Communication Systems

A Networking Idea That's Out of This World

2009-03-20T23:07:00.000-07:00

Millions of miles above us, a new networking technology is taking shape that could one day help improve how applications are networked on Earth.

NASA is testing a network layer technology that can withstand the rigorous demands of space communication better than the standard TCP/IP protocol, which dominates terrestrial (non-space) networking technologies.

Officially called Disruption Tolerant Networking (DTN), the technology went through testing earlier this month with a space probe that is currently more than 20 million miles from Earth.

DTN uses a different kind of approach than TCP/IP for packet delivery that is less cumbersome and more resilient to disruption than TCP/IP.

"In fact, far more research has been done to date on the application of DTN to terrestrial communication problems than on its use in space flight missions," said Scott Burleigh, a senior engineer for the Deep Impact Networking Experiment of NASA's JPL (Jet Propulsion Lab). "DTN has potential benefits in providing connectivity to parts of the world that are under-served by existing network infrastructure, in supporting oceanographic research, in tactical military communications, and more," he told InternetNews. com. "It's a pretty active field."

The basic idea behind DTN network endpoints aren't always continuously connected. In order to facilitate data transfer, DTN uses a store-and-forward approach across routers that is more disruption-tolerant than TCP/IP. However, the DTN approach doesn't necessarily mean that all DTN routers on a network would require large storage capacity in order to maintain end-to-end data integrity.

"It's always possible to have a DTN router that happens to be in constant communication with all of its neighbors over links on which round-trip times are very short, in which case very little storage would be needed," Burleigh explained. "All the bundles it received would immediately be forwarded, much as in an Internet router."

For complete story: Visit: http:/ /www.devxnews. com/article. php/3787641

IPv6… the Year NAT-Enforced IPv4 Dam Showed Seepage

2009-03-18T21:49:00.000-07:00

We don't give enough credit to people who will sacrifice themselves trying to plug the IPv4 dam with some NAT-putty. They even dream of a NAT66 filled afterlife. The growing IPv6 traffic trickle was given evidence at the recent RIPE 57 meeting in Dubai in a number of presentations, including a most edifying Google presentation (see Global IPv6 statistics by Lorenzo Colliti, and the related CircleID post). Noteworthy to see France with a 0.65% IPv6 penetration, largely courtesy of free.fr, a major ISP offering an extremely successful triple play service based on their customer premise freebox which they decided to also IPv6 enable. And, oh yes, 95% of French IPv6 traffic is native. In the meantime it felt good to see the US and Canada doing quite well with a 0.45% penetration. Major difference with France is that here in North America 95% of the traffic was 6to4. Most likely the popularity of Mac's and the Airport Extreme has something to do with it. When ranked by operating system, Mac OS leads in IPv6 penetration with 2.44% followed by Linux and Vista while XP and Windows 2000 are negligible.

The rather modest showing of IPv6 powerhouse Japan with only 0.15% IPv6 penetration was rather surprising and warrants some further analysis. The other real surprise was to see Russia claiming the overall number one ranking with 0.76% penetration!

Besides access and OS support, the third variable in the equation is the IPv6 routing between ISP's and their respective Autonomous System Numbers. There was a lot of speculation about the "brokenness" of IPv6. Google measurements show 0.09% of clients lost and 150ms extra latency; some way to go but not that bad and improving. Major tier1 ISP's including AS6453 are dual stack and peer with each other in both IPv4 and IPv6; we also see a growing number of IP transit customers upgrading their connections to dual stack.

When the fourth variable, IPv6 accessible content, will be in place the dam burst will come tantalizingly closer. IPv6 accessible websites and e-mail are not commonplace yet but slowly growing. And interesting phenomena happen to CDN's when issuing AAAA addresses to content: immediate increases in IPv6 traffic are visible!

2009 will see the seeping become leaking, 2010 will witness the first serious cracks, 2011 will see the dam buckle, 2012…

In the meantime I wish you all a happy and IPv6 filled New Year.

U.K. mobile operator charges £2 per day for 500 Meg of data; offers unlimited WiFi through The Cloud.

2009-03-17T21:45:00.000-07:00

O2 claimed on Monday to have the cheapest prepay mobile broadband offer in the U.K., with the launch of a USB dongle priced at £29.99.

The mobile operator's Pay & Go package gives users the sub-£30 modem with various data offers; users can take 500 megabytes for £2 per day, 1 gigabyte for £7.50 per week, or 3 gigabytes for £15 per month.

O2 has slashed the cost of the USB dongle itself since it first entered the mobile broadband sector in April.

At the time there was no prepay option, and the nearest offering took the form of a £20-per-month rolling contract, with a modem priced at £119.99.

The dongle announced on Monday, which provides download speeds of up to 3.6 megabits per second compared to top-line HSDPA speeds of up to 7.2 megabits per second, is aimed at light users and the youth market, O2 said.

"Pay & Go will appeal to those customers who want to snack on the Internet without the need to commit to a long term contract," said Peter Rampling, O2's marketing director.

"Our younger customers want to live online and can do so with Pay & Go at affordable prices," he commented.

Each price point also gives customers unlimited WiFi access via O2's partnership with wireless hotspot operator The Cloud, which provides coverage at more than 6,000 locations.

Bundling unlimited WiFi access with low-cost mobile broadband packages could deliver an important boost to customer satisfaction in the U.K; user reactions have been mixed due to complaints over connection speeds and high prices.

http://www.totaltel e.com/View. aspx?t=2&ID=102048

O2 has also launched a network coverage checking service, and a 50-day happiness guarantee, which allows customers to return the device within 50 days of purchase for a full refund with no early termination charges.

The operator said that since launching these measures in November it has seen a considerable drop in returned modems, because its customers are better informed.

Meanwhile Virgin Mobile also announced Monday a new tariff for what it claims is unlimited Internet access specifically from handsets rather than via USB modems.

The MVNO said that for 30 pence per day, both contract and prepay customers can access any Website from their mobile phone.

The new tariff coincides with the launch of Virgin's new mobile Web portal, which provides links to branded content including news, music and games, and incorporates Yahoo's mobile search service.

"We are giving all our customers the opportunity to use the Internet on their phone, without having to worry about racking up huge bills or working out complicated price structures," commented Graeme Oxby, managing directory of Virgin Mobile, in a statement.

However, a fair usage policy of 25 megabytes per day applies, and users that wish to download more data will be charged at £2 per megabyte.

Creating a High-Speed Control System to Test MEMS Microshutters Using NI LabVIEW FPGA

2009-03-15T21:32:00.000-07:00

"The LabVIEW FPGA Module and PXI-7813R saved hundreds of man-hours and thousands of dollars over custom chip development. In addition, we can inexpensively modify the control algorithm to improve testing, explore shutter issues, and further NASA MEMS microshutter array development."

The James Webb Space Telescope (JWST) is the next big telescope at NASA. More ambitious than its predecessor, the Hubble Space Telescope, NASA will place the JWST at a stable Lagrange point approximately 1 million miles from the earth. This telescope is the next stepping stone toward understanding the universe and studying the Big Bang theory at NASA.

The near infrared spectrometer (NIRSpec), developed by the European Space Agency (ESA) with major NASA contributions, is the primary instrument on the telescope. It observes thousands of distant galaxies to probe the epoch of initial galaxy formations in the universe. To measure numerous faint objects, the instrument must simultaneously observe a large number of objects in previously unknown positions.

To observe objects at these positions, NASA developed the microshutter array, a 171 by 365 matrix of 100 by 200 µm shutters that can open under random access control. Four microshutter arrays in a 2 by 2 matrix create a programmable transmission mask of about 250,000 shutters so that the NIRSpec can simultaneously target more than 100 faint objects, proportionally improving the efficiency of this major scientific facility. This system is essential to the development of the microshutter array, and it will be critical for the array’s flight qualification in this major international mission.

What Is a Microshutter?

A microshutter is a 100 by 200 µm rectangular door that opens and closes to block light or let it pass through. The shutters pivot on a silicon nitride flexure, actuate magnetically with the help of magnetic coating, and latch electrostatically through electrical connections.

When we began working on this project, manufacturing shutter arrays was a new and complex process that was still under development. NASA manufactures the shutters in arrays with 365 columns and 171 rows for a total of more than 62,000 shutters per array. We mounted the shutters on a substrate and wired the array in a grid so that we can assert its rows and columns to address each shutter. To open a shutter, we passed a magnet across the front of the array while applying high voltage to the row and column of each shutter. The magnetic field opened the shutter, and the static charge at the intersection of the row and column held it open.

We fabricated each shutter array to test some aspect of the overall design. Tests in this facility inform the further definition of the fabrication process. Using the NI PCI-7344 four-axis stepper motor controllers and the NI MID-7604 power motor drivers, we developed the software that controls the vacuum chamber, shutter control instrumentation, cameras, and other apparatuses to evaluate array performance.

Testing with this system revealed that uncontrolled shutter release limits shutter performance. In this uncontrolled approach, one closed a shutter by turning off the power to the row and column of the shutter. With each approach, the shutter impacts its light baffle in a way that significantly limits its lifetime.

The development team decided that we should release the shutters in synchronization with a passing magnet so that the magnetic field cushions the impact as the shutter closes. A test chamber completed in 2005 includes this new synchronized latching-and-release capability.

Microshutter Control System

The microshutters must function reliably for up to 100,000 cycles on different shutter designs. Instead of testing for years, the new test chamber must cycle the shutters rapidly. The motor rotates at up to 240 rpm; thus, the sled, connected to the motor with off-center cables, crosses back and forth in front of the shutter array four times per second. The control system needs to latch or release each of the 365 columns of the shutter array exactly as the magnet passes. To get an idea of the precision and speed required, imagine that each column of the shutter array is a slat 1 in. wide in a picket fence that is 30 ft long. The magnet would be like a jet plane moving past it at more than 700 mph and only 3 ft away.

To control the shutters, we have to communicate with the control electronics and custom high-voltage shift registers. The new system also needs to rapidly communicate and provide utilities to test and verify many operations of the 584 chips. The system must meet all of these control requirements and be fail-safe. The tests run for days at a time, opening and closing all 62,000 shutters 240 times per minute. If the system loses synchronization, the loss can damage the shutters in just a few minutes.

In order to meet these requirements, we had to either design and manufacture a custom chip or use the LabVIEW FPGA Module. We selected a PXI chassis and controller containing a PXI-7813R reconfigurable I/O module and used the LabVIEW FPGA Module to perform shutter control.

The Control Design

The entire system contains a host computer that controls the test chamber, a field-programmable gate array (FPGA) host program that runs on the PXI controller, and FPGA software that runs on the PXI-7813R. With the FPGA host interface, engineers can calibrate the system and perform manual control functions, create and download bitmaps to write to the arrays, and run self-test diagnostics on the other functions of the 584 chips.

The FPGA software reads the position of the magnet from a quadrature encoder or an absolute encoder. We placed the encoder-decoding algorithm in a single-cycle loop running at 40 MHz to ensure it does not miss any steps. After some filtering to remove jitter, we placed the position value in a first-in-first-out memory buffer (FIFO). Another loop on the FPGA reads the FIFO and determines what to do with the shutters based on the current location of the magnet. This state machine communicates with the 584 chips using the protocol to turn the appropriate rows and columns on or off.

If the FIFO overflows, the state machine controlling the shutters is not going fast enough. The software indicates a synchronization error to the host computer so the system can shut down.

This algorithm works very well and has become the foundation for control experimentation on the shutter arrays. As engineers develop new ideas to improve shutter operation, we can easily add or change algorithms in the state machine block.

The LabVIEW FPGA Module and PXI-7813R saved us hundreds of man-hours and thousands of dollars over developing a custom chip. In addition to saving costs, the control algorithm is also inexpensively modified to improve testing, explore shutter issues, and further the development of the NASA microshutter arrays.

Source: www.sine.ni.com

Cisco takes enterprise telephony lead, riding an IP PBX wave

2009-03-12T21:21:00.000-07:00

After years of steady gains, Cisco's IP PBX solutions captured the enterprise telephony market lead over rivals Avaya and Nortel, according to a recent report by Infonetics.

Infonetics found that Cisco grew its IP telephony revenues by 19% in Q3 2008, riding a wave of IP PBX adoption even as traditional TDM equipment sales were projected to drop below the $1 billion mark for the year.

"It would have been surprising if you had looked at the market five years ago, but [Cisco] has been doing well for a few years now, slow and steady, and it's finally paid off, at least in this one quarter," Matthias Machowinski, Infonetics' directing analyst, said of the networking giant's telephony success.

He said that Cisco was unlikely to have the sales lead for the year, however, leaving that title to Avaya or Nortel, which ranked second and third, respectively, in quarterly sales. But in a year or two, barring unforeseen changes, Cisco will probably take the overall lead: It was the only company to make significant gains in the market this last quarter, while its competition either held steady or dropped a few points. After years of steady gains, Cisco's IP PBX solutions captured the enterprise telephony market lead over rivals Avaya and Nortel, according to a recent report by Infonetics.

The enterprise telephony market experienced strong growth of 8% over the previous quarter, but Machowinski predicts that spending will drop and the competition will get fierce as the recession wears on.

"So far, the market has been growing fast enough that even though Cisco is taking share, the other guys haven't been declining," he said. "We do expect the markets to decline, so somebody's going to see their revenues decrease. I wouldn't call the market in disarray, but there's going to be some pressure."

Even then, Cisco will remain in an enviable position, Machowinski said, particularly as the TDM-to-IP PBX transition continues and networking professionals -- the people who know Cisco best -- continue to be given more authority on communications deployments.

"I think Cisco's going to be well positioned," he said, "just because they have such a huge position on the networking side."

Source: www.searchunifiedcommunications.techtarget.com

Make your Voice heard :: All about CCVP

2009-02-09T14:34:00.000-08:00

Unless you've been stuck on Mars for the past couple of years, you're probably already well aware of the fact that most of the buzz words projected out of jubilant salesmen's (and saleswomen's! !) mouths around Cisco canteens invariably have something to do with voice. The CCIE Voice track has almost caught up with the Service Provider track in terms of numbers qualified, despite the latter being available for a much longer period of time. But gaining a Voice certification doesn't always have to mean IE. Have you considered its little sister, namely the CCVP?

Whether you use it as a stepping stone towards attaining your IE or merely as a measure of consolidating your skill set, it certainly has proved to be as popular and valuable to a great number of engineers- just look at the comments received on the various Discussion forums, such as the Cisco Netpro Forum.

The questions one typically asks before embarking on the pursuit of a CCVP go along the lines of… ï¿½ï¿½oeSo is it worth the effort? does it enhance job prospects? what does it entail? how much experience in Cisco's voice technologies do I need? how much Routing and Switching knowledge do I need? etc, etc…"

Whether you like it or not, right now and probably for the next 3-5 years, convergence is hot. Not only inside the US, where technology used for carrying voice over a packetized network is at its most mature, but across the globe. So having a CCVP under your belt is most definitely worth the paper it's printed on and then some. Whether job prospects are enhanced by having a CCVP is a more difficult issue to tackle- it certainly helps getting you to the interview, but ultimately the rest is down to you. One thing in your favor however is that companies pursuing the Cisco ï¿½ï¿½oeUnified Communications Specialization partnership" program require a number of engineers holding a CCVP on their books.

Why should you bother with the CCVP when you can go straight to CCIE Voice? It depends on your experience and certainly there are at least 80-90 CCIE Voice engineers who passed the CCIE Voice lab exam without acquiring the CCVP. These figures are not based upon published statistics but simple mathematics- the IE Voice track was released in September '03 whereas the VP debuted in February '05. However the VP, now that it does exist, is very much worthy as a preparation tool for the IE, as well as a qualification in its own right. It would be impossible to pass the Voice lab without first becoming comfortable with the concepts covered in the VP material so that begs the question- why wouldn't you start with the VP?

As with anything, nothing can substitute real world experience, so every minute working on Call Manager, gateways, QoS, gatekeepers and Catalyst switches, which make up the core components tested on the CCVP exams, helps. The official pre-requisite is having a valid CCNA certification, but don't be fooled- there is virtually no knowledge of routing and switching protocols required to pass the exams, although it does help in tying everything together when looking at the big picture . Whereas the CCNA/CCDA most definitely have tangible benefits when certifying on the CCNP or CCDA, the CCNA is only a pre-requisite for the VP because, er…., there isn't a CCVA! When it comes to teaching the CCVP bootcamp this makes life very difficult because there is absolutely no knowledge of voice required whatsoever.

In total there are five exams that need to be cleared before attaining the VP

* Cisco Voice over IP (CVOICE)
* Cisco IP Telephony (CIPT)
* Implementing Gateways and Gatekeepers (GWGK)
* Quality of Service (QOS)
* IP Telephony and Troubleshooting (IPTT)

Lets start with CVOICE. Earlier releases of this exam dealt with VoFR and VoATM, as well as VoIP. Thankfully Cisco changed their minds and restricted the tested technologies to VoIP. This is the first part of the magic quintet you should deal with and provides a good introduction and grounding for the more glorified exams that follow. In my own view, I would propose that CVOICE should be one half of the CCVA (the other half being the ICND exam that is required for the CCNA). Let's wait and see!

There's two methods of training for this exam- (1) the official Cisco authorized training constituting of the self-study Cisco Voice Fundamentals (CVF) and the 4 day classroom-based CVOICE. (2) A CCVP bootcamp which is not authorized by Cisco but does ï¿½ï¿½oecram" much of the same information.

After CVOICE, the next hurdle is the CIPT exam. This is probably the toughest exam out of the five. It's so difficult that Cisco decided to split the material covered into two separate classes- CIPT1 (5 days in the class-room) and CIPT2 (3 days in the class-room). In doing so they forgot to split the exam in two halves, hence you are presented with one pretty tough exam. How many times have you heard people say ï¿½ï¿½oe…real world experience really does help get you thru this exam…" and end up in the testing center answering a load of questions which do not pertain to any real-world situation? Well, real-world experience really does help you get thru this exam. No, really it does!

Of the five, the CIPT test is probably the most important in its application to the real world. The reason is that the primary focus of the test is Call Manager, which so happens to be the call processing element of any Unified Communications solution. Let's put it another way- Call Manager is the brain of the IP Telephony network. Your Skinny IP Phones, MGCP gateways, Quality of Service, IPCC Express and Unity servers are fairly reliant on Call Manager. In fact, they are totally dependant on Call Manager.

Again, you have 8 days of official Cisco training or material covered in bootcamp format. Whatever means you choose, hands-on lab exercises is vital for this class. There's a whole new dialog that you will come across- call manager groups, regions, device pools, locations, inter-cluster trunks, calling search spaces, partitions, route plans, CDR, BAT, BARS and a whole lot more. And the neat thing is, as well as being useful for the CIPT test, they are the building blocks of engineering any Cisco IPT network in the real world.

Next up is GWGK. Along with CIPT, GWGK provides the real business end of the CCVP cert in that it is very useful in your day to day activities and also for your CCIE prep. Have you read the Cisco Press ï¿½ï¿½oeImplementing Gateway Gatekeeper" book? I'm guessing the answer to this question is an emphatic ï¿½ï¿½oeNo", mainly because there isn't one out yet! That is precisely the big challenge with this test in that it is the newest exam (all the other tests existed before the conception of the CCVP) and there isn't much out there in terms of useful reading literature away from CCO.

You data guys might like this class too! CIPT is heavily focused on the Call Manager Administration GUI- you might get fed up of pointing and clicking in Windows by the end of the CIPT class, so don't say you haven't been warned!. The good news with the GWGK class (5 days with Cisco training or bootcamp) is that it is by and large IOS-based. Hence you don't need to stray too far away from your beloved router. Which, by the way, is now renamed gateway or gatekeeper. That is not to say you're going to spend all your time configuring OSPF, EIGRP, RIP, etc…No, there's a whole new world that exists on a Cisco router that you knew nothing about.

If the data guys didn't like GWGK they are sure to love QoS, the next logical step in the Fibonacci sequence. The reason is that you spend most, if not all, your time configuring and learning about class-maps, policy-maps and the things you can do inside WAN interfaces to prioritize beautiful Cinderella-like voice packets ahead of the ugly cumbersome data packets. Not only that, you can also learn about how you can bring down Serialization delay so voice packets don't need to queue up for very long before being transmitted. And if that wasn't enough, you can finally work out what the ï¿½ï¿½oewrr-queue" command does on a Catalyst switch and what the hell ï¿½ï¿½oe2q2t" queues are.

In all seriousness, the QoS class/test is extremely useful not only for the CCVP certification but for a number of other Cisco Certifications. How applicable is it in the real world? I'm not going to say it isn't at all because it's definitely better to understand all the various commands that show up on configurations, but AutoQos makes life in the real-world an awful lot easier (AutoQos being a macro that implements all the QoS you need with two or three commands).

QoS is covered in a 5 day Cisco class or in any bootcamp. Beware, they also choose to throw in a little BGP which is interesting to say the least. Did you know you can classify packets based on BGBP community lists and AS paths? Well, you can using the QoS Policy Propagation via BGP feature (a.k.a. QPPB). Another interesting point to note on the QoS exam is the number of simlets and simulations that the candidate is required to negotiate. Of all the exams that comprise the CCVP, the QoS exam tests the candidate's ability to apply knowledge gained more than any other test. This makes it, from a personal perspective, the best, and if there is such a thing, the most enjoyable exam.

Once you've passed the CVOICE, CIPT and QOS exams, congratulations are in order. No you haven't got your CCVP yet, but you have now qualified as a ï¿½ï¿½oeCisco IP Communications Support Specialist" which is a nice bonus to pick up along the way to your VP and also useful for partner specialization programs.

The final bridge to cross is the IPTT exam. Troubleshooting always sounds interesting, as if you are on the brink of learning some ground-breaking skill that's been missing your entire life. Well, if that's what you're expecting you might be bitterly disappointed. Troubleshooting methodology is a very difficult skill to learn in a classroom- the best troubleshooters in a particular technology invariably have the most experience troubleshooting that technology. At first glance that may not be such a radical statement to make, however the knock-on affect makes one wonder what the point of the IPTT exam is? And that would be a very good question.

Let me explain why. When sitting the QoS class, a large part of any lab exercise demands that the candidate uses the appropriate ï¿½ï¿½oedebug" and ï¿½ï¿½oeshow" commands in order to fully appreciate and verify the particular QoS technique in operation. When performing lab exercises in the GWGK class, you will also need to be able to perform the appropriate verification. In the CIPT class, CIPT2 focuses largely on the Internal Server Tools that aid troubleshooting Call Manager. In other words, the IPTT class is a culmination of all the other classes with one little exception, namely Unity.

For some reason, probably because the people responsible for writing the IPTT course content decided that all the content was already covered, decided to throw in troubleshooting Unity into the mix. We end up with the interesting prospect of not needing to know anything about designing or supporting Unity for the CCVP certification, but candidates are expected to know how to troubleshoot Unity. Puzzled? You're not the only one.

What we end up with is a mix of all the other tests and some intricate details on Cisco Unity. It's quite an easy exam if the candidate has passed all the other exams in the not-so-distant past and has the Cisco Unity Systems Engineer (CUSE) certification. It's still quite easy if you're comfortable with everything apart from the Unity sections, but the IPTT exam gets quite difficult if you've forgotten the content of the other parts of the CCVP cert.

IPTT should be the last exam any CCVP candidate attempts and training constitutes a 5 day Cisco official class or the final chapter of a CCVP bootcamp.

So what are you waiting for? Go ahead and make your voice heard!

Vik Malhi, CCIE #13890 Voice, CCVP, Cisco IP Telephony Support Specialist, Cisco IP Telephony Operations Specialist, Cisco IP Telephony Design Specialist and Cisco Wireless LAN Design Specialist.

What is NGN

2009-02-08T14:32:00.000-08:00

The Next Generation Network (NGN) is a popular phrase used to describe the network that will replace the current ▲PSTN network around the world today used to carry voice, fax, modem signals, etc.
By definition, the NGN is essentially a managed ▲IP-based (i.e., packet-switched) network that enables a wide variety of services. Among those services are ▲VoIP, ▲videoconferencing, ▲Instant Messaging, e-mail, and all other kinds of pakcet-switched communication services.
The ITU defined the term NGN in Recommendation Y.2001 as follows:
Next Generation Network (NGN): 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.
One of the most important aspects of NGN is the deliberate separation of the access provider from the "service" provider (see the highlighted text above). For those that do not understand what this means, it means that the access provider (the service provider that provides you, the customer, with access to the NGN) may be different than the service provider that provides you with various services, such as voice and video communication, e-mail, stock quotes, or other services.
We say "may", because the access provider and service provider might be the same company. For example, as a subscriber to cable services, you may elect to purchase voice (telephone) services from your cable company. In that case, your access provider and your voice service provider is one in the same. However, the NGN removes this restriction from you—you have a choice. If you prefer to purchase voice services from another company (e.g., Vonage or Lingo ), you have that option, too. Never before have consumers had so many options.
Of course, not everybody is happy with the ability for consumers to have a choice. Why? Because the NGN represents a real threat to the current business model of incumbent service providers. The old-world carriers would prefer to control both the access and the services, blocking competitors from being able to come into the market and offer competitive services.
However, times change and consumers have the right to choose the service providers that provide them services. We have just entered a new era where customers with broadband Internet access can now select their voice service provider of choice—perhaps one that physically exists in an entirely different country! As the incumbent carriers start exploring the possibilities the NGN will bring, they will soon realize an unbounded opportunity for new sources of revenue through a multiplicity of new kinds of services.
As we move forward deploying the Next Generation Network, users may have one or many access providers providing access in a variety of ways, including cable, DSL, ▲Wi-Fi, ▲WiMAX, fiber, etc. into the NGN. Once connected, the options for service providers for voice, video, and data services will be virtually unlimited.
We live in exciting times and are just on the verge of a revolution.

Huawei Wins World' s First 4G/LTE Commercial Contract from TeliaSonera

2009-02-07T14:11:00.000-08:00

Huawei Technologies Co Ltd. ("Huawei"), a leader in providing next generation telecommunications network solutions for operators around the world, today announced that it has been chosen by TeliaSonera, the largest telecoms operator in Scandinavia and the Baltic countries, to supply the world' s first commercial 4G/LTE (Long Term Evolution) network, in Oslo, Norway. Huawei will, together with TeliaSonera, raise mobile broadband speeds significantly by dramatically improving quality and capacity with Huawei' s advanced LTE solution.

Under the agreement, Huawei provides an environmentally friendly end-to-end LTE solution including LTE base stations, core network and OSS (Operating Support System) covering Oslo. Huawei also provides services including network design, implementation, systems integration

The contract marks a significant step in TeliaSonera' s evolution to the next generation network. Thanks to Huawei' s 4^th Generation Base Station platform, the Huawei LTE solution provides an All-IP, high speed, low latency and high frequency efficiency mobile network. LTE technology will deliver new mobile data rates, which will enable TeliaSonera to introduce the fastest mobile broadband experience to their customers.

Mr. Lars Klasson, Senior Vice President and CTO Business Area Mobility Services of TeliaSonera says, "Our customers will enjoy 4G with high-speed and high-quality mobility communications already in 2010. We have chosen Huawei as our partner based on their strong focus on LTE development and early deployment capabilities, as well as Huawei' s impressive and proven worldwide track record in advanced mobile technology."

"We are excited to help TeliaSonera to construct the world' s first commercial LTE network", says Chengdong Yu, President of Huawei European region. "Huawei always focuses on addressing the challenges of operators and the needs of its customers. With our leading LTE technology, we are confident to provide TeliaSonera with winning solutions for their businesses and help them provide the fastest mobile broadband experience for their customers." and support.

LTE Base Station Strategies

2009-02-06T14:07:00.000-08:00

With operators around the world turning their attention to LTE, the equipment suppliers developing the underlying technology and network products are practically busting a lung to make it happen.
The key product in any LTE network will be the radio base station, or eNodeB. Equipment suppliers commercializing LTE radio equipment (and by extension, operators) face a vast number of decisions that hinge on assumptions around deployment scenarios and upgrade strategies, what will create a competitive advantage, and what will deliver an appropriate profit margin over a given timeframe.

Most obviously, there's debate in the market between integrated, multi-standard radio access networks (RANs) and discrete LTE overlays, with the best product portfolios able to support both. Other, more nuanced factors – such as the appropriate balance between software upgradeability and hardware refresh – are also shaping vendors' eNodeB product strategies significantly. For what is ostensibly a highly standardized network element, this is generating some surprisingly different eNodeB designs.
Below, this article provides brief summaries of the LTE base station product strategies of the big four RAN vendors – Ericsson AB (Nasdaq: ERIC - message board), Nokia Siemens Networks , Alcatel-Lucent(NYSE: ALU - message board), and Huawei Technologies Co. Ltd. – featured in the latest Heavy Readingreport "LTE Base Stations & the Evolved Radio Access Network," the first detailed, product-specific, independent market research available off-the-shelf.
In a follow-up piece, I'll provide the same information for Fujitsu Ltd. (Tokyo: 6702 - message board; London: FUJ; OTC: FJTSY), Motorola Inc. (NYSE: MOT - message board), NEC Corp. (Tokyo: 6701 -message board), Nortel Networks Ltd. (NYSE/Toronto: NT - message board), and ZTE Corp. (Shenzhen:000063 - message board; Hong Kong: 0763), all of which have high hopes to revitalize their wireless infrastructure businesses through LTE and win back some market share from the big four.
Ericsson
While the RAN market leader is planning to offer LTE upgrade options for its GSM and 3G base stations, the real action is the new RBS 6000 platform scheduled for availability in mid 2009. Ericsson has bet the farm on the RBS 6000, which, on paper at least, will offer the most complete LTE base station portfolio, ranging from compact LTE overlay products to multi-standard cabinet designs that incorporate GSM and UMTS refresh alongside LTE. Ultimately, the intent is to tightly integrate different generations of RAN technology and drive opex per cell site as low as possible.
Ericsson's approaches to software-defined functions are interesting. A common radio module will be software-configurab le to support GSM, UMTS, and LTE in the same device. But for the baseband, the company will use hardware-specific boards for each technology in a common form factor and cabinet design. Ericsson argues this will deliver superior price/performance over software-defined basebands because it is better able to optimize products and capture the ongoing benefits of improved processor technology. High volume is the key to making this strategy work.
Nokia Siemens
The launch of the second-generation Flexi base station in October 2008 was something of scoop for Nokia Siemens, enabling it to claim bragging rights for shipping the first LTE-capable hardware.
Nokia Siemens has opted for a fully software-defined system that can be configured for LTE or UMTS in both the baseband and radio modules, each of which is housed in Flexi packaging that is field-proven to bring down site costs. The company's recent wins in Canada to provide an HSPA network that can be upgraded in software to LTE (backed up with contractual commitments) demonstrate the advantage of this approach and, crucially, show how having a new-generation base station platform is a competitive advantage.
Such a lead will only last so long, however, and I would expect Nokia Siemens to cycle relatively quickly through a new version of the product, bringing in the latest processors, etc., as available. Notably, GSM is not integrated with 3G and LTE today, as the company argues that GSM is best provided as a highly optimized, standalone product, although that may change in the future.
Alcatel-Lucent
LTE is a great opportunity for Alcatel-Lucent, yet equally, competitors see the CDMA account base as ripe for picking as the world moves to LTE. In this context, the company's decision to ramp investment in LTE independently of its scaled-back cooperation with NEC is clearly a smart move.
Alcatel-Lucent' s product focus is on a new ultra-compact LTE platform intended initially for discrete LTE overlay deployments, but over time its strategy will evolve into more of a multi-standard concept. Conceptually, the company leans toward innovative rackable systems, rather than classic cabinet-based base stations. The 1U baseband module is interesting in that such a small device has a modular design that can be hardware-optimized for particular deployment scenarios or capacity requirements. The baseband device will support a 3G software load in future.
The radio is designed to be software-configurab le to support GSM, UMTS, and LTE. The same radio module can also be used with the current 3G base station, which itself can be upgraded to add LTE support via the addition of a new modem board.
Huawei
Huawei's LTE product focus is on the 3900-series base stations (a.k.a. fourth-generation BTS) unveiled in February 2008, which are already shipping in large volume (several thousands of cabinets already live) into GSM and UMTS networks. This series forms the basis for the company's push into LTE.
The design is very much around the multi-standard base station concept with common, software-defined hardware modules used for GSM, UMTS, and LTE. Huawei's contract to supply the planned Canadian HSPA-to-LTE network speaks to the appeal of this software-upgrade approach and again reinforces the competitive advantage that comes from offering the latest-generation platform against "legacy" 3G products.
Huawei has so far been quiet about its plans to support LTE-compatible multi-standard radio heads (i.e., capable of GSM, UMTS, and LTE), but the company's advanced work on software-defined GSM/UMTS radios is a clear indicator of where it is headed. (See Huawei, VOD Go Soft in the RAN.)
— Gabriel Brown, Senior Analyst, Heavy Reading

Nokia Siemens Networks pilots 4 calls in GSM radio Timeslot

2009-02-05T13:56:00.000-08:00

Nokia Siemens Networks has carried four calls in one GSM radio timeslot for the first time ever on 21 January of this year. The network equipment manufacturer successfully completed a drive test for the Orthogonal Sub Channel (OSC) that doubles the voice capacity of GSM radio network. The OSC innovation is an important addition to Nokia Siemens Networks' offering of sustainable and green products and solutions. With OSC, operators can gain more capacity from the same base station hardware, meaning fewer base station sites are needed in network roll outs and capacity extensions, which in turn saves energy and decreases the CO2 emissions. The OSC demonstration was conducted as a drive test using four handsets sharing only one radio timeslot and without compromising the call quality.

PTCL Broadband on the Go: EVDO Soft Launch

2009-02-03T14:38:00.000-08:00

We have been discussing the competitive landscape of always on wireless broadband in Pakistan. PTCL has entered this lucrative area with a soft launch of personal broadband wireless service using cdma EVDO technology. Service currently available in Karachi, Lahore, Islamabad, Rawalpindi and Mirpur.
For customers, its the performance which matters and PTCL is claiming up to 500 to 1,000 Kbps but in reality average speeds are likely to vary from 300Kbps to 700Kbps for downlink and 200 Kbps to 400Kbps for uplink. Here’s a summary of the rates.

Package 1 (no CPE cost) - Unlimited access:
Monthly service charge: Rs.2500 (first year) and Rs. 2000 from second year onwards.
No initial charges required and the charges will be billed with PTCL landline bill.
Package 2 - Unlimited access:
Initial CPE charge : Rs.4000
Monthly service charges : Rs.2000 for unlimited access
Monthly service charges will be billed with PTCL landline bill.

PTCL provides EvDO PCMCIA cards. You need a laptop with PCMCIA slot. EVDO Rev.A offers up to 3.1 Mbps downlink and up to 1.8Mbps uplink. Average speeds vary from 300Kbps to 700Kbps for downlink and 200 Kbps to 400Kbps for uplink. However it varies depending on the physical situation of the user and the network at a particular time.
Beside the initial coverage in the cities described above, consumers can also use internet in other big cities as well where there is 1900MHz network but on dialup speed (153Kbps). PTCL provides a very large coverage area where you can stay connected.
The broadband speed depends on how far you are from CDMA BTS. Either with EVDO network or CDMA 1x, you will get very good speeds if the tower is within 2 – 3 KMs radius

Mobilink ranked 68th in the Global Market

2009-02-03T13:35:00.000-08:00

Mobilink- the pioneer of GSM networks in Pakistan and currently the market leader with over more than 30 million customers is the only Pakistani operator to be ranked among the top 100 Mobile Operator Brands.
The research survey which ranked Mobilink (PMCL) 68th in the global mobile operator sector and 16th among the Asian Mobile Operators, was carried out by Intangible Business and is published in the December Issue of Mobile Communications International (MCI) Magazine.
In “Brands Punching Above Their Weight” category, Mobilink has been ranked 3 places ahead of Airtel, Tata Indicom and Relience.

"We are truly honored to have been recognized by the international fraternity. This accolade is a reflection of Mobilink’s consistency in driving brand recall and motivation to continue delivering the very best to our consumers and stakeholders. I am thankful to Orascom for their tremendous support and every member of the Mobilink family for their selfless dedication that has made this possible."
Bilal Munir Sheikh
VP Marketing, Mobilink

The Brand Valuation calculations were based on Forecast Sales, Royality Rates and Discount Rates. Each brand was also measured on three years of hard data including turnover, subscriptions, churn, market share, growth, penetration, average revenue per user (ARPU), and profitability

Nortel exits WiMax business

2009-02-03T13:00:00.000-08:00

Nortel Networks Corp, the ailing telecom equipment provider, has decided to abandon its WiMax wireless technology business as it restructures in bankruptcy protection.

In a statement late on Thursday, Toronto-based Nortel said it will also end an agreement with Alvarion Ltd. Under that deal, struck last June, Nortel was to resell Alvarion's WiMax access products and help it fund the development of the technology.

In the months leading up to its filing for bankruptcy protection, Nortel hailed WiMax -- essentially a type of fast, next-generation wireless technology -- as very promising.

Nortel said it will work closely with Alvarion to make sure it can continue to support existing WiMax customers.

"We are taking rapid action to narrow our strategic focus to areas where we can drive maximum return on investment," Richard Lowe, president of Nortel's carrier networks business, said in a statement.

Nortel said last fall that it would look at mitigating the risks related to investments it had made in next-generation technology.

GMPLS Generalized Multiprotocol Label Switching

2009-02-03T11:19:00.000-08:00

GMPLS (Generalized Multiprotocol Label Switching), also known as Multiprotocol Lambda Switching, is a technology that provides enhancements to Multiprotocol Label Switching (MPLS) to support network switching for time, wavelength, and space switching as well as for packet switching in particular, GMPLS provides support for photonic networking , also known as optical communications.
GMPLS enhances MPLS architecture by the complete separation of the control and data planes of various networking layers. It enables a seamless interconnection and convergence of new and legacy networks by allowing end-to-end provisioning, control and traffic engineering even when the start and the end nodes belong to heterogeneous networks.

While the technology used by the GMPLS control plane remains IP-based, the data plane (traffic plane) can now diversify to include more varieties of traffic (TDM, Lambda, packet, and fiber, etc). Generalized MPLS (GMPLS) supports multiple types of switching, i.e., the addition of support for TDM, lambda, and fiber (port) switching. In summary, GMPLS extends MPLS functionality by establishing and provisioning paths for:

TDM paths, where time slots are the labels (SONET).
FDM paths, where electromagnetic frequency is the label (light waves).
Space division multiplexed paths, where the label indicates the physical position of data (Photonic Cross-connect)

Ufone Collect Call

2009-02-01T11:58:00.000-08:00

Ufone has officially come up with this new package (they say its first time in Pakistan ), called Collect Call. With Ufone Collect Call you can remain connected and make calls even when you don’t wish to pay for a call, so you no longer require sending people missed calls as the receiving party pays for the call charges.
With Collect Call you can even make call even with ZERO balance in your account.
This Package is available for both prepaid and postpaid customers, and both (calling/receiving) parties should have Ufone.
At the same time, this package can be used in an efficient manner by managers, who want their subordinates to stay in touch with them – so now they can pay for the calls made to their numbers, while subordinates will have to pay for all other calls. Same can be applied for children-parents scenario.
How to make a FREE Call?
In order to initiate an Ufone Collect Call request, simply dial 11 followed by the desired Ufone number. e.g. 1103335199942.
If the other party accepts your collect call your call will be connected right away. However If the dialed number does not have sufficient balance or cancels your request you will receive a SMS notification.
How to Block Collect Calls?
Don’t worry guys; there is a method available to block all those muft callers. Try following commands…
BR ALL Block all collect call requests.
UN ALL Unblock all collect call requests
BR To block a specific number
UN To unblock a specific number
BR LIST To get the list of Barred numbers
Terms & Conditions

This Service is available to calls from Ufone to Ufone only. A & B party must both be Ufone subscribers.
There are no charges for initiating Collect Call.
Call will be charged to B party as per regular package plan.(If connected)
All Post-Pay & Prepaid users can initiate collect call request.(as A party)
All Post-Pay & Prepaid users can receive collect call request.(as B party)
A subscriber on international roaming will not be able to initiate collect call request.
If B party is on international roaming then collect call request will not be sent to him.
If B party’s outgoing calls are barred (Post-Pay) then A party will be informed accordingly & collect call request will not be initiated.
Postpaid users will not be able to initiate Ufone Collect Call if their outgoing calls are barred.
Prepaid subs will be able to initiate a Collect Call request even if they have zero credit in their account.

Layer 3 Switch

2009-01-31T11:09:00.000-08:00

A Layer 3 switch is a high-performance device for network routing. Layer 3 switches actually differ very little from routers. A Layer 3 switch can support the same routing protocols as network routers do. Both inspect incoming packets and make dynamic routing decisions based on the source and destination addresses inside.

The fundamental difference between a router and a Layer 3 switch is that Layer 3 switches have optimized hardware to pass data as fast as Layer 2 switches, yet they make decisions on how to transmit traffic at Layer 3, just like a router. Within the LAN environment, a Layer 3 switch is usually faster than a router because it is built on switching hardware. In fact, many of Cisco's Layer 3 switches are actually routers that operate faster because they are built on "switching" hardware with customized chips inside the box.

Layer 3 switches often cost less than traditional routers. Designed for use within local networks, a Layer 3 switch will typically not possess the WAN ports and wide area network features a traditional router will always have.Therefore, L3 switches do not completely eliminate the need for routers. Layer 3 switches may still connect to such routers to learn their tables and route packets to them when these packets need to be sent over the WAN. The switches will be very
effective on the workgroup and the backbone within an enterprise, but most likely will not replace the router at the edge of the WAN (read Internet in many cases). Routers perform numerous other functions like filtering with access lists, inter-Autonomous System (AS) routing with protocols such as the Border Gateway Protocol (BGP), and so on. Some Layer 3 switches may completely replace the need for a router if they can provide all these functions.

Mobile WiMax Network Launched in Libya

2009-01-31T10:01:00.000-08:00

Libya Telecom & Technology (LTT), Libya's main and biggest Internet service provider (ISP) has launched a mobile WiMAX network - covering some 300,000 potential subscribers in eighteen cities. The WiMAX service was officially launched by Mohammad Al Gaddafi at the Libya Telecom & Technology head office in Tripoli.
The network infrastructure was supplied by Alcatel-Lucent in a US$56 million contract which was awarded last April.
The WiMAX service covers Tripoli, Benghazi, Sabha, Sirte, Misurata, Khomus, Ijdabia, Al Baidha, Gharyan, Benwaleed, Zuwara, Tarhun, Al Brega, Obari, Zleitin, Zawia, Raslanuf and Ghadames.
LTT was bought over by the General Post and Telecommunication Company (GPTC) in 2004. Muhammad Qaddafi, the eldest son of Libyan president Muammar al-Gaddafi is the Chairman.

FREE Internet for all PTCL landline customers

2008-10-18T07:13:00.000-07:00

Islamabad: Pakistan Telecommunication Company Limited (PTCL) has decided to offer free internet service to all its customers, effective from October 18, 2008.

Dr Sadik Al-Jadir, SEVP Commercial stated," With this perspective, PTCL, the largest IP service provider in Pakistan now brings a revolutionary offer for all its landline subscribers which is unmatched anywhere in the world.

At PTCL Dr. Sadik said "we strive to create value for our customers. We put value creation first and pursue it with a passion to achieve excellence and facilitate our customers".
Effective October 18th, 2008, all our landline subscribers will have dial-up internet services free of charge.

All PTCL landline subscribers can now experience the best dial up speeds with unlimited internet usage during night hours from 10:00pm to 07:00am. Furthermore subscribers can also avail up to 100 hours of free dialup internet on monthly basis from 0:700 am to 10:00pm every day.

In order to access the internet our subscriber can simply dial 131-77777 with "ptcl" as login and password.

From October 18th, 2008, PTCL customers availing unlimited dial up internet package of Rs. 199/month will not be charged anymore. However, customer exceeding 100 hr of free internet during day time will be charged as per existing tariff of PTCL dial up internet at Rs. 6/h.

In addition, PTCL is also shifting customers of value plus (Rs. 75/month for 20 hrs of dial up internet) to basic plus which has no additional charges effective from 1st November 2008.

Dr. Sadik also added, "We at PTCL can do this since we have the best IP network infrastructure and the associated international internet connectivity capable of supporting a very large user base. Hence serving the aspirations of Pakistan to join the leading countries of the world in internet penetration and usage"

We will also be replacing our existing promotional offers with new offers on regular basis to create more value for customers.
In this connection, this Free Internet offer has replaced our limited time (3 months) free local calls promotion allowing free local calls between 11pm to 6am which has ended in October.

The Half Rate Sundays' promotion is another value addition to the PTCL landline which offers half rates on all local, nationwide and mobile calls made on Sundays. This promotion is effective from October 5, 2008.

Dr. Sadik stated that 'PTCL continues to strive to serve our customers better by creating value and shall soon be marketing more innovative and attractive products and offers in the future'

Alcatel-Lucent | Planning and Deploying the 3G/UMTS Network

2008-10-03T12:35:00.000-07:00

The telecommunications marketplace is changing rapidly, placing increasing emphasis on providing consumers with personalized lifestyle services and mobility. There has been an explosion in the availability of sophisticated user devices that support integrated voice, data, and video applications. According to a UMTS Forum white paper, at the end of 2006 consumers had a choice of more than 400 WCDMA terminal designs worldwide representing handsets optimized for voice, video, and other multimedia services.1 UMTS also estimates that worldwide 3G subscribers will exceed 275 million by the end of 2007. By the end of the decade, the total of global 3G WCDMA subscribers should approach 800 million.

Driving consumer acceptance and increased demand are new features such as high resolutions screens, multi-megapixel cameras with quality optics, fast USB and WiFI connectivity, and large amounts of removable storage capacity such as compact flash, memory sticks, and other formats that can store gigabytes of music, pictures, and videos. The business market is equally active with handsets equipped with business-oriented operating systems, keyboards for text entry, and PDA handhelds that are moving into the territory that was once exclusively held by notebook PCs. 3G/UMTS will make possible a whole new range of mobile data applications such as telemedicine, electronic banking, and location-based services.
Mobile TV, video, and music are three hot new services. Better phones with high resolution screens featuring large color palettes, combined with improved power management, are helping drive the demand for mobile TV, a premiere applications based on 3G/UMTS-based services. Mobile video is another killer application — witness the number of video clips created on mobile devices that are posted daily to YouTube. And mobile music — downloading songs to your mobile device — although currently accounting for only a small percentage of service provider ARPU, is growing rapidly.
All of this activity presents service providers with great opportunities — and some formidable challenges as well.
Alcatel-Lucent studies indicate that customers are increasingly looking to bundle their various communications services and have no qualms about switching if their provider is unable to offer a whole range of new 3G services backed up by high levels of quality of service (QoS). This means service providers have to present their subscribers with a top notch service experience and QoS for content-based services, while, at the same time, re-engineering their operations to significantly increase service velocity. To achieve this goal they must be able to cost-effectively manage the migration to 3G/UMTS.

Planning and Deploying the 3G/UMTS Network
In order to take advantage of the rapidly growing market for new converged mobile services, operators need to not only deploy UMTS, but to make sure that the 3G service is integrated with its existing 2G GSM network. In working with its service provider customers, Alcatel-Lucent has employed a variety of techniques to bring their UMTS clusters (a group of 20~25 contiguous NodeBs) to acceptable performance status within a significantly compressed time line. Key areas addressed by he deployment teams include:
• RF Design;
• RNC planning and configuration;
• Initial neighbor list – UMTS;
• Initial neighbor list – IRAT; and
• Drive route and cluster definition and 3rd party audit of antenna installation

RF Design

Fundamental to a successful UMTS deployment is an excellent design that has as its foundation high quality geographic data and accurate signal propagation models. Developing the right propagation models is essential — failure to do so can compromise the entire deployment. 3G networks demand rigorous precision;
A hefty percentage of the initial design time — in some cases, a month or longer — can be expended in identifying the right sites in the target market area. This essential preliminary work includes gathering, sorting, and tuning data in order to simulate actual market conditions in the target area and develop accurate propagation models assigned to the appropriate cells or sectors in both the 3G and 2G spectrums. Continuous wave (CW) tests performed over both spectrums (for example, at 1900 and 850 MHz) ensure the validity of the UMTS and GSM propagation models.

Understanding WiMAX and 3G for Portable/Mobile Broadband Wireless

2008-10-03T12:29:00.000-07:00

Introduction

Although the broadband data market segment has been
rather anemic for the past couple decades, declining
average revenue per user has caused carriers to look at
wireless broadband data as a means to drive revenue
growth. While growth of low-bandwidth applications
such as downloading ring tones and SMS are
experiencing sharp growth, the growth of broadband
data applications such as email and downloading/
uploading files with a laptop computer or PDA has been
slow. Primary inhibitors of portable broadband services
have included service price, slow data speed and spotty
coverage.
Early Wide Area Network Technologies (WAN) such as
General Packet Radio Service (GPRS) offered average
throughput speeds of 10 Kbps, which was far too slow
for user satisfaction. In 2003, carriers began deploying
services such as Enhanced Data rates for Global
Evolution (EDGE), which delivers average speeds of
100-130 Kbps and bursty traffic up to 200 Kbps. Code
Division Multiple Access (CDMA) technologies such as
1xEvDO provide average speeds of ~300–400 Kbps
with bursts up to 700 Kbps; EVDV boosts these speeds
even higher.[5]
Recent research from In-Stat/MDR* (4/04)[5] indicates
that laptop computers are becoming the access devices
of choice for broadband wireless data. Personal
productivity applications such as email, address books,
calendars, and internet browsers, are among the top
applications used.
While many service providers and operators may be
somewhat familiar with the previously mentioned 2.5G
services, they are now hearing about newer 3G
technologies such as UMTS and HSDPA, and other
technologies such as WiMAX (IEEE 802.16e), which
offer substantial improvements in data rate and spectral
efficiency. This paper focuses on the technical
differences between these technologies by comparing
the differences between the modulation techniques
used in CDMA and OFDMA.

Technology Overview

WCDMA

Wideband Code Division Multiple Access uses Direct
Sequence Spread Spectrum (DSSS) to spread the
signal over a 5 MHz spectrum. It is based on 3GPP
Release 99 and provides data rates of 384 Kbps for
wide area coverage and up to 2 Mbps for hot-spot
areas. In addition to the use of orthogonal spreading
codes, it uses Quadrature Phase Shift Keying (QPSK)
for its modulation.

High Speed Downlink Packet Access
(HSDPA) Overview

WCDMA 3GPP Release 5 extends the WCDMA
specification with High Speed Downlink Packet Access
(HSDPA). HSDPA adds a new transport channel, the
high-speed downlink shared channel (HS-DSCH),
which is optimized for shared data. It also provides
higher-order modulation (Quadrature Amplitude
Modulation or QAM), short transmission time interval
(TTI), fast link adaptation, fast scheduling, and fast
hybrid automatic-repeat-request (ARQ).

WiMAX (IEEE 802.16e)

The portable version of WiMAX, IEEE 802.16e utilizes
Orthogonal Frequency Division Multiplexing Access
(OFDM/OFDMA) where the spectrum is divided into
many sub-carriers. Each sub-carrier then uses QPSK or
QAM for modulation. For more on the basics of OFDM,
refer to Orthogonal Frequency Division Multiplexing.

IP Telephony

2008-10-03T12:17:00.000-07:00

What is IP Telephony?
The public Internet, of course, is the best-known example of an IP network. In IP networks, information is digitized and transmitted as a stream of packets over a digital data network. IP networks allow each packet to independently find the most efficient path to the intended destination, thereby best using the network resources at any given instant. The packets associated with a single source may thus take many different paths to the destination in traversing the network. At the destination, however, the packets are re-assembled and converted back into the original signal. Cost efficiencies arise because of the superior utilization of the network as it shares traffic among many sources.
IP telephony is the collection of technologies that emulates and extends today's circuit-switched
telecommunications services to operate on packet-switched data networks based on the Internet Protocol (IP). Defined in this way, IP telephony encompasses these technologies and extends those capabilities even further to include new telecommunications applications made possible by the convergence of voice and data.
The data carried by an IP network can be as simple as transactional queries and responses or as
complex as broadband multimedia services. In particular, the technology of IP telephony supports all the functions of voice communications, fax communications, routing, authorization, authentication,
accounting, billing, and network management that are now provided by the PSTN. In addition, IP
Telephony will allow the PSTN to function directly with other networks and products. With set standards, Internet Telephony supplies the framework for the integration of computer and voice networks to enable a range of new services, including Virtual Private Networks, Unified Messaging, and Web-enabled Calls. IP Telephony is flexible enough to provide unlimited new services, allowing it to reach into every corner of the globe.
The IP Telephony Network it is widely accepted and acknowledged by the communications industry and industry analysts as a whole, that the Internet Protocol (IP) will become the universal transport of the future. The rapid adoption and migration of both service providers and vendors to the utilization of IP as a transport for data, voice, and video applications further endorses this transition to a converged networking paradigm. This includes those networks that have historically used time-division multiplexing (TDM) infrastructures and relied upon "old world" practices.
The IP Telephony networks seen today are still expanding and evolving. In many ways, any large-scale network today still has some dependency on the PSTN. The PSTN network may be used as an access or termination point to the IP Telephony network, the intelligence that is stored in the global SS7 networks may be used to help with call processing and routing within the IP Telephony network, or the PSTN network may still provide some services currently not available in the products of today. But the products that are now starting to emerge offer not only the features of the PSTN network with greater capability and efficiency, but brand new ideas that will bring new heights to the telecommunications of the world.

Gateway is used to originate and terminate calls with the PSTN, controlled by the Call Agent. The Call Agent also has the capability to connect to a Signaling Transport Point (STP) on the PSTN network, terminating the SS7 signaling. When interfacing with SS7, the Call Agent uses the Media Gateway to interface with the SS7 voice paths.
The Gatekeeper is used to manage all routing and call control within the node. All IP call signaling must travel through the Gatekeeper so that routing and network configuration stays centrally managed.
The IVR and Call Accounting server are used to provide back-office support to the node. By using the
IVR to prompt callers for relevant information, the network provides a quick and efficient interface for the caller to enter an account number, the destination number, etc. The Call Accounting server then stores each call transaction in a database, as well as performing balance tracking, account validation, call tracking, pre/post-paid calling services, and detailed records for all calls within the network.
With the idea of an IP Telephony Network replacing a PSTN network, it becomes paramount for Networks to be able to interoperate, including be able to exchange traffic, information, and “roaming” services.
Each network must also maintain security and protection against loss of service. Using Encryption and
using private security keys, it becomes possible for each endpoint to know the exact identity of any other endpoint.

QAM: Quadrature Amplitude Modulation

2008-06-10T00:53:00.001-07:00

Quadrature amplitude modulation (QAM) is a modulation scheme in which two sinusoidal carriers, one exactly 90 degrees out of phase with respect to the other, are used to transmit data over a given physical channel. Because the orthogonal carriers occupy the same frequency band and differ by a 90 degree phase shift, each can be modulated independently, transmitted over the same frequency band, and separated by demodulation at the receiver. For a given available bandwidth, QAM enables data transmission at twice the rate of standard pulse amplitude modulation (PAM) without any degradation in the bit error rate (BER). QAM and its derivatives are used in both mobile radio and satellite communication systems.
You can use the numerically controlled oscillator (NCO) Compiler to design a dual-output oscillator that accurately generates the in-phase and quadrature carriers used by a QAM modulator. The carrier frequency of each sinusoid can be set to any precision by defining the phase increment input to the NCO. A block diagram of the QAM modulator is shown Figure 1. A raised cosine finite impulse response (FIR) filter is used to filter the data streams before modulation onto the quadrature carriers. When passed through a band-limited channel, rectangular pulses suffer from the effects of time dispersion and tend to smear into one another. This pulse shaping filter eliminates inter-symbol interference by ensuring that at a given sampling instance, the contribution to the response from all other symbols is zero.
Figure 1. QAM Modulator Block Diagram
Design Files Provided
The QAM reference design files are located in the \refdesigns\qam\* folder. Table 1 describes the files provided with this reference design.
To download the reference design files, go to the NCO product page and click on the free test drive icon.
Table 1. QAM Reference Design Files
File
Description
qam.gdf
Top-level design file, which can be compiled and simulated in the MAX+PLUS® II or QuartusTM II software.
qam.v
Top-level Verilog HDL model.
tbqam.v
Verilog HDL testbench, which writes the I and Q outputs and the QAM signal to text files.
qam.mdl
Simulink model of the QAM design.
Functional Description
16-QAM is achieved by modulating two 4-level PAM signals onto orthogonal carriers. The FIR filters in this design are 23-tap raised-cosine filters, and were designed using Altera’s FIR Compiler. The NCO generates two 1-MHz orthogonal carriers and uses a dual-output, ROM-based architecture with the following parameters:
Phase accumulator width: 24
ROM address width: 10
Magnitude precision: 8
A sample of the output from the testbench is shown in Figure 2. The in-phase and quadrature sequences have been modulated onto the orthogonal carriers to occupy a single channel centered at 1 MHz.
Figure 2. QAM Testbench Output
You can implement continuous-amplitude modulation schemes such as quadrature phase shift keying (QPSK), offset QPSK, and differential phase shift keying (DPSK) by making minor modifications to this reference design. Phase shift keying modulation schemes modify the phase of the carrier to transmit data through the channel.
QPSK is used when the phase transitions of 180 degrees lead to zero crossings in a pulse-shaped QPSK signal causing undesirable sidelobes in the spectrum of the modulated signal. You can implement the offset can by incorporating a delay of half the symbol interval between the two quadrature datapaths and then modulating the carriers as before. Similarly, DPSK requires additional logic at the front end of the QAM modulator to generate the encoded data sequences to the I and Q inputs of the modulator

Orthogonal Frequency - Division Multiplexing

2008-06-10T00:51:00.002-07:00

Orthogonal frequency-division multiplexing (OFDM) — essentially identical to Coded OFDM (COFDM) and Discrete multi-tone modulation (DMT) — is a frequency-division multiplexing (FDM) scheme utilized as a digital multi-carrier modulation method. A large number of closely-spaced orthogonal sub-carriers are used to carry data. The data are divided into several parallel data streams or channels, one for each sub-carrier. Each sub-carrier is modulated with a conventional modulation scheme (such as quadrature amplitude modulation or phase shift keying) at a low symbol rate, maintaining total data rates similar to conventional single-carrier modulation schemes in the same bandwidth.
OFDM has developed into a popular scheme for wideband digital communication, whether wireless or over copper wires, used in applications such as digital television and audio broadcasting, wireless networking and broadband internet access.
The primary advantage of OFDM over single-carrier schemes is its ability to cope with severe channel conditions — for example, attenuation of high frequencies in a long copper wire, narrowband interference and frequency-selective fading due to multipath — without complex equalization filters. Channel equalization is simplified because OFDM may be viewed as using many slowly-modulated narrowband signals rather than one rapidly-modulated wideband signal. The low symbol rate makes the use of a guard interval between symbols affordable, making it possible to handle time-spreading and eliminate intersymbol interference (ISI). This mechanism also facilitates the design of single-frequency networks, where several adjacent transmitters send the same signal simultaneously at the same frequency, as the signals from multiple distant transmitters may be combined constructively, rather than interfering as would typically occur in a traditional single-carrier system.
Contents[hide]
1 Example of applications
1.1 Cable
1.2 Wireless
2 Key features
2.1 Summary of advantages
2.2 Summary of disadvantages
3 Characteristics and principles of operation
3.1 Orthogonality
3.2 Guard interval for elimination of inter-symbol interference
3.3 Simplified equalization
3.4 Channel coding and interleaving
3.5 Adaptive transmission
3.6 OFDM extended with multiple access
3.7 Space diversity
3.8 Linear transmitter power amplifier
4 Idealized system model
4.1 Transmitter
4.2 Receiver
5 Mathematical description
6 Usage
6.1 OFDM system comparison table
6.2 ADSL
6.3 Powerline Technology
6.4 Wireless local area networks (LAN) and metropolitan area networks (MAN)
6.5 Wireless personal area networks (PAN)
6.6 Terrestrial digital radio and television broadcasting
6.6.1 DVB-T
6.6.2 COFDM vs. VSB
6.6.3 Digital radio
6.6.4 BST-OFDM used in ISDB
6.7 Ultra-wideband
6.8 Flash-OFDM
7 History
8 See also
9 References
10 Notes
11 External links
//

[edit] Example of applications
The following list is a summary of existing OFDM based standards and products. For further details, see the Usage section in the end of the article.

[edit] Cable
ADSL and VDSL broadband access via POTS copper wiring.
Power line communication (PLC).
Multimedia over Coax Alliance (MoCA) home networking.

[edit] Wireless
The wireless LAN radio interfaces IEEE 802.11a, g, n and HIPERLAN/2.
The digital radio systems DAB/EUREKA 147, DAB+, Digital Radio Mondiale, HD Radio, T-DMB and ISDB-TSB.
The terrestrial digital TV system DVB-T.
The terrestrial mobile TV systems DVB-H, T-DMB, ISDB-T and MediaFLO forward link.
The cellular communication systems Flash-OFDM
The mobile broadband 3GPP Long Term Evolution air interface named High Speed OFDM Packet Access (HSOPA)
The Wireless MAN / Fixed broadband wireless access (BWA) standard IEEE 802.16 (or WiMAX).
The Mobile Broadband Wireless Access (MBWA) standards IEEE 802.20, IEEE 802.16e (Mobile WiMAX) and WiBro.
The wireless Personal Area Network (PAN) Ultra wideband (UWB) IEEE 802.15.3a implementation suggested by WiMedia Alliance.

[edit] Key features
The advantages and disadvantages listed below are further discussed in the Characteristics and principles of operation section.

[edit] Summary of advantages
Can easily adapt to severe channel conditions without complex equalization
Robust against narrow-band co-channel interference
Robust against Intersymbol interference (ISI) and fading caused by multipath propagation
High spectral efficiency
Efficient implementation using FFT
Low sensitivity to time synchronization errors
Tuned sub-channel receiver filters are not required (unlike conventional FDM)
Facilitates Single Frequency Networks, i.e. transmitter macrodiversity.

[edit] Summary of disadvantages
Sensitive to Doppler shift.
Sensitive to frequency synchronization problems.
High peak-to-average-power ratio (PAPR), requiring linear transmitter circuitry, which suffers from poor power efficiency.

[edit] Characteristics and principles of operation

[edit] Orthogonality
In OFDM, the sub-carrier frequencies are chosen so that the sub-carriers are orthogonal to each other, meaning that cross-talk between the sub-channels is eliminated and inter-carrier guard bands are not required. This greatly simplifies the design of both the transmitter and the receiver; unlike conventional FDM, a separate filter for each sub-channel is not required.
The orthogonality also allows high spectral efficiency, near the Nyquist rate. Almost the whole available frequency band can be utilized. OFDM generally has a nearly 'white' spectrum, giving it benign electromagnetic interference properties with respect to other co-channel users.
The orthogonality allows for efficient modulator and demodulator implementation using the FFT algorithm. Although the principles and some of the benefits have been known since the 1960s, OFDM is popular for wideband communications today by way of low-cost digital signal processing components that can efficiently calculate the FFT.
OFDM requires very accurate frequency synchronization between the receiver and the transmitter; with frequency deviation the sub-carriers will no longer be orthogonal, causing inter-carrier interference (ICI), i.e. cross-talk between the sub-carriers. Frequency offsets are typically caused by mismatched transmitter and receiver oscillators, or by Doppler shift due to movement. Whilst Doppler shift alone may be compensated for by the receiver, the situation is worsened when combined with multipath, as reflections will appear at various frequency offsets, which is much harder to correct. This effect typically worsens as speed increases, and is an important factor limiting the use of OFDM in high-speed vehicles. Several techniques for ICI suppression are suggested, but they may increase the receiver complexity.

[edit] Guard interval for elimination of inter-symbol interference
One key principle of OFDM is that since low symbol rate modulation schemes (i.e. where the symbols are relatively long compared to the channel time characteristics) suffer less from intersymbol interference caused by multipath, it is advantageous to transmit a number of low-rate streams in parallel instead of a single high-rate stream. Since the duration of each symbol is long, it is feasible to insert a guard interval between the OFDM symbols, thus eliminating the intersymbol interference.
The guard interval also eliminates the need for a pulse-shaping filter, and it reduces the sensitivity to time synchronization problems.
A simple example: If one sends a million symbols per second using conventional single-carrier modulation over a wireless channel, then the duration of each symbol would be one microsecond or less. This imposes severe constraints on synchronization and necessitates the removal of multipath interference. If the same million symbols per second are spread among one thousand sub-channels, the duration of each symbol can be longer by a factor of thousand, i.e. one millisecond, for orthogonality with approximately the same bandwidth. Assume that a guard interval of 1/8 of the symbol length is inserted between each symbol. Intersymbol interference can be avoided if the multipath time-spreading (the time between the reception of the first and the last echo) is shorter than the guard interval, i.e. 125 microseconds. This corresponds to a maximum difference of 37.5 kilometers between the lengths of the paths.
The cyclic prefix, which is transmitted during the guard interval, consists of the end of the OFDM symbol copied into the guard interval, and the guard interval is transmitted followed by the OFDM symbol. The reason that the guard interval consists of a copy of the end of the OFDM symbol is so that the receiver will integrate over an integer number of sinusoid cycles for each of the multipaths when it performs OFDM demodulation with the FFT.

[edit] Simplified equalization
The effects of frequency-selective channel conditions, for example fading caused by multipath propagation, can be considered as constant (flat) over an OFDM sub-channel if the sub-channel is sufficiently narrow-banded, i.e. if the number of sub-channels is sufficiently large. This makes equalization far simpler at the receiver in OFDM in comparison to conventional single-carrier modulation. The equalizer only has to multiply each sub-carrier by a constant value, or a rarely changed value.
Our example: The OFDM equalization in the above numerical example would require N = 1000 complex multiplications per OFDM symbol, i.e. one million multiplications per second, at the receiver. The FFT algorithm requires Nlog2N = 10000 complex-valued multiplications per OFDM symbol, i.e. 10 million multiplications per second, at both the receiver and transmitter side. This should be compared with the corresponding one million symbols/second single-carrier modulation case mentioned in the example, where the equalization of 125 microseconds time-spreading using a FIR filter would require 125 multiplications per symbol, i.e. 125 million multiplications per second.
Some of the sub-carriers in some of the OFDM symbols may carry pilot signals for measurement of the channel conditions, i.e. the equalizer gain for each sub-carrier. Pilot signals may also be used for synchronization.
If differential modulation such as DPSK is applied to each sub-carrier, equalization can be completely omitted, since these schemes are insensitive to slowly changing amplitude and phase distortion.

[edit] Channel coding and interleaving
OFDM is invariably used in conjunction with channel coding (forward error correction), and almost always uses frequency and/or time interleaving.
Frequency (subcarrier) interleaving increases resistance to frequency-selective channel conditions such as fading. For example, when a part of the channel bandwidth is faded, frequency interleaving ensures that the bit errors that would result from those subcarriers in the faded part of the bandwidth are spread out in the bit-stream rather than being concentrated. Similarly, time interleaving ensures that bits that are originally close together in the bit-stream are transmitted far apart in time, thus mitigating against severe fading as would happen when travelling at high speed.
However, time interleaving is of little benefit in slowly fading channels, such as for stationary reception, and frequency interleaving offers little to no benefit for narrowband channels that suffer from flat-fading (where the whole channel bandwidth is faded at the same time).
The reason why interleaving is used on OFDM is to attempt to spread the errors out in the bit-stream that is presented to the error correction decoder, because when such decoders are presented with a high concentration of errors the decoder is unable to correct all the bit errors, and a burst of uncorrected errors occurs.
A common type of error correction coding used with OFDM-based systems is convolutional coding, which is often concatenated with Reed-Solomon coding. Convolutional coding is used as the inner code and Reed-Solomon coding is used for the outer code — usually with additional interleaving (on top of the time and frequency interleaving mentioned above) in between the two layers of coding. The reason why this combination of error correction coding is used is that the Viterbi decoder used for convolutional decoding produces short errors bursts when there is a high concentration of errors, and Reed-Solomon codes are inherently well-suited to correcting bursts of errors.
Newer systems, however, usually now adopt the near-optimal types of error correction coding that use the turbo decoding principle, where the decoder iterates towards the desired solution. Examples of such error correction coding types include turbo codes and LDPC codes. These codes only perform close to the Shannon limit for the Additive White Gaussian Noise (AWGN) channel, however, and some systems that have adopted these codes have concatenated them with either Reed-Solomon (for example on the MediaFLO system) or BCH codes (on the DVB-S2 system) to improve performance further over the wireless channel.

[edit] Adaptive transmission
The resilience to severe channel conditions can be further enhanced if information about the channel is sent over a return-channel. Based on this feedback information, adaptive modulation, channel coding and power allocation may be applied across all sub-carriers, or individually to each sub-carrier. In the latter case, if a particular range of frequencies suffers from interference or attenuation, the carriers within that range can be disabled or made to run slower by applying more robust modulation or error coding to those sub-carriers.
The term discrete multitone modulation (DMT) denotes OFDM based communication systems that adapt the transmission to the channel conditions individually for each sub-carrier, by means of so called bit-loading. Examples are ADSL and VDSL.
The upstream and downstream speeds can be varied by allocating either more or fewer carriers for each purpose. Some forms of Rate-adaptive DSL use this feature in real time, so that the bitrate is adopted to the co-channel interference and bandwidth is allocated to whichever subscriber that needs it most.

[edit] OFDM extended with multiple access
OFDM in its primary form is considered as a digital modulation technique, and not a multi-user channel access technique, since it is utilized for transferring one bit stream over one communication channel using one sequence of OFDM symbols. However, OFDM can be combined with multiple access using time, frequency or coding separation of the users.
In Orthogonal Frequency Division Multiple Access (OFDMA), frequency-division multiple access is achieved by assigning different OFDM sub-channels to different users. OFDMA supports differentiated quality-of-service by assigning different number of sub-carriers to different users in a similar fashion as in CDMA, and thus complex packet scheduling or media access control schemes can be avoided. OFDMA is used in the IEEE 802.16 Wireless MAN standard, commonly referred to as WiMAX.
In Multi-carrier code division multiple access (MC-CDMA), also known as OFDM-CDMA, OFDM is combined with CDMA spread spectrum communication for coding separation of the users. Co-channel interference can be mitigated against, meaning that manual fixed channel allocation (FCA) frequency planning is simplified, or complex dynamic channel allocation (DCA) schemes are avoided.

[edit] Space diversity
In OFDM based wide area broadcasting, receivers can benefit from receiving signals from several spatially-dispersed transmitters simultaneously, since transmitters will only destructively interfere with each other on a limited number of sub-carriers, whereas in general they will actually reinforce coverage over a wide area. This is very beneficial in many countries, as it permits the operation of national single-frequency networks (SFNs), where many transmitters send the same signal simultaneously over the same channel frequency. SFNs utilise the available spectrum more effectively than conventional multi-frequency broadcast networks (MFN), where program content is replicated on different carrier frequencies. SFNs also result in a diversity gain in receivers situated midway between the transmitters. The coverage area is increased and the outage probability decreased in comparison to an MFN, due to increased received signal strength averaged over all sub-carriers.
Although the guard interval only contains redundant data, which means that it reduces the capacity, some OFDM-based systems, such as some of the broadcasting systems, deliberately use a long guard interval in order to allow the transmitters to be spaced farther apart in an SFN, and longer guard intervals allow larger SFN cell-sizes. A rule of thumb for the maximum distance between transmitters in an SFN is equal to the distance a signal travels during the guard interval — for instance, a guard interval of 200 microseconds would allow transmitters to be spaced 60 km apart.
Single-frequency networks is a form of transmitter macrodiversity. The concept can be further utilized in Dynamic single-frequency networks (DSFN), where the SFN grouping is changed from timeslot to timeslot.
OFDM may be combined with other forms of space diversity, for example antenna arrays and MIMO channels. This is done in the IEEE802.11n Wireless LAN standard.

[edit] Linear transmitter power amplifier
An OFDM signal exhibits a high peak-to-average power ratio (PAPR) because the independent phases of the sub-carriers mean that they will often combine constructively. Handling this high PAPR requires:
a high-resolution digital-to-analog converter (DAC) in the transmitter
a high-resolution analog-to-digital converter (ADC) in the receiver
a linear signal chain.
Any non-linearity in the signal chain will cause intermodulation distortion that
raises the noise floor
may cause intersymbol interference
generates out-of-band spurious radiation.
The linearity requirement is demanding, especially for transmitter RF output circuitry where amplifiers are often designed to be non-linear in order to minimise power consumption. In practical OFDM systems a small amount of peak clipping is allowed to limit the PAPR in a judicious tradeoff against the above consequences. However, the transmitter output filter which is required to reduce out-of-band spurrii to legal levels has the effect of restoring peak levels that were clipped, so clipping is not an effective way to reduce PAPR.
Although the spectral efficiency of OFDM is attractive for both terrestrial and space communications, the high PAPR requirements have so far limited OFDM applications to terrestrial systems.

[edit] Idealized system model
This section describes a simple idealized OFDM system model suitable for a time-invariant AWGN channel.

[edit] Transmitter
An OFDM carrier signal is the sum of a number of orthogonal sub-carriers, with baseband data on each sub-carrier being independently modulated commonly using some type of quadrature amplitude modulation (QAM) or phase-shift keying (PSK). This composite baseband signal is typically used to modulate a main RF carrier.

s[n] is a serial stream of binary digits. By inverse multiplexing, these are first demultiplexed into N parallel streams, and each one mapped to a (possibly complex) symbol stream using some modulation constellation (QAM, PSK, etc.). Note that the constellations may be different, so some streams may carry a higher bit-rate than others.
An inverse FFT is computed on each set of symbols, giving a set of complex time-domain samples. These samples are then quadrature-mixed to passband in the standard way. The real and imaginary components are first converted to the analogue domain using digital-to-analogue converters (DACs); the analogue signals are then used to modulate cosine and sine waves at the carrier frequency, fc, respectively. These signals are then summed to give the transmission signal, s(t).

[edit] Receiver

The receiver picks up the signal r(t), which is then quadrature-mixed down to baseband using cosine and sine waves at the carrier frequency. This also creates signals centered on 2fc, so low-pass filters are used to reject these. The baseband signals are then sampled and digitised using analogue-to-digital converters (ADCs), and a forward FFT is used to convert back to the frequency domain.
This returns N parallel streams, each of which is converted to a binary stream using an appropriate symbol detector. These streams are then re-combined into a serial stream, , which is an estimate of the original binary stream at the transmitter.

[edit] Mathematical description
If N sub-carriers are used, and each sub-carrier is modulated using M alternative symbols, the OFDM symbol alphabet consists of MN combined symbols.
The low-pass equivalent OFDM signal is expressed as:
where {Xk} are the data symbols, N is the number of sub-carriers, and T is the OFDM symbol time. The sub-carrier spacing of 1 / T makes them orthogonal over each symbol period; this property is expressed as:
where denotes the complex conjugate operator and is the Kronecker delta..
To avoid intersymbol interference in multipath fading channels, a guard interval of length Tg is inserted prior to the OFDM block. During this interval, a cyclic prefix is transmitted such that the signal in the interval equals the signal in the interval . The OFDM signal with cyclic prefix is thus:
The low-pass signal above can be either real or complex-valued. Real-valued low-pass equivalent signals are typically transmitted at baseband—wireline applications such as DSL use this approach. For wireless applications, the low-pass signal is typically complex-valued; in which case, the transmitted signal is up-converted to a carrier frequency fc. In general, the transmitted signal can be represented as:

[edit] Usage

[edit] OFDM system comparison table
Key features of some common OFDM based systems are presented in the following table.
Standard name
DAB Eureka 147
DVB-T
DVB-H
DMB-T/H
IEEE 802.11a
Ratified year
1995
1997
2004
2006
1999
Frequency range oftoday's equipment(MHz)
174 - 2401452 - 1492
470 - 862
470 - 862
470 – 862
4915 - 5825
Channel spacing B(MHz)
1.712
8
8
8
20
Number of subcarriers N
192, 384, 768 or 1536
2K mode: 17058K mode: 6817
1705, 3409, 6817
1 (single-carrier)3780 (multi-carrier)
52
Subcarrier modulation scheme
DQPSK
QPSK, 16QAM or 64QAM
QPSK, 16QAM or 64QAM
4QAM, 4QAM-NR,[1]16QAM, 32QAM and 64QAM.
BPSK, QPSK, 16QAM or 64QAM
Useful symbol length TU(μs)
2K mode: 2248K mode: 896
224, 448, 896
500 (multi-carrier)
3.2
Additional guard interval TG(Fraction of TU)
1/4, 1/8, 1/16, 1/32
1/4, 1/8, 1/16, 1/32
1/4, 1/6, 1/9
1/4
Subcarrier spacingΔf = 1/(TU) ≈ B/N(Hz)
2K mode: 44648K mode: 1116
4464, 2232, 1116
8 M (single-carrier)2000 (multi-carrier)
312.5K
Net bit rate R(Mbit/s)
0.576 - 1.152
4.98 - 31.67(typically 24)
3.7 - 23.8
4.81 - 32.49
6 - 54
Link spectral efficiency R/B(bit/s/Hz)
0.34 - 0.67
0.62 - 4.0
0.62 - 4.0
0.60 - 4.1
0.30 - 2.7
Inner FEC
Conv coding with code rates
1/4, 3/8 or 1/2
Conv coding with code rates
1/2, 2/3, 3/4, 5/6 or 7/8
Conv coding with code rates
1/2, 2/3, 3/4, 5/6 or 7/8
LDPC with code rates
0.4, 0.6 or 0.8
Conv coding with code rates
1/2, 2/3 or 3/4
Outer FEC (if any)
None
RS(204,188,t=8)
RS(204,188,t=8) + MPE-FEC
BCH code (762,752)
Maximum travelling speed(km/h)
200 - 600
53 - 185
Time interleaving depth(ms)
385
0.6 - 3.5
0.6 - 3.5
200 - 500
Adaptive transmission(if any)
None
None
Multiple access method(if any)
None
None
Typical source coding
192 kbit/sMPEG2 Audiolayer 2
4 Mbit/sMPEG2
Not defined(MPEG-2 or H.264 w/MP2)

[edit] ADSL
OFDM is used in ADSL connections that follow the G.DMT (ITU G.992.1) standard, in which existing copper wires are used to achieve high-speed data connections.
Long copper wires suffer from attenuation at high frequencies. The fact that OFDM can cope with this frequency selective attenuation and with narrow-band interference are the main reasons it is frequently used in applications such as ADSL modems. However, DSL cannot be used on every copper pair; interference may become significant if more than 25% of phone lines coming into a central office are used for DSL.[citation needed]
For experimental amateur radio applications, users have even hooked up commercial off-the-shelf ADSL equipment to radio transceivers which simply shift the bands used to the radio frequencies the user has licensed.

[edit] Powerline Technology
OFDM is used by powerline devices to extend Ethernet connections to other rooms in a home through its power wiring. Adaptive modulation is particularly important with such a noisy channel as electrical wiring.[citation needed]

[edit] Wireless local area networks (LAN) and metropolitan area networks (MAN)
OFDM is also now being used in some wireless LAN and MAN applications, including IEEE 802.11a/g (and the defunct European alternative HIPERLAN/2) and WiMAX.
IEEE 802.11a, operating in the 5 GHz band, specifies airside data rates ranging from 6 to 54 Mbit/s. Four different modulation schemes are used: BPSK, 4-QAM, 16-QAM, and 64-QAM, along with a number of convolutional encoding schemes. This allows the system to adapt to the optimum data rate vs. error rate for the current conditions.
Clearwire, a wireless Internet Service Provider who provides access to metropolitan areas across the United States, utilizes OFDM in both their current 2.5 GHz network and the planned expansion of their WiMax network.

[edit] Wireless personal area networks (PAN)
OFDM is also now being used in the WiMedia / Ecma-368 standard for high-speed wireless personal area networks in the 3.1-10.6 GHz ultrawideband spectrum. See www.wimedia.com.

[edit] Terrestrial digital radio and television broadcasting
Much of Europe and Asia has adopted OFDM for terrestrial broadcasting of digital television (DVB-T, DVB-H and T-DMB) and radio (EUREKA 147 DAB, Digital Radio Mondiale, HD Radio and T-DMB).

[edit] DVB-T
By Directive of the European Commission, all television services transmitted to viewers in the European Community must use a transmission system that has been standardized by a recognized European standardization body,[2] and such a standard has been developed and codified by the DVB Project, Digital Video Broadcasting (DVB); Framing structure, channel coding and modulation for digital terrestrial television.[3] Customarily referred to as DVB-T, the standard calls for the exclusive use of COFDM for modulation. DVB-T is now widely used in Europe and elsewhere for terrestrial digital TV.

[edit] COFDM vs. VSB
The question of the relative technical merits of COFDM versus 8VSB has been a subject of some controversy, especially between Europe and USA. The United States has rejected several proposals to adopt COFDM for its digital television services, and has instead opted for 8VSB (vestigial sideband modulation) operation.
One of the major benefits provided by COFDM is that it renders radio broadcasts relatively immune to multipath distortion and signal fading due to atmospheric conditions or passing aircraft. Proponents of COFDM argue that it resists multipath far better than 8VSB. Early 8VSB DTV (digital television) receivers often had difficulty receiving a signal in urban environments.
However, newer 8VSB receivers are far better at dealing with multipath, hence the difference in performance may diminish with advances in demodulator design. Moreover, 8VSB modulation requires less power to transmit a signal the same distance, i.e., the received carrier-to-noise threshold is lower for the same bit error rate. In less-populated areas, 8VSB may have an advantage because of this. In urban areas, however, COFDM is believed to offer better reception than 8VSB.
In practice, it may be impossible to settle this debate without empirical history. One difficulty in fully assessing the two systems' relative performance in multipath environments is that the spatial distribution of multipath cannot be modeled well. Due to the chaotic nature of multipath, the process is non-stationary, both temporally and spatially, in the stochastic sense. Thus, the probability distribution of impaired receiving locations is not tractable.

[edit] Digital radio
COFDM is also used for other radio standards, for digital audio broadcasting (DAB), the standard for digital audio broadcasting at VHF frequencies, and also for Digital Radio Mondiale (DRM), the standard for digital broadcasting at shortwave and mediumwave frequencies (below 30 MHz).
The USA again uses an alternate standard, a proprietary system developed by iBiquity dubbed "HD Radio". However, it uses COFDM as the underlying broadcast technology to add digital audio to AM (medium wave) and FM broadcasts.
Both Digital Radio Mondiale and HD Radio are classified as in-band on-channel systems, unlike Eureka 147 (DAB: Digital audio broadcasting) which uses separate VHF or UHF frequency bands instead.

[edit] BST-OFDM used in ISDB
The BST-OFDM (Band Segmented Transmission Orthogonal Frequency Division Multiplexing) system proposed for Japan — in the ISDB-T, ISDB-TSB and ISDB-C broadcasting systems — improves upon COFDM by exploiting the fact that some OFDM carriers may be modulated differently from others within the same multiplex. Some forms of COFDM already offer this kind of hierarchical modulation, though BST-OFDM is intended to make it more flexible. The 6 MHz television channel may therefore be "segmented", with different segments being modulated differently and used for different services.
It is possible, for example, to send an audio service on a segment that includes a segment comprised of a number of carriers, a data service on another segment and a television service on yet another segment - all within the same 6 MHz television channel. Furthermore, these may be modulated with different parameters so that, for example, the audio and data services could be optimized for mobile reception, while the television service is optimized for stationary reception in a high-multipath environment.

[edit] Ultra-wideband
UWB (ultra-wideband) wireless personal area network technology may also utilise OFDM, such as in Multiband OFDM (MB-OFDM). This UWB specification is advocated by the WiMedia Alliance (formerly by both the Multiband OFDM Alliance {MBOA} and the WiMedia Alliance, but the two have now merged), and is one of the competing UWB radio interfaces.

[edit] Flash-OFDM
Flash-OFDM (Fast Low-latency Access with Seamless Handoff Orthogonal Frequency Division Multiplexing), which is also referred to as F-OFDM, is a system that is based on OFDM and specifies also higher protocol layers. It has been developed and is marketed by Flarion. Flash-OFDM has generated interest as a packet-switched cellular bearer, on which area it would compete with GSM and 3G networks. As an example, old 450 MHz frequency bands that were used by NMT-450 and C-Net C450 (both 1G analog networks, now mostly decommissioned) in Europe are already being licensed to Flash-OFDM operators. In Finland the license holder Digita has begun deployment of its nationwide "@450" wireless network, operational in parts of the country since April 2007 and planned coverage of all of Finland in 2009.
American wireless carrier Sprint Nextel had stated plans for field testing Flash-OFDM (along with other wireless broadband network technologies) for their 4G offering, which will be deployed using the licenses they own nationwide in the 2.5 GHz frequency range. Sprint subsequently has decided to deploy the mobile version of WiMAX, which is based on SOFDMA, scalable orthogonal frequency division multiple access technology.
T-Mobile already offers Flash-OFDM connection to its subscribers in Slovakia. The maximum download speed is 5.3 Mbit/s, whereas upload speed is limited to 1.8 Mbit/s.
T-Mobile Germany uses FLASH-OFDM to backhaul Wi-Fi connections on Deutsche Bahn's ICE high speed trains. This deployment showcases the excellent mobility support of FLASH-OFDM as the ICE trains reach speeds up to 300 km/h.
Citizens Telephone Cooperative launched a Flash-OFDM service to subscribers in parts of Virginia in March, 2006. The maximum speed available is 1.5 Mbit/s.[4]
Digiweb Ltd. launched a mobile broadband network using FLASH-OFDM technology at 872 MHz in July 2007 in Ireland and also will be launching in Norway. Voice handsets are not yet available at the time of writing (November 2007). The deployment is live in a small area north of Dublin only.
Butler Networks is currently trialing FLASH-OFDM technology in Denmark at 872 MHz as well.
In The Netherlands, KPN-telecom will start a pilot around July 2007.