers can use the gateways to avail VoIP, SIP trunking, hosted legacy, and IP PBX applications. WAN connectivity can be achieved by means of a T1 port, connectivity to an Ethernet switch can be had by means of a 10/100 Base T Ethernet port, a PRI/Ti port and voice ports provide PBX connectivity and support for analog devices, respectively.

Toll-free technical support is provided on a 24x7 basis, the gateways are accompanied with a 10-year warranty, and users are provided with free firmware updates. The 4- and 8-port gateways cost $ 1,025 and $ 1,150, respectively.

Open source IP PBXs

Open source IP PBXs are developing at a fast clip and could soon be providing the scalability and availability that is expected of enterprise telephony solutions. By May 2006, sipX could well provide 1,000 seats a server. However, it may still be some time before open source IP PBXs pose a serious challenge to proprietary IP PBXs.

This is because open source IP PBXs make message transfer difficult due to server-centric voicemail implementations and configurations cannot be managed on a system-wide basis. itarchitect.com reports:

As with Linux and other open-source projects, open-source VoIP is offered both as an ongoing project developed by the grass roots community and as a prepackaged product that's tested, maintained, and sold by a sponsoring vendor.

Read More: Find the REAL Value

TJ 995 from Terayon

The TJ 955 embedded multimedia terminal adapter from Terayon Communication Systems, Inc. has been awarded the CableLabs® certification for the PacketCable™ 1.5 cable voice over IP (VoIP) specification. The TJ 955 can support high-speed Internet access and it delivers two primary lines of VoIP telephony for every household in a low-cost manner.

Broadband carriers and broadband services

Cisco has stated that broadband carriers need to focus on bettering the quality of their applications so that customers can enjoy better access services. Cisco has stated that it will help service providers to add IPTV to their offerings. networkworld.com reports:

The idea is not to interfere with services consumers might receive over the Web, but to ensure any carrier services connected to the access service have a guaranteed level of quality, no matter what else is running over that link.

Read More: Carriers must take control of their broadband, IPTV services, Cisco says

Hybrid PBXs find favor

Network professionals are of the opinion that hybrid IP/legacy PBXs are enabling companies to experience increased productivity and cost savings without having to completely replace the existing communications network. The hybrid of IP and TDM technology has been available from PBX vendors for quite some time.

The main drivers of this technology were reduced costs of inter-company long distance calls and converging of voice/data T-1s. Legacy PBXs use cards for connecting to the LAN and gateways for translating voice signals between IP and TDM. TDM handsets that have been IP-enabled can be used from desktops and obtain features such as click-to-call and unified voice/e-mail. networkworld.com reports:

Avaya is one company following the hybrid telephony trend in the industry - moving towards IP while maintaining TDM presence. Merrill Lynch says Avaya's TDM PBX sales shrunk 3% from the second to the third quarter of 2005, and compared with the same quarter a year ago sales are down 20%. Meanwhile, its hybrid IP voice sales grew 14%.

Read More: Users: Hybrid PBXs work

Italtel and Cisco for Polish PSTN

Italtel and Cisco systems will be providing TP, the primary telecom operator in Poland, with equipment for implementing a PSTN gateway solution. The two companies will also assist in system integration. The new multi-service IP network will allow TP to offer VoIP and high-speed Internet access to customers. webwire.com reports:

The efficiency and robustness of the proposed Italtel platform have been demonstrated by its performance in other network infrastructures which have commercial traffic capacities of up to 500 billion voice minutes per year.

Read More: Italtel and Cisco Systems Connect Telekomunikacja Polska

SAN for Coffs Harbour

The Coffs Harbour City Council, NSW, Australia, has taken recourse to SAN virtualization in order to consolidate its infrastructure. computerworld.com reports:

Andrew Sales, the council's special IT projects manager, said that after seeking bids for an integrated infrastructure solution, $337,000 was spent on new HP Proliant servers and an EVA3000 SAN.

Read More: Council moves on virtualization, IP telephony

UNH-IOL tests IPv6

The University of New Hampshire InterOperability Laboratory (UNH-IOL) was assisted by 10 companies and the U.S military in a project that successfully demonstrated that IPv6 can be used to make secure, international voice calls. The tests were carried out on Moonv6 network, which the largest multi-vendor IPv6 network in the world. convergedigest.com reports:

UNH-IOL engineers employed realistic traffic streams, passing mixed VoIP and data traffic over the IPv6 network and successfully demonstrating basic application layer functionality and IPv4 equivalency in areas such as addressing.

Read More: UNH-IOL Tests Security, Mobility and VoIP on IPv6 Testbed

IMG 1010 from Excel Switching Corporation

The enhanced version of the IMG 1010 integrated signaling and media gateway from Excel Switching Corporation will carry SIP and ISDN functionality. This will allow service providers to incorporate VoIP capacity and make a smooth transition to an all-VoIP scenario. convergedigest.com reports:

The IMG 1010 supports SIP, H.323, SS7 and ISDN simultaneously, all in a compact 1U design that can accommodate up to 768 channels.

Read More: Excel Switching Enhances its Integrated Signaling/Media Gateway

Spectra2 Release 4.4

The new Spectra2 Release 4.4 software by Tektronix allows the Spectra2 VoIP testing solution to achieve twice its VoIP and converged network RTP large-scale load generation features. convergedigest.com reports:

Spectra2 combined with the WTI RTP board performs large-scale media tests, allowing users to evaluate busy hour call capacity by simulating load under ever-increasing, real-world operating conditions.

Read More: Tektronix Scales its Spectra2 VoIP Test Solution

Linksys and Telabria come together

Linksys, which is a division of Cisco Systems, Inc will be collaborating with UK-based Telabria for making VoIP services available to business and residential subscribers. Linksys analogue telephone adaptors (ATAs) and VoIP handsets will be used in this endeavor to cover the huge WiMAX-class wireless broadband network, which happens to be one of the largest in Europe.

I-Gate 4000 Edge

The I-Gate 4000 Edge is a media gateway from Veraz Networks that is meant for supporting distributed low density sites. It can scale up to 500 simultaneous calls. The I-Gate 4000 Edge supports diverse compression, switching, and applications like VoIP trunking, legacy PBX access, mobile network compression and switching, etc. The gateway enables carriers to implement compressed backhaul and local switching applications.

New products from Cisco

Cisco Systems has launched several products and product enhancements that are linked with the service exchange framework (SEF), which is the service convergence layer of the IP NGN architecture. SEF is interoperable with both IMS and non-IMS networks. It facilitates the availability of a number of SIP applications like dual-mode telephony, push to talk, etc. It enables users to offer bundled services that can be customized by means of selection, prepaid and postpaid alternatives, different billing and usage models, etc.

SEF facilitates monitoring of VoIP quality in real time in non-IMS applications. It also enables the combining of network with third-party anti virus and intrusion detection appliances. The new offering comes with a session border controller integrated on the XR 12000 series router. This allows for interoperability of the SIP and H.323 signaling and the MPLS-enabled media gateway. Product enhancements have been made in the call session control platform v 3, service control engine v 3, PGW2200 media gateway controller v 9.7, and the BTS 10200 softswitch v 4.5.

4500 series from 3Com

The 4500 series of switches from 3Com will enable SMBs to support VoIP and converging applications. The series includes a 26 port switch and a 56 port switch; the switches can work in networks installed by different vendors. Layer 2 switching and dynamic layer 3 routing functionality is provided as well. The switches are also available with PoE. The 26- and 50-port switches cost $ 695 and $ 1295, respectively. Their cost with PoE is $ 2,292 and $ 4,295, respectively.

VoIP deployment

VoIP deployment requires changes in the existing network in order to make it capable of handling real-time traffic. This also calls for a thorough pre-deployment assessment of the network. According to Gartner, around 85% of the existing networks are not geared for handling VoIP transmissions and three fourths of the companies that do not carry out a predeployment analysis of their IP network infrastructure are bound to falter with the actual deployment.

In order to prevent seen and unseen problems from hampering a VoIP deployment, it is important that companies execute necessary changes such as replacing LAN/WAN links, including PoE for the network switches, etc. Predeployment assessment tools such as AppareNet Voice can help in assessing problems with the NIC drivers such as full- and half-duplex conflicts. Rate-limiting queues and latency can lead to packet loss and degradation of voice quality. Faulty media and incorrect media deployment such as incorrect cabling and imperfect optic ends, electromagnetic interference, etc can lead to variable jitter. eweek.com reports:

"You'll see call-quality issues there first," said Kurt Wright, senior network engineer at IPC. Engineers execute tests that measure bandwidth utilization, packet loss, round-trip time, latency and packet reordering. They also look at mean Opinion scores, which measure call quality.

Read More: Dodging VOIP Predeployment Pitfalls

IMOD for the army

The Infrastructure Modernization Program (IMOD), which is to commence in 2006, will cost $ 4 billion and is one of the most significant telecom projects issued by the federal government. The program aims to modernize the fiber-optic cable and wireless communication networks at the various US army bases globally. This will help in providing better support to critical missions. The contract, which has a five-year base term and one five-year option, will be awarded in April 2006. The program will also help the army to make a shift from POTS to VoIP in the next five years.

The IMOD program will be carried out under the Installation Information Infrastructure Modernization Program (I3MP). IMOD was preceded by the billion Digital Switched Systems Modernization Program, which cost $ 1 billion. Under the program, 18 contracts were awarded to 17 vendors in June 1997. washingtonpost.com reports:

Other big Army telecommunications contracts to be awarded in 2006 are for the $20 billion Information Technology Enterprise Solutions 2 Services (ITES 2S) and the $5 billion Worldwide Satellite Systems Program (WWSS).

Read More: Army Prepares Major Telecom Transition

VoIP interconnect

In order to consider VoIP interconnect, a VoIP carrier has to take into account the expenditure and quality of the calls. Even though the number of VoIP carriers is on the rise, the carriers are trying to reduce costs by cutting down on the number of interconnects. The cost of establishing a bilateral VoIP-peering arrangement is too high even for Tier 1 carriers.

A single private interconnection for VoIP peering requires equipment such as routers, switches, session border controllers, and a significant amount of effort. The cost of all this can run into hundreds of thousands of dollars. The cost of running and maintaining such an operation would amount to tens of thousands of dollars. The Border Gateway Protocol (BGP) is used by carriers to interconnect through the public Internet. This helps to bring down transport expenses. However, since BGP offers “best effort” routing, It is not very suitable for real-time VoIP applications. BGP chooses the route with the fewest number of hops but does not take into account variables such as network conditions and cost.

The QoS of VoIP traffic routed through various exchanges cannot be controlled by the carrier and voice quality invariably suffers from a degrade with each handoff. The peering solution do not enable easy introduction of new services by carriers. The different aspects in a network need to function seamlessly if carriers wish to benefit from a smooth interconnection of services and business practices.

A new model of VoIP interconnect service proposes to offer seamless interconnection at low cost. Several service providers are now providing direct and transparent VoIP peering that allows carriers to access the networks of other carriers through a single interconnection. The quality of voice is also high due to optimization of the routing decisions. The routing techniques used take into account factors such as load balancing, line quality, etc. The paths are judged for latency, packet loss, and jitter. A seamless VoIP interconnect across the media and protocols of the various carriers can open up the doors to a number of profit-making services.

iNAV 9400

The iNAV 9400 is a SS7 signaling gateway that is used with IETF SIGTRAN signaling networks. It has been developed by Interphase. The SS7 signaling gateway works with low speed, high speed and ATM line interfaces. It supports a number of protocols such as ANSI, ITU-T, ETSI, TTC-Japan, NTT-Japan and China for SS7.

Communication between networks

Even though connectivity is becoming increasingly easy to access for the lay user, communication between the various networks that make it possible is not easy to achieve. The reason for this is that standards and protocols vary from network to network. The Internet Protocol for Multimedia Subsystems (IMS) is a standard that looks to bridge the gap between networks. mobilepipeline.com reports:

From simple phone calls, voice mail and call waiting, to wireless text messaging and multimedia downloads, most existing telecom services were designed to perform their specific functions as if walled off into distinct silos on the network.

Read More: IMS Convergence Technology Gains Momentum

Global Crossing joins hands with CPCNet

Global Crossing and CPCNet have linked their networks together. CPCNet Hong Kong Limited provides IP VPN service across 12 cities in China. It does so by using its MPLS network. Global Crossing will assist CPCNet in providing IP-based enterprise services along with end-to-end QoS.

IP-VPN market set for growth

According to a report by the Vertical Systems Group, the revenues from dedicated IP VPNs will total $ 34.6 billion in the US for the period 2004-2009. Of the total revenue, network-based services that will use MPLS or IP networks will account for 50% of the revenue. According to the report, there will be around 1.7 million dedicated access VPN sites by 2009 and custom Internet-based VPN networks will add $ 12.2 billion to the total whereas $ 5.3 billion will accrue from site-to-site service revenues.

The market for dedicated IP VPNs is cost sensitive. Other factors that businesses consider include interconnectivity between various networks, VoIP, level of service, the service-level agreements, etc. Users are moving away from frame relay Access technologies include DSL, Ethernet, and OC-3+. DSL is the most preferred technology.

IMS products from Lucent

Lucent counts BellSouth, SBC, and Cingular Wireless among its customers for IMS products and services. The IMS architecture uses a digital interface to provide multimedia services over wireless and wireline networks. Lucent has several IMS contracts in its bag and the potential revenue is to the tune of hundreds of dollars.

According to UBS Warburg, Lucent could be involved in IMS deployments in the range of $ 75 million to begin with but will see revenue generation only in FY 2006. Lucent has to evolve a business plan that will enable it to earn revenue from the end users. Lucent is also concentrating on 3G wireless mobility, optical/data convergence, and broadband access. The demand for these services is set to increase and according to UBS, these services could grow at a CAGR of up to 41% in the next four years.

Switches from HP benefit Canadian resort

The Sunshine Village Ski & Snowboard Resort at the Banff National Park in Canada had initiated an IP-based network upgrade in 2002. It had cost US $ 236,717 at that time and according to the resort, it has recovered the investment many times over.

The upgrade involved using ProCurve switches from HP and laying a fiber-optic network. This has resulted in savings on long-distance calls and reduced maintenance costs. Earlier, the bank had to shell out Canadian $ 1,800 every month for a single leased-line to a bank in order to facilitate credit card transactions. The cost has now come down Canadian $ 10.

The IP Multimedia Subsystem

VoIP systems can be integrated with the circuit-switched networks and with cellular networks. VoIP integration with cellular networks is achieved with the help of softswitches and the idea is referred to as the IP Multimedia Subsystem (IMS). Several softswitch vendors have begun to incorporate IMS into their products.

The objective of IMS is to provide Internet connectivity to cell phone users. This opens the doors for a host of IP-dependent services such as VoIP, video-conferencing, etc. This requires that the carriers being used by two communicators be IMS-compliant. Thus, IMS opens up new opportunities for revenue for the vendors. IMS uses SIP. IETF has done valuable work in the development of IMS. For using IMS in cellular networks, the SIP servers such as registration and authentication servers have to be deployed inside the home network.

The market for NGN networks

The market for NGN equipment continues to grow as carriers and service providers seek to increase the deployment of advanced network services. Sales of NGN equipment saw a rise of 70% in the last quarter. Media gateways, VoIP ports, and softswitches saw a major boost in sales and added up to 27.3 million pieces of equipment. Globally, the revenues increased by 49%. Revenues from VoIP touched $ 609 million in the third quarter. Apparently, the trend will continue as carriers look to upgrade their packet networks.

Huawei Technologies from China, Nortel, and Siemens continue to be the dominant players in the field. The Asia Pacific market grew by 202%, the Caribbean and Latin American market registered a growth of 81%; however, the American market reduced in size by 12% due to network consolidation.

VoIP checklist

Before deploying VoIP, it is imperative to verify that the wall jack, patch cable, switch ports, etc are working; power should flow from the switch to the phone; communication between the phone and server should be consistent; and baseline metrics should be established.

Once a network is deployed, the conversation between the phone and the network should be monitored; the call setup time should be assessed; QoS scores should be compared with objectives; and the baseline performance should be measured.

Universal Service Fund

Lawmakers in the US contend that the Universal Service Fund can be utilized for financing broadband access in rural areas. The USF contribution fund was originally created to foster the penetration of telephone service in rural America.

Local and long distance companies, paging companies, wireless providers, and payphone firms have been contributing to the USF since 1996. However, neither cable companies nor VoIP providers contribute to the USF. The payments to the USF are directed to the subsidizing telephone rates in high cost areas.

VPF

Very often when VoIP users are connected over different networks, their calls have to use the PSTN. Even though IP trunking is a standard feature with most VoIP equipment, frequently VoIP networks are not connected with each other. One solution that can help VoIP networks to completely bypass the PSTN involves the creation of exchanges that will enable multiple carriers to peer with each other.

Voice Peering Fabric (VPF) was launched by Stealth Communications in June 2005. VPF is an opportunity for VoIP providers to interconnect with each other. VegaStream and XO Communications are two new members of VPF. XO Communications is a major CLEC in the US. It chose VPF in order to gain access to a mature customer base and the opportunity to peer with a large number of carriers and enterprises. The VPF ENUM registry enables members to exchange VoIP calls with other VPF members without having to route calls through the PSTN. This leads to a reduction in operating costs and an increase in revenues. VPF also offers facilities such as the VPF Minutes Market that allows members to transact wholesale voice origination and termination services. Members can access CNAM, LNP, and other TCAP services with the help of the VPF ASP Market.

Organizations can use products from VegaStream to VoIP-enable the legacy PBXs. The companies can work with their existing infrastructure and connect with the VPF. Equinix and NeuStar have come together to initiate a VoIP peering effort. Equinix operates 15 IBX data centers that can be interconnected with every major global network for peering. NeuStar is developing SIP-IX application peering services that will offer addressing directory services and policy-enabled shared routing among other things. voipplanet.com reports:

While most IP-PBX vendors and VoIP services offer SIP peering on their own network or product, the NeuStar effort, like the VPF, is aimed at creating a wider peering capability.

Read More: VoIP Peering Market Pairs Up

Redback Networks

The SmartEdge Service Gateway platform from Redback Networks has been selected by Covad to boost its network capacity and make its IP-based services more resilient. voipplanet.com reports:

San Jose, California-based Redback Networks was founded in 1996 and claims a global customer base of over 500 service providers and carriers. Financial terms of the deal with Covad were not publicly disclosed.

Read More: Redback Provides Covad with Smart VoIP Edge

SmartNode

The SmartNode VoIP routers developed by Patton Electronics Company will now be able to offer triple play services by means of IGMP and IP-multicast in SmartWare release 3.20. Operators offering several services can make use of the new SmartWare to incorporate triple-play services to their existing offerings without making any further investments in the Patton customer-premise equipment.

The new SmartWare can be downloaded to the CPEs and will enable carrier-providers to deliver packet-based video-on-demand, IPTV, etc to a set-top box like the Aminet 110. SmartWare also reduces network congestion that operators face with unicast Video-over-IP. A single stream of video packets is transmitted to the hosts that request it, thereby eliminating duplication. tmcnet.com reports:

Because SmartNode offers converged-network solutions for both POTS and ISDN telephony, IGMP multicast in SmartWare makes SmartNode the first and only line of triple-play CPEs for the world's ISDN carriers and networks.

Read More: Patton’s SmartNode VoIP Routers Now Support Triple Play Via SmartWare

Network management

In order to deploy VoIP successfully it is important to have a long-term management plan in place. The plan should encompass factors such as tracking, supporting, and maintaining reliability across the network. A management plan should consider technical as well as personnel issues.

VoIP management at a basic level can be achieved by following the standard network management practices. VoIP management is related to successful management of the network resources. VoIP QoS has to measure up to the standards set by traditional telephony over the years. This requires effective monitoring of bandwidth utilization, CPU utilization, memory usage, etc.

SNMP, RMON, and Syslog are tools useful for managing a converged voice and data network. These tools notify in case of errors and yield performance related statistics. Enterprises can maintain network stability by acting upon the information provided by these tools. Adequate change control should be available to prevent voice outages and to resolve the issue should an outage occur. A network running on a single system is easier to manage than one that uses multiple operating systems. Standardized software and hardware also facilitates network documentation management.

In order to maintain a QoS, it is important that the network equipment facilitates traffic distinction. Data packets can be detected and prioritized using marking methods like CoS, port number matching, Differentiated Services code Point, etc. Ideally, the marking configurations should be standard across the network so that the management process does not get too complicated. It is important that the VoIP network be monitored consistently after being deployed successfully. Network traffic analyzers should be deployed to detect and rectify VoIP problems such as call-setup time, jitter, and delay. Products like Observer and Sniffer voice modules enable the capturing of voice data and decoding of voice protocols. Prognosis is an application that is capable of in-depth VoIP monitoring by calculating a MOS for each call.

Although, MOS values are subjective they help to devise baselines for quality standards. The baseline metrics can also be updated based on the information provided by these monitoring devices. Certain products generate simulations of different traffic conditions that can be used to test the readiness of the network before deploying VoIP or adding features to it.

VoIP in the enterprise

The selection of a VoIP vendor depends upon the deployment requirements and services offered by the vendor. VoIP deployments can be carried out successfully by referring to layout and design models for integrating VoIP and data networks. Also, by following vendor-neutral protocols voice traffic can be separated for ease of management.

VoIP can be deployed in a centralized or a distributed manner. A centralized model uses one or multiple PBXs for call setup and teardown. Call setup and teardown in a distributed model are performed from multiple locations. A distributed model is useful for sites that are similar in size and have predominantly local traffic. Links are established to ensure connectivity between the sites.

Centralized models offer the advantage of PSTN and hardware consolidation. Usually, more than one VoIP server is used in a centralized model in order to manage the load and to have redundancy in the network. The servers should be backed up with a UPS to counter brownouts and blackouts. Centralized models are appropriate for corporate headquarters or multiple small remote locations that can be connected by a WAN. WANs also enable the administration of the voice communications as well as call setup and teardown. Redundant WAN links are an added cost but they have to be present in order to ensure connectivity with the central site. Failover options help to bring down the cost of redundant WAN links.

Distributed networks can be managed with less redundant capacity. The multiple sites use VoIP servers present on location and communicate with each other by means of trunk lines. However, a distributed model is complex in design and can lead to high hardware costs. The design of the network equipment will have significant bearing on the successful integration of VoIP with another network. VLANs allow enterprises to segregate voice and data traffic and yet have them on the same medium. VLANs also facilitate the monitoring of the different traffic types.

The Layer 2 and Layer 3 headers in the data packets are used for establishing the QoS. With 802.1p, Layer 2 QoS is achieved by enabling multiple levels of prioritization to the traffic that enters the port. WAN links are examples of Layer 3 links in which the QoS is used for matching fields within the IP headers for the purpose of prioritizing traffic. The functioning of QoS processes is affected by the manner in which the bandwidth is distributed for voice traffic, data, and call setup/teardown activities. The quality of communication between sites depends upon the QoS policy implemented.

Compression helps to utilize the available bandwidth in a better manner by shirking redundant information. Usually, in voice packets, the headers are compressed. The choice of codec will affect the number of calls made with acceptable voice quality. G.711 is a high-quality codec that leads to high per second bandwidth rates. G.711 requires bandwidth at the rate of 90 Kbps for a one way call; as against this the G .729 uses 90 Kbps but it has a lesser sampling rate that result in low call quality.

A resilient VoIP network is critical for achieving a reliable service. In order to ensure the success of the deployed solution, the network should have redundant hardware and sufficient UPS backup. In case power over Ethernet (PoE) is used, backup should be provided for the local network switches that transmit the power to the individual telephones.

Assessing network capacity

Network capacity can be expressed in terms of the amount of traffic that it can manage. With VoIP, network capacity is measured with respect to the number of simultaneous calls that it can process.

VoIP capacity planning for a network should be based on the peak load that the network will be called upon to handle. Factors such as LAN/WAN design, existing data traffic load, type of voice codecs, hardware capabilities, redundancy in the network, etc need to be considered. The first step in implementing VoIP is to have knowledge of the bandwidth capacity of each link in the existing network as this helps in identifying the potential bottlenecks. VoIP deployment on a single-site network that has a high-speed infrastructure is not likely to be encumbered by network capacity; instead it is the network layout that may result in bandwidth-related issues.

VoIP communications on WANs can suffer as bandwidth bottlenecks are created on the serial-based connections that work using the T1 or fractional T1 lines. VoIP QoS guidelines state that voice traffic be prioritized but if this is followed, then data traffic will slow down during peak traffic hours. If standard PRI or voice T1 lines are used, it may not be possible to place additional VoIP calls once the available channels are used. If the number of VoIP calls is more than what the network can handle, users will face operator error or fast-busy messages. The voice quality will drop even if data is compressed and QoS standards are in place.

The mix of PSTN connectivity and additional bandwidth can be arrived at by evaluating the average and peak usage for each connection. The decision to either increase the speed of existing circuits or add more lines is influenced by the number of users in each location and peak usage. The increased use of services such as Metropolitan-area networks (MAN) will make the addition of higher bandwidth lines a cheaper alternative.

Bandwidth monitoring techniques should be employed to study the effect of VoIP traffic on bandwidth utilization. VoIP-specific tools are useful for recreating voice traffic scenarios, checking for errors, and monitoring for problems like jitter and delay. Call loads vary with the sampling rates provided by different codecs such as G.711 and G.729. Compression and call loads should be tested in real-time. Remote locations can either have PSTN connectivity or they can be managed from a central location by using a WAN.

Centralized PSTN connectivity helps to reduce deployment costs and increase network redundancy. The hardware should be able to cope with the increased overall traffic; the distribution-level points should not turn into bottlenecks. Almost all current Ethernet hardware supports a 100 Mbps connection to the phone. Modular network hardware allows users to increase the port density while using the existing equipment.

Sirocom

Shaw Trust, a charity in the UK that assists the handicapped to obtain work, is tying up with virtual network operator (VNO), Sirocom. The £2.5 m deal is for a period of four years and will link around 200 work-from-home employees and 1,300 office workers. The charity may also move its intra-company communication traffic to VoIP. Currently, it is using the PSTN.

The charity is growing at a rapid rate; the deal with Sirocom has resulted in savings and also improved its ability to scale. The move is also significant as the UK government has stated its desire of curtailing its contribution to the incapacity benefits. Currently, the government expenditure on incapacity benefits is approximately £12.8bn.

Broadband over Power Lines

Broadband over Power Lines (BPL) has been around for ten years. It is now being looked upon as an alternative to DSL and cable for broadband Internet access. BPL can be used to transfer radio frequency across power lines and its biggest advantages are its reach and an already established infrastructure.

More than 100 trials and pilots in 40 countries in Europe have attested to the feasibility of BPL. Germany and Spain already have commercial BPL networks. Pilots and tests at a commercial level are taking place in the United States. The acceptance of BPL will be affected by the absence of uniform standards and the chances of bandwidth and radio interference. Energy leakage in the power lines can lead to noise. BPL providers have to try and limit the interference so that it conforms to the levels allowed by the FCC.

BPL can be of help in enabling the rural population to access broadband who are currently dependent on dial-up. However, this entails setting up an infrastructure, which will incur costs that can only be met with government subsidies. In urban markets, BPL faces stiff competition from already established players. Being a new entrant, it is possible that BPL has not yet identified all the impediments that it may face while trying to gain mass acceptance.

HKBN

HKBN, which is the largest alternative residential voice and broadband service in Hong Kong, is planning to increase its capacity by 1 million subscribers by upgrading its VoIP network. HKBN had launched its 2b broadband phone service by deploying the Nortel Communication Server (CS) 2000-Compact along with other solutions. tmcnet.com reports:

HKBN's 2b Broadband subscribers will now be able to enjoy VoIP with advanced multimedia services such as secure instant messaging, video calling and broadband access rolled out to subscribers in Hong Kong, North America, Europe, Southeast Asia and Australia.

Read More: Hong Kong Broadband Network

VoIP peering

The charges of long-distance telephony will plummet steadily as peering networks grow. As enterprises and government agencies join peering networks, the benefits to be had from peering will increase. tmcnet.com reports:

That is where VoIP 2.0 comes in. The companies who make a living in communications tomorrow are going to have to be 100% committed to coming up with new applications that consumers and businesses will pay for.

Read More:Wake Up RBOC

Media Gateway architecture

A softswitch architecture enables the separation of the logical and physical switching functions, thus enabling these to reside in geographically separated devices. The physical connection to the LAN or the WAN is established by the media gateway that resides in the transport plane. The logical functions are managed by the call agent.

According to the ISO, a gateway converts protocols from one system to protocols for another system and can function at all of the seven layers of the Open Systems Interconnection (OSI) model. A gateway has two interfaces that may have to communicate with the cellular or circuit switching network on one side and a packet switching network on the other side. The gateway establishes the physical connection and also manages the speed accordingly. The gateways also convert the information present in TDM systems into packets used with IP networks. They contribute toward maintaining the QoS of the service by canceling line echoes and managing the jitter buffer.

NetStructure Host Media Processing

Intel is designing products that will help companies to implement VoIP and at the same time obtain the maximum benefit from their investments in legacy systems. Intel has released the NetStructure Host Media Processing (HMP) 1.5 software for Linux; it supports up to 120 video channels that can be utilized for services such as video mail, video color ring back, video caller ID, etc.

Media enterprises can avail VoIP telephony by using the host media processing technology provided by Intel. In the next three months, Intel will release the NetStructure HMP 2.0 for Windows and the NetStructure Digital Network Interface Boards. Intel has implemented a hybrid VoIP solution at its site at Parsippany, New Jersey. The solution will help bring down MAC costs by 72% and reduce the floor-space requirements for equipment by 89%.

Intel has released a reference design, the Converged Application Platform for the Distributed Enterprise, which aims to facilitate the deployment of multimedia services by SMBs. The design allows integration of discrete pieces within the network into a single device. The working prototype uses HMP and processors and should enable telecom vendors to access the market faster.

Compunetix Summit platform

Mercuri Teleconferencing is an established player in the business teleconferencing market and provides the Compunetix Summit platform, which can support 10,000 active ports. The company has now added an IP-based component, the SiteScape Zon platform developed by SiteScape Inc. The IllustrateTM Instant Collaborator provided by Mercuri enables audio conferencing, web conferencing with whiteboarding, etc. It is accessible over the PSTN and IP-networks. voipplanet.com reports:

"You can use it to integrate with Outlook and create buddy lists that allow for instant meetings, instant messaging, presence management, and even to start a conference call directly from the interface," Balaz continued.

Read More: VoIP Security Framework Emerges amidst Vendor Releases

Aruba Networks

Aruba Networks has designed a network architecture that enables mobile workers to VoIP and data networks from anywhere. According to Ken Dulaney, analyst, Gartner Inc. the new architecture makes it possible to unify access methods into a single system. eweek.com reports:

"Enterprises have traditionally thought about LANs and WANs and other networks as separate, but Aruba is trying to show that they can be seamless and unified, and that's good."

Read More: Aruba Extends Enterprise Networks

Amedia Networks

Amedia Networks is providing a standards-compliant VDSL2-based Ethernet Home Gateway, the HG-V100. It is meant for carriers that use IP DSLAMs in their access networks.

The HG-V100's customer interfaces include four 10/100 BaseT Ethernet ports (RJ-45) and two VoIP FXS ports (RJ-11).

Read More: Amedia Networks to Offer VDSL2 Home Gateway for Triple Play Services

Convergence in the universities

The residence halls of the Case Western Reserve University and Duke University have state-of-the-art wired and wireless data networks, streaming video, improved voice coverage over cell phones, etc. Future dormitories are being planned to facilitate VoIP networks. networkworld.com reports:

All areas in the residences are blanketed with an 802.11g wireless LAN (WLAN) based on 140 Cisco Aironet 1231g access points. Even the football and track fields are covered wirelessly by four Vivato VP2210 Wi-Fi base stations.

Read More: High-tech dorms move to head of the class at colleges

EarthLink

EarthLink, which uses DSL networks to deliver its services, was badly hit by the FCC ruling which stated that DSL providers do not have to offer discounts to ISPs. zdnet.com reports:

But EarthLink isn't taking the setbacks lying down. Instead, the company has been busy exploring new technologies that would allow it to bypass the cable and DSL networks altogether.

Read More: EarthLink aims to evolve

Broadband connectivity

Broadband is becoming increasingly user centric, with the vendors promoting it as a universal service delivery vehicle. The services offered include VoIP, data transfer and access, and video on demand. These services are bundled to form the triple play offer and are regarded by many providers as useful in maintaining revenues and retaining customers. The bundling of these services over a broadband network is resulting in changes in the access networks.

Services such as Video on Demand (VoD) require a deep fiber presence over the area covered by the service provider. The access devices are more intelligent in keeping with more stringent policy enforcement. The high level of interest in carrier-grade VoIP could be a harbinger for the decline of the PSTN. The success of the Digital Subscriber Line (DSL) technology has allowed service providers to deliver triple-play services using copper networks.

The IEEE 802.11 (WiFi) and 802.16 (WiMAX) standards have been successfully implemented which has added momentum to broadband deployment. A network that is run using broadband can be developed in two ways. The first alternative is to have a public network such as the Internet which can support all the services required if sufficient bandwidth is provided. However, such a network is disadvantaged by the fact that all the intelligence is present only at the boundaries or terminals. This model has been adopted by Vonage and AT&T CallVantageSM. The other option is to create a service-based segmentation that is achieved using the same techniques that are used to create VPNs. This ensures end-to-end delivery of a richer set of services in an intelligent network.

Broadband is also helping service providers to offer VoIP with or without multimedia features. SIP-based solutions like the Alcatel Intelligent Mobile Redirect (IMR) solution can route voice traffic over broadband and mobile networks depending upon the location of the user. A combination of broadband and narrowband networks can be used for a smooth transition from a PSTN to a converged network. The narrowband linecards can be used as VoIP gateways, enabling both PSTN emulation and PSTN simulation. In order to support the user and terminal mobility, which will occur as fixed and mobile usage converges, the broadband networks need to offer cheap bandwidth. There will also be a convergence of the access networks. lightreading.com reports:

Terrestrial and satellite broadcasting networks are evolving by adapting their capabilities to the mobile environment with the development of the Digital Video Broadcast – Handheld (DVB-H) and of the Satellite – Digital Multimedia Broadcasting (S-DMB) standards.

Read More: ALCATEL TELECOMMUNICATIONS REVIEW

Carrier technology

Ethernet-based carrier services first made their presence felt in the late 1990’s. These services were faster and cheaper than leased lines and Frame Relay services. The early carriers were required to support the complete range of the 802.1Q VLANs as well as a large number of MAC addresses. Later requirements are related to scalability and cost-effective operation of the service provider Ethernet backbones. These include VLAN translation and the ability to carry customer Spanning Tree BPDUs.

For Ethernet-based services to offer scalability, it required carrier-class switches with the same security features as provided by the traditional TDM and ATM switches. MPLS has the required attributes that enables it to support tunneling, traffic engineering, etc. Tags, known as MPLS labels, create a circuit that boosts IP with a connection oriented approach.

These tags are used for tagging the IP packets that enter the MPLS network. This reduces the workload of the hops in the network that only need to perform a lookup instead of the longest prefix lookup. An MPLS network offers the advantage of improved traffic engineering (TE) that enabled the placement of IP traffic on designated paths, thereby offering improved control without additional cost. lightreading.com reports:

Dynamic signaling protocols such as the Label Distribution Protocol [LDP] and Resource Reservation Protocol [RSVP-TE] allow tunnels to be set-up over such a routed network. Such tunnels can be protected with the use of back up paths or RSVP-TE fast reroute to deliver sub-1 second restoration time. These MPLS features paved the way for both traffic engineering and QoS support into IP. In 2001, such MPLS capabilities were added to Riverstone routers.

Read More: MPLS/VPLS Evolution: A Riverstone

NGN

Next Generation Network (NGN) refers to the IP-based network that could well supplant the PSTN network for providing telecommunication services. The NGN supports several multimedia services such as VoIP, videoconferencing, email, IM, etc. The ITU-T Recommendation Y.2001 defines NGN as follows:

A packet-based network able to provide telecommunication services and able to make use of multiple broadband, QoS-enabled transport technologies and in which service-related functions are independent from underlying transport-related technologies. It offers unrestricted access by users to different service providers. It supports generalized mobility which will allow consistent and ubiquitous provision of services to users.

MAC in a VoIP environment

VoIP implementation in the corporate sector will occur at the rate of 44% by the year 2008. The main driver behind the accelerated growth of VoIP is the savings that accrue in the adding and changing (MAC) process, reduced bandwidth expenditure, and savings due to lower manpower requirements.

According to data made available by Nemertes Research, an outsourced MAC operation can cost a company around $ 120 for a single MAC. MAC carried out in-house can be in the range of $ 30 - $ 90.

In a VoIP network, the cost per MAC is around $ 11 per hour. The time required for a single MAC in a VoIP environment is around 15 minutes. The moving and changing that is a part of a MAC operation can be reduced to zero if the phones are IP-based. This would allow the users shifting offices to simply plug-in their phones and log on.

Delay issues in VoIP calls

Delay is the most important aspect of the QoS expectations that a user has from VoIP. The two main causes of delay are echo and conversation overlap. Echo occurs when the reflections of a speaker’s voice do a round-trip from the other end of the telephone equipment. A delay of more than 50 milliseconds is considered to lead to significant echo problems. A talker overlap can become a serious problem if the one-way delay is of more than 250 milliseconds. In order to reduce the delay in a VoIP call, the end-to-end delay budget has to be considered.

Accumulation or algorithmic delay is a result of the time taken to gather voice samples that are to be processed by the voice coder. The delay can vary from 0.125 microseconds to several milliseconds depending upon the type of coder used.

The process of encoding the data and combining the data samples to form a packet leads to processing delay. The processing delay is influenced by the processor speed and the type of algorithm utilized.

The delay that occurs due to limitations in the network is referred to as network delay. The protocols that govern transmission of the data packets through the physical medium, the jitter buffers, and firewalls are the main contributors to network delay. In some IP networks, the packets can be delayed by up to 100 milliseconds.

Setting up a VoIP network

The following steps are executed to set up a VoIP system

• The first step is to use an analog to digital converter (ADC) in order to convert the analog voice waves to bits.

• The bits are then compressed into a suitable format using any of the protocols available.

• H.323 and SIP are signaling protocols used for the purpose of calling users.

• The sound packets are dissembled and the data is extracted at the receiver’s end in real time.

A card integrated ADC is used for converting analog to digital. The digital data is converted into a standard format that compresses the bulk data thereby increasing speed and also fosters greater acceptance. Pulse Code Modulation and Adaptive differential PCM are two of the compression protocols used.

VoIP architecture

VoIP systems can be divided into three major groups that include systems that have been developed from the traditional PBX platforms, systems evolved from the traditional data-switching networks, and VoIP systems that have been developed from scratch. Each system has its own merits, cost and implementation implications and can be used to achieve 99.999% availability.

The traditional PBX systems have consistently delivered 99.999% reliability that is now being used as a standard for VoIP systems. However, these legacy PBX systems operate independently of one another and cannot back each other up. Each PBX at a given site is in effect isolated from the rest of the network and is a single point of failure. It is a fragmented and centralized architecture.

Data-optimized switch platforms can also be used to transmit voice; however even though the bandwidth requirement of voice is small, it has a very low tolerance of delay. To develop a VoIP platform, a call control is implemented in a centralized server which is a single point of failure. The installation of multiple call servers can help to mitigate the risk of server failure but the call servers depend upon the availability of the IP network that connects remote offices. In the event of a WAN outage, survival mode features can be used to connect the IP phones. These features are optional and the voice quality is reduced. Setting up voice capabilities in a data network can lead to complexities that increase the difficulties in providing a high level of availability. Also, the reliability at every stage such as design and operation becomes difficult as several devices have to be integrated before one can achieve a decent level of voice quality using VoIP.

A voice platform created from scratch can leverage the inherent resilience of IP networks. IP voice switches can be used to distribute a voice system that runs on a peer-to-peer architecture, with more than one point of failure ensuring greater availability. A switch can also act as a standalone PBX capable of making best-effort calls using a failover PSTN trunk, in case the IP backbone is down. The switches work in tandem and are capable of providing PSTN access to a site that has its own switch out of order. This means that a native VoIP system can provide 99.999% of availability; the network will go down only in the event of a WAN outage or all the switches going down at the same time.

Voice quality and QoS in a VoIP network - part 2

IP traffic is classified in the following ways:

• Diffserv/TOS bits: These are used for prioritizing the traffic. RFC 791 describes the second byte in the header of an IP frame as the Type of Service (TOS) byte. This byte is used for assigning priority to the packet. The same byte is called the Differentiated Services (DS) field by RFC 2474. The second byte is used to signal the edge devices and to mark the high-priority traffic for the router to recognize it.

• RSVP signaling: Is used for checking and reserving resources such that the QoS requirements regarding jitter and delay are met. The intermediate routers receive instructions from the IP control flows that the RSVP introduces from end to end. RSVP works well in a private WAN where the applications use it to contact their TCP/IP stacks.

• Port numbers and addresses: These lead to a better handling of the applications marked by the destination port numbers. It is a simple IP QoS technique used by several edge devices to check the port numbers and addresses.

• RTP header information: Useful for managing audio data packets. The RTP header helps the receiver to gauge the timing of the original data and to manage out-of-sequence data packets. This protocol is useful in prioritizing streams of audio and video.

• Data content: Useful for quick transmission of binary data. It can analyze URLs to classify web traffic in a better manner.

• Data rate: Used for better management of low-volume traffic by applying the Weighted Fair Queuing technique. The dispatch of data can also be controlled by the applications either on every API Send call or at each connection.

• Buffer Size: For prioritizing the frames based on their size. It can be used to prioritize small frames over large frames; it works on the assumption that small frames are more time-critical than the larger frames. The size of the buffer can be controlled for each API send and receive call. For TCP and UDP, the default buffer size is 32K bytes and 8K bytes, respectively.

Devices such as traffic shapers and bandwidth managers classify the network at its edges and provide a central point of administration. The traffic can also be classified in the middle of the network by devices such as routers but such connections are unable to provide much information about the traffic.

The different classes of traffic are managed on the basis of the flow rate, paths, and queuing. These help in taking decisions regarding the reservation of bandwidth and the fixing of latency. RSVP, WFQ, LFI, LLQ, RED, and WRED are examples of queue-based traffic management techniques. The traffic shapers and bandwidth managers at the edge of a network manage the traffic on the basis of the flow rate. They can be used to limit the throughput of traffic on a given route. MPLS is an example of path-based traffic management, in which the traffic is identified at the network’s edge and allotted either a preferred path or a “best effort” path. The following things should be taken care of in order to maintain the desired QoS in a network:

• Look up for relevant information on QoS as it is a new and fast evolving technology.

• Classifying traffic should be need based and not driven by any other rationale.

• Given the numerous QoS schemes that exist, IT employees need to be aware of the parameters and tuning requirements associated with each.

• QoS should only be measured against the backdrop of a heavy load to confirm its configuration and if the desired classes are getting better handling.

Voice quality and QoS in a VoIP network – part 1

Data networks are usually measured in terms of various metrics and in the traditional telephony setup, the voice quality is assigned a single number signifying its quality. VoIP is a combination of both. Mean Opinion Score (MOS) is a popular subjective measurement of voice quality, prescribed by the ITU. Various approaches have also been developed to measure call quality objectively. These include the Perceptual Speech Quality Measure (PSQM), Measuring Normalized Blocks (MNB), Perceptual Evaluation of Speech Quality (PESQ), Perceptual Analysis Measurement System (PAMS), etc.

PSQM and PAMS use digital signal processing algorithms that compare telephony signals with an analog reference signal. However, these techniques are not very well suited to test the quality of VoIP calls, they are unable to track and report on the issues of delay and jitter. An actual two-way conversation cannot be measured satisfactorily by these techniques and these are not scalable. Delay in conversation comprises the following components:

• Propagation delay: This delay is proportional to the speed of light and depends upon the physical distance between the two communicators.

• Transport delay: Transport delay occurs because of devices such as routers, firewalls, traffic shapers, etc. The delay can either be constant or vary with the traffic.

• Packetization delay: This is a function of the codec speeds. Low-speed codecs, such as the G.723, take around 67.5 milliseconds to convert analog signals into digital packets. The extra time is required because these packets have to compress the packets to reduce their size. High-speed codecs such as the G.711 can do the packetization in approximately one millisecond.

• Jitter buffer delay: A jitter buffer helps to minimize the variations in the arrival times of the voice datagrams. However, sometimes in the event of excessive delay, packets have to be discarded.

Audio signals degrade to a greater extent with low-speed codecs due to lossy compression. VoIP datagrams travel using RTP and are lost in the network either due to congestion or due to arriving late at the receiver’s buffer. Attempts at improving voice quality should consider factors such as the total one-way delay in both directions, delay variation, and packet loss in bursts.

VoIP quality is tested by conducting a VoIP Readiness Assessment, which has three stages. In the first stage, if the MOS value for a single VoIP communication is low, it implies that the data network needs to be upgraded. In the next stage, the MOS value is acquired for several call volumes at peak traffic. The third stage is to test the network performance by running concurrent calls with heavy background traffic. It is not often that router-based data networks deliver VoIP calls with toll-quality. Issues with jitter and packet loss have to be sorted out before a fair number of calls with good quality can be supported by the network.

The bottlenecks in a VoIP network can be removed in various ways including increasing bandwidth, replacing equipment, reconfiguring the network or changing its layout, etc.

Bandwidth consumption with VoIP calls depends upon the codecs used and is invariably greater than what is assumed. One can safely assume 160 kbps bandwidth consumption with a G.711 codec and around 50 kbps with the low-speed codecs. Bandwidth can be conserved by using the following techniques:

Compressed RTP headers: These help to conserve bandwidth by reducing the size of the RTP headers to around one-tenth of their original size. However, this technique increases latency and can add to the delay.

Silence Suppression: This helps to reduce the size of the payload by making use of the times in a conversation when both users are quiet.

RTP multiplexing: It conserves bandwidth by placing multiple packets of voice information in a single datagram. This reduces the number of IP/UDP headers required for each audio packet. However, this leads to delay as the data can be sent only when multiple packets have been generated. Also, even a single datagram lost implies the loss of several audio packets.

Call Admission: This helps you to manage the number of calls such that the bandwidth is not overloaded by having to support several concurrent conversations. This frees the bandwidth for other network activities.

Sometimes, it is possible to boost network performance by upgrading the existing infrastructure without having to invest in additional bandwidth. Upgrades can help to reduce latency and increase capacity as well. In a heavily trafficked LAN, switches are a better option to manage IP multicast traffic as compared to hubs. Increasing the RAM of a router can be an inexpensive upgrade. A hardware-based firewall has greater capacity than a software-based one and should be installed as it reduces transport delay.

Changing the network architecture is a major decision and should be considered under circumstances where a more direct route with fewer hops could be implemented thereby controlling transport losses and clustering of traffic patterns to explore the possibility of placing the servers closer to the clients, this will help in reducing the backbone traffic.

The QoS of network devices should be maintained at all times to ensure consistency of performance. The type of traffic dictates the nature of QoS technique to be applied. A network that does not have QoS standards treats all traffic with equal importance and is referred to as a “best effort” network. With a QoS standard in place, a network either reserves bandwidth for premium traffic or gives it priority in the event of congestion.

Non-integrated and integrated IP networks

Service providers are increasingly switching over from a circuit-switched network to a packet-based network. The switch in technology has to be made in the face of the following challenges:

• Obtaining the maximum returns from the existing TDM technology during the course of making the switch to a packet-based network.

• The new technology should provide the same level of redundancy and service levels as TDM technology, which is achieved by integrating the diverse elements of a packet-switched network so that they perform as a whole.

The IP network can be either non-integrated or integrated. Both networks have the drawback of a single point of failure, which is the centralized call server that does the routing and signaling. In a non-integrated platform, an increase in the number of users leads to a degradation of network. This affects network performance and increases the time required for the call setup process and an increased number of call requests consume more bandwidth. The choking of the network that results from the bandwidth unavailability can lead to an up to 80% reduction in call-handling capacity and dropped calls. Integrating a network can lead to a forced reliance on centralized call-routing processors, which can affect network efficiency. If the components have been procured from multiple vendors, integrating them requires greater effort and the complexity of the network increases the cost of ownership of the network.

An integrated IP/TDM switching platform provides capabilities such as signaling, multiple transport protocols like SIP and H.323 packet aggregation, IVR, etc. An integrated IP network allows the service providers greater scope to expand their network and is easier to manage. The call setup time in an integrated system can be as low as 100 milliseconds as compared to 4 seconds, which can sometimes happen in a non-integrated network. The call completion in an integrated network is good because of the redundant nodes provided.

The bandwidth signaling in an integrated network is better as there is no need to constantly signal to the other network components. This is because all the components are housed in a single box. A non-integrated network requires separate servers for IVR and SS7 support, which adds to the cost and complexity.

Converged networks and converged threats

VoIP networks are susceptible to threats that originate at the data networks. Companies are well aware of and therefore prepared to handle the threats that originate in the VoIP networks but managing the threats that can occur due to the convergence of networks may well be a hidden cost that most companies need to factor.

The traditional tools of protecting data do not afford adequate protection to VoIP data. In fact, even modems that can be accessed via phone lines possess a threat to VoIP data. A hacker can use the company’s phone lines to make expensive long distance calls, misuse the bandwidth, and commit subscription frauds. A converged network brings together data and voice networks and in doing so, it exposes one network to threats from the other. A thorough consideration of the threats helps to deploy the necessary security tools, often without much additional expenditure.

SecureLogix Corp. is a company that offers VoIP monitoring products that manage the traffic that passes through voice gateway devices in a manner that is similar to how firewalls manage data traffic. Other security measures include boosting perimeter defenses, encrypting data, hardening servers, etc.

Power and cooling solutions for VoIP telephony

In order to supplant the existing telecommunication systems, VoIP telephony has to not only satisfy the QoS mandates but also ensure that the system makes efficient use of power. The patch panels and hubs in the legacy wiring closets will be replaced by routers and UPS that will require sustained cooling to ensure their smooth working. An IP telephony network is made up of layers and has 4 physical locations. IP phones, access layer, distribution layer, core switch, server farm, and call servers make up the different layers. The four physical locations are desktop, wiring closet, main distribution facility, and the data center.

IP phones need around 7 watts of power. The IEEE 802.3af has stipulated that a maximum current of 350mA can be drawn by these endpoints via CAT5 cables. This standard allows for 15 W of power to be delivered to a distance of 100 m. The communication devices are powered either by the data lines or by the wall outlets. In-line power does not require power at the desktop as the instrument draws power from a network switch that is run by a UPS system. If the communication device draws power from a wall outlet, a UPS, with a battery that can run for extended periods, should be provided.

The wiring closets have distribution switches, hubs, patches, routers, etc. Compared to the legacy systems, IP telephony systems use and release more power. Equipment that can be stacked in 1-3 racks can use up to 4000 W of single phase AC power. The power drawn can be at either 120 or 280 VAC and is a function of the network architecture and the switching technology used. Providing circuit breaker protection and the correct receptacles, for example L5-20, L5-30, L6-20, etc is important. A UPS system should be available to protect the system. Factors that affect the choice of UPS include the power requirement, the run time, redundancy to be incorporated, etc.

APC Smart-UPS is a rack-mounted UPS that will ensure 99.99% availability whereas the APC Symmetra RM, which is N+1 redundant, will provide 99.999% availability. Critical applications such as the 911 service may require a higher percentage of availability that may well go into 7 nines. Such requirements can be fulfilled by using dual UPS and dual network switches as well as backup generators.

Rack PDUs should be used only if there is a lot of equipment and the equipment cannot be plugged directly to the UPS. The PDUs should have a meter that displays the power consumption and lessens the possibility of overloading due to oversight. In order to address the problem of cooling the closets, the power dissipation needs to be calculated.

For conditions where the heat load is for less than a 100 W in the closet and the rest of the structure is properly conditioned, wall conduction can provide sufficient cooling. For the same heat load, if there is no HVAC system and the building is not properly conditioned then a small air-conditioner can be installed in the closet. If the heat generated is for greater than 1000 W and a dropped ceiling HVAC system exists with the remainder of the space being conditioned, then to manage the conditioning, the bottom portion of the closet door should be fitted with a vent grill and the rack on which the equipment is placed should carry a hot exhaust air scavenging system.

Important VoIP equipment such as the layer 3 routers and switches is housed in the point of ping (POP), also known as the main equipment rooms (MERs). An MER may have equipment that may use up to 40kW of power and may occupy close to 12 racks. An MER may or may not have a UPS or even sufficient battery backup. In order to ensure 99.999% of availability, a modular UPS system with a backup of at least 30 minutes should be provided. Hot spots can occur at higher density racks; these can be avoided by using air distribution and removal units.

Data centers house sensitive equipment like the application servers and the related software. A large data center may house hundreds of servers that support ERP, WMS, and CRM applications. The data centers can draw more than a 100 kW of 3 phase 480VAC power. The addition of a VoIP network can result in incremental load on the data center necessitating longer runtimes. A separate UPS should be provided for the IP telephony systems that should ideally be housed separately. Redundant air conditioning systems can be installed to ensure higher availability.

VoIP network implementation

VoIP network implementation In order to ensure a high availability of VoIP networks, the following factors must be considered:

Components of call processing: The different components that make up voice systems have different requirements that need to be fulfilled before implementing a VoIP network. The components include voice mail, toll bypass, call center applications, etc. Centralized call processing consists of IP-PBXs, soft phones, etc in centralized locations and the clients can access them through a WAN. Another alternative is on-site call processing, which may be an expensive option if there are too many sites in a WAN. Survivable remote site telephony (SRST) enables centralized call processing but if there is an outage, the phones can still connect via a local router.

Cost: The cost of establishing a network depends upon factors such as the level of redundancy that needs to be incorporated into the network. The network has to be architected keeping the budgetary constraints and ease of use.

Network architecture: WAN and LAN networks are possible. The QoS levels are achieved in different ways for these networks. LANs provide star, bus, and ring topologies. WAN topologies are represented in terms of their technology, examples being Frame Relay, MPLS, etc. LANs use a load-balanced backup circuit in which there are two circuits to share the traffic load. Bandwidth is not a problem with LANs and the load-balancing circuits are relatively inexpensive to install and maintain. WANs use active/passive networks in which the passive network comes into play if the primary circuit experiences an outage. WANs have limited bandwidth to work with and employ expensive circuitry. One has to consider the routes that the circuits will take as these will affect the latency. Network administrators have to ensure that backups provide voice quality similar to that provided by the primary networks. In order to ensure consistent voice quality, the Resource Reservation Protocol should be used for the backup circuit and the utilization of the active circuit should be managed such that in the case of an outage, the increased load on the secondary circuit does not affect the quality of transmission. Another alternative is to route calls over a PSTN network. The choice of backup is a function of costs involved and business considerations.

Cloud Diversity: Carriers such as MPLS, Frame Relay, and ATM are maintained by the carriers. It is safer to have a backup circuit that uses a different cloud than that of the active circuit. This is particularly useful when the entire WAN setup has been provided by a single vendor. Introducing different routings may complicate the network and increase cost but will allow for greater protection against service outages that may happen if a cloud does not work.

Different service layers: Voice and other data traffic should have different access layers. By moving the IP-PBX cluster away from the office network, it is possible to develop an abstracted layer that permits the voice service to continue operating even during maintenance work. The network should be designed after considering the functionalities of the core, access, and distribution layers. Voice virtual LANs provide another layer of abstraction and are often a basic requirement for availing the many proprietary benefits that vendors offer.

DHCP: A Dynamic Host Configuration Protocol (DHCP) provides information regarding the call processing to the VoIP networks. VoIP networks have to strike a balance between the DHCP and DNS service. The DHCP service should be able to cope with an increase in phones in the network.

Subnets: Subnets can sometimes act as points of failure that may affect network performance. Servers and desktops can be given the IP address of the IP gateway either on purpose or by mistake. This can lead to a traffic overflow, which may bring down the subnet. A safeguard is to reduce the size of the subnet, for example one subnet per switch.

Power Consumption: Most networks draw power from the Ethernet (PoE) by using switches. The amount of power drawn should be carefully regulated, especially if large chassis switches are used to provide power to a large number of phones.

Causes of jitter and methods of jitter measurement

Jitter is the variation in the transit delay that packets experience while traversing a network. It is caused by queuing and the serialization effects on the packet path. Class based queuing, reserving bandwidths, high speed links like SDH and E3/T3 are some of the QoS initiatives that will help in controlling jitter.

Jitter is of the following types:

Constant jitter: In this, the variation in delay is more or less constant.

Transient jitter: An unnatural incremental delay, sometimes only by single packets.

Short term delay variation: It occurs due to changing routes and exhibits increasing delay for some packets as well as an increase in packet to packet delay.

Examples of delay

System packet scheduling delay: It is a transient jitter. VoIP with soft phones often experiences jitter as more than one program may be running on the CPU, thereby slowing it and transmission time jitter is introduced.

Congestion in the Local Area Network: It is a transient jitter and occurs for short durations. It is governed by the maximum back-off time and the delay between packets. If the LAN cannot be contacted by the VoIP endpoint and the back-off time limit is reached or if another packet is ready for transmission, then the previous packet may be dropped. A 10 Mbit ethernet has a high back-off time as compared to the VoIP packet spacing and hence the jitter limits are governed more by the packet spacing and are usually in the range of 10-30 milliseconds.

Firewall routers: It can lead to a transient delay as well as short term variations. Firewall routers such as double socket routers reestablish an IP flow on the inner side of the firewalls after they have terminated it on the outer side. This helps in regulating the payload that gets forwarded to the inner networks. However, this leads to variable delay.

Access Links: These lead to short term variations and are often responsible for jitter as they constitute a bottleneck in the network. As ISDN and cable modems have bandwidth problems, the jitter introduced due to access links can be severe, sometimes up to 30 milliseconds of delay for each packet.

Load Sharing: Load sharing between IP service providers can lead to a constant jitter. Sometimes, multiple access links are routed through one IP service provider and this can lead to jitter if the delays across the links differ.

Load Sharing by an IP service: It can lead to a constant jitter. When IP service providers route traffic over more than one internal route in order to even out the load on the network, the difference in delay on each route can lead to delay.

Load Sharing within routers: It results in a constant jitter. When routers process packet in multiple queues in order to boost router capacity, it can lead to low levels of jitter. In order to support high capacity some routers employ a multi-processing approach in which packets are processed by multiple parallel queues. This can introduce low levels of jitter due to short term differences in queue size.

Routing table updates: These can lead to transient jitters. Routers perform periodic updates in order to ascertain packet priority and dispatch the high-priority packet first. This can lead to a delay in the transmission of some packets and sometimes some packets can experience very high delays.

Route Flapping: This causes transient jitters and can be traced to varying levels of congestion and link breakdowns. Route flapping occurs when a routing table is updated and is characterized by a low frequency oscillation.

Timing Drifts: It causes transient jitters and can result in "jitter buffer events", in which the buffer can either be overfilled or it has excess capacity. The timing can be reset if an NTP server is used.

A few approaches used for measuring jitter have been described below:

RTCP jitter is measured in terms of packet to packet delay. If we consider the delay between two consecutive packets to be Ta and Tb, then the variation is represented as abs(Tb-Ta). The mean of the packet to packet delay variation can be given by MPPDV = mean( abs(ti – ti-1) ). The MPPDV in this case represents the jitter levels in scenarios in which the packets arrive early and late in an alternate fashion.

The mean absolute delay variation (MAPDV) for a packet having a nominal arrival time of ai is given by mean(abs(ti - ai) ). In case of a route change, the value may not be an accurate estimate of the jitter buffer size or discard rate. Jitter buffer behavior can be understood in a better manner by considering the MAPDV in relation to the average or the adjusted value.

An alternative approach is to determine the mean absolute packet delay variation with regard to a short term average or minimum value – termed here the adjusted absolute packet delay variation. This can provide a more meaningful relationship to jitter buffer behavior.

If the nominal arrival time (denoted below ai) for a packet is known or can be determined then the absolute delay variation is abs(ti – ai).

The mean absolute packet delay variation is therefore:

MAPDV = mean( abs(ti – ai) )

This value can be misleading if a delay change occurs (e.g. route change), as a constant offset would be included. As even fixed jitter buffers can adapt to delay shifts, this means that the reported jitter value would not necessarily be a good indicator of ideal jitter buffer size or discard rate.

In a given time interval, the difference between the minimum and maximum transaction delays is given by IP Delay Variation. The time gap between successive measurements has an effect on the IP Delay Variation readings.

A Time Series Analysis is an alternative method in which sequences are fed into a filter function and matched with the data that is being modeled, for example with a Moving Average filter function. The jitter can be modeled as the sum of the processes that occur randomly. This will help in understanding the time varying nature of jitter and can also be used to emulate the jitter in IP networks. By modeling the jitter buffer operation, it is possible to estimate the packet losses in a live scenario. This helps to speed up the measurement process as there is no need to relate the jitter metric to a discard rate.

Jitter buffers are used to reduce jitter from the voice stream; however, in the process of reducing jitter, the buffers can increase delay and packet loss. Jitter buffers are either adaptive or fixed. Adaptive jitter buffers can vary their size as per the amount of traffic. Jitter buffers can adjust automatically with the delay in traffic, this permits the data to be retained for maximum time before it has to be discarded. The jitter buffer is sensitive to the recent minimum delays and is aware of the maximum permissible delay. This helps it to adjust to any changes in delay.

An increase in jitter levels or the presence of a discard event is a trigger for adaptive jitter buffers to react. In the presence of a discard event, the jitter buffer size is increased. For jitter events that happen close to one another, an adaptive jitter is preferred; however, for jitter that occurs over a period of time, as in a LAN, increasing the size of the jitter buffer may lead to delay. Jitter modeling should be such that IP network emulation can be carried out with the help of data obtained by using a time series model. The impact of jitter can be measured on a VoIP service by using a jitter buffer emulator that can deduce the number of packets that will be discarded.

VoIP, Inc. launched Network 911 Service for VoIP calls

VoIP, Inc has announced the lunch of industry's first private network 911 service for broadband and packet communications. VoIP, Inc is a global provider of VoIP. The private network service 911 is provided by its subsidiary, VoiceOne Communications LLC. The Industry now-a-days has focused on creating quick solutions to meet FCC deadline which require all VoIP service providers to offer 911 services to the customers.

VoIP, Inc's VoiceOne network provides five entry points for IP 911 calls to enter the network through the Internet. These entry points are placed to provide the shortest possible path for a phone call originating anywhere in the US. Carriers and service providers can manage the entire 911 service feature set through the VoiceOne web portal. It also offers the XML provisioning system, which is not offered by solution providers in the 911 industry today. businesswire.com reports:

Currently VoiceOne is working with select carriers and service providers in the industry through its 911 product trials. VoiceOne plans to release the entire product at the CompTel/ASCENT Fall 2005 Convention and Expo in Orlando, Fla., Oct. 9-12, where demonstrations of the provisioning interface, reporting, etc. will be demonstrated at the VoiceOne Booth (#500).

Read More: VoIP, Inc. Launches Industry's First Private Network 911 Service for VoIP Calls

3com launches new VoIP Applications

3Com has launched a new set of VoIP applications. Using these applications, users will be able to gain access to to a company's converged communication network. The new applications are designed to increase employee productivity. At the same time, they will reduce the costs and build stronger customer interaction irrespective of the location of the employees. The applications are targeted at the branch and mobile workers because employees can make call using IP telephone system from anywhere.

Mobile and remote users can also redirect an office call to a mobile phone, hotel room or any other office. Further, they will be able to use corporate IP telephony applications like IP messaging; IP conferencing and IP contact centres. These features will help to enhance the mobility. informationweek.com reports:

3Com and Ingate Systems are addressing an important need for enterprises to provide an open standards-based telecommuting solution which improves business processes, Zeus Kerravala, vice president of enterprise infrastructure for The Yankee Group said in a statement. It is imperative today that enterprises find a way to help their associates who telecommute and travel a great deal to remain productive while accessing their communication applications in a secure, converged fashion.

Read More: 3COM Rolls Out New VoIP Applications

PABX systems are on the way out

PABX systems have been the backbone of communication in most corporates. However, the influx of new technologies such as IP networks and SIP is making IT managers evaluate their options. In many cases, PABXs are being saved from outright scrapping by the fact that they are an expensive technology and companies prefer to recover their money on them before deciding to upgrade. The rate at which the upgrades happen is also dictated by the complexity of the process. According to Mick Reegan, chief convergence officer at Nortel Asia Pacific, approximately 50% of the corporates will adopt IP based systems that will support a converged environment. The real value from implementing IP technology will be derived when applications and voice are seamlessly integrated. This may be achieved by opting for IP-enabled PABXs or IP networks only. Siemens plans to achieve this integration through its OpenScape product that will combine its data and voice networks. Yet, it appears that using an IP-enabled PABX may not be such a good idea. This is because IP-PABX infrastructures are developed on proprietary instead of open standards. Therefore , the rate of growth of IP-PABX will not be able to keep pace with the developments in telecommunications. According to a report by Gartner, voice communication in future will be dependent on applications and not infrastructures, this cannot be achieved by persisting with IP-PABXs. Instead SIP technology will be one of the important drivers influencing voice communications, this is because SIP enables applications to work with voice regardless of the equipment used. Moving to an open infrastructure enables companies to envisage new uses for their software applications and work in a more scalable and flexible setting. Integrating applications such as databases and email clients with voice call allows for improvements in processes and productivity. Robbie Kruger, development manager at Avaya solutions feels that companies will shift entirely from a legacy hardware to a server-based environment. In this regard, companies have to decide between purchasing the equipment and a hosted service. When the idea of hosting, also known as IP centrix was first mooted, it was greeted with considerable enthusiasm. It was predicted that by 2008, approximately 25% of the companies would opt for IP centrix in order to cut costs as they would not be required to install any voice switching equipment at their space. However, security concerns have resulted in a mixed response from companies. An alternative is a grouping of hosted systems that manage voice with in-house systems that manage databases. This is facilitated by the fact that the systems are being developed on open standards and therefore different systems can be linked to meet a company's specific requirements. The next step in wireless VoIP telephony appears to be the fixed-to-mobile convergence, which should eliminate the need for a landline phone. A handset can be configured through a wireless network when in the office and can be switched to a roaming network when the employee is on the move.   Another option is to install a system that routes all calls over a mobile network. Ericsson has installed such a system for a software firm HP, in Sweden, which has resulted in savings of almost 40% for the company because of reduction in call costs and by being able to reduce the number of PABXs in the company. Mobile carriers that offer this "one -phone" service allow free calls between users in a company. Moreover, PABX functionalities like auto call-backs and conferencing are available on these phones.

Turinco is exploring VoIP opportunities in Latin America

In a significant development, Turinco Inc. has announced that it is exploring VoIP opportunities in Latin America. Turinco has acquired Arvana Network Inc., a voice service provider. Arvana will focus it business operations on the vast market opportunities that exist in Latin America. It intends to make a mark in the growing international market for digital telephony solutions using VoIP. As the VoIP service providers can provide telephone and other related service at much cheaper rates, Turinco eyes to reach a large number of consumers through Arvana.

The company hopes to generate revenues by providing a number of services to the subscribers. These include cheaper outbound and inbound calls, monthly subscriptions with free minutes for domestic and international calling. It also plans to launch long distance/international calling cards for corporate and consumer markets. tmcnet.com reports:

Arvana's strategy is to partner with in-country companies that already have the components required to provide VoIP services. Partners will typically be broadband Internet service providers (ISPs) that can deploy voice services over existing data networks without requiring any specific telephony expertise or infrastructure.

Read More: Turinco to Develop VoIP Opportunities in Latin America

VoIP Testing Is Essential

Anyone who has had experience with networking would tell you that it is always best to test things before fully implementing them.  If you don't test your hardware or software configurations, you may just end up with a lot of machines connected by wires that aren't talking and a major headache.  This mentality is something that needs to be taken into account with VoIP as well.  Unfortunately, many companies are finding out the hard way that their standing network can't control the live streaming packets that VoIP creates.  If the network becomes too laden with traffic, not only will the voice transfer be lost, but most information sent between machines will be terminated.  Lost information means lost revenue, and then a backlash to the VoIP machinery for destroying their peaceful network.

In fear of backlash from unhappy customers, Cisco systems, a major manufacture of VoIP equipment is pushing the idea of testing before implementing. Although Cisco can only do so much to promote the test phase, they are beginning to hold their distributors accountable.  While there hasn't been any refusal of services to distributors for not following Cisco's marching orders, the notice is meant to be taken seriously.  It seems like good business because both Cisco and the distributors have an equal share in the VoIP market.  Both can not exist without one another, and neither can last without consumers.  According to itBuisness:

”I wouldn’t let them do it,” said Zeus Kerravala, vice-president of enterprise infrastructure at the Boston-based Yankee Group. If users are having problems with the quality of the voice service, they will blame the manufacturer of the IP telephony equipment, he added.

Read more at: Cisco urges channel to do VoIP homework