Wednesday, August 31, 2011

Justice Dept. to block AT&T's T-Mobile deal

The U.S. Department of Justice has filed a petition with a federal court in Washington, D.C. to block AT&T's proposed takeover of T-Mobile USA.
The Justice Department said that the deal would "substantially lessen competition" in the wireless industry, and thus, should be blocked from approval. The government organization said that if the deal was approved, it would violate U.S. antitrust laws, and potentially cause "higher prices, poorer quality services, fewer choices and fewer innovative products for the millions of American consumers who rely on mobile wireless services in their everyday lives."
"The combination of AT&T and T-Mobile would result in tens of millions of consumers all across the United States facing higher prices, fewer choices and lower quality products for mobile wireless services," Deputy Attorney General James M. Cole said in a statement. "Consumers across the country, including those in rural areas and those with lower incomes, benefit from competition among the nation's wireless carriers, particularly the four remaining national carriers. This lawsuit seeks to ensure that everyone can continue to receive the benefits of that competition."
Bloomberg was first to report on the news.
Earlier this year, AT&T announced its plans to acquire T-Mobile USA from Deutsche Telekom in a deal valued at $39 billion. As soon as the deal was announced, critics immediately chimed in, saying that it could stifle competition in the marketplace and ultimately hurt both consumers and competitors. Sprint, which would have been dwarfed by the combined AT&T and T-Mobile, was especially outspoken on the deal, saying that it would fight it to the end.
"Sprint urges the United States government to block this anticompetitive acquisition," the company said in a statement following the announcement of the deal. "This transaction will harm consumers and harm competition at a time when this country can least afford it. So on behalf of our customers, our industry, and our country, Sprint will fight this attempt by AT&T to undo the progress of the past 25 years and create a new Ma Bell duopoly."
However, both the Federal Communications Commission and the Department of Justice continued their review of the deal, and were expected to make their decision on the merger early next year. The fact that the Justice Department has already filed a petition against the deal is somewhat of a surprise

Read more:

Wednesday, March 23, 2011

AT&T Buying T-Mobile For $39 Billion

AT&T on Sunday announced that it would acquire T-Mobile from Deutsche Telekom in a $39 billion deal that will create the largest wireless carrier in the U.S. Almost immediately, AT&T started its campaign to get the deal approved by regulators. According to a report by Bloomberg News, AT&T had been in talks for sometime with T-Mobile's parent company, Deutsche Telekom, and won because it was willing to offer a higher cash component to the purchase price. AT&T was also willing to offer a higher than average breakup fee if the sale is cancelled, possibly due to regulatory requirements making it undesirable to complete on the deal.

T-Mobile and Sprint have been rumored to merge. Indeed, Sprint will need a merger partner and Verizon Wireless may be a potential option. That move would create a U.S. wireless duopoly with a bevy of smaller players. Verizon is planning to upgrade its entire nationwide network to LTE by the end of 2013. This unexpected AT&T/T-Mobile tie-up could have the effect of accelerating the 4G race in the US, which would make the American 4G networks some of the best on the planet. 
"It will improve network quality, and it will bring advanced LTE capabilities to more than 294 million people. Mobile broadband networks drive economic opportunity everywhere, and they enable the expanding high-tech ecosystem that includes device makers, cloud and content providers, app developers, customers, and more…This transaction delivers significant customer, shareowner and public benefits that are available at this level only from the combination of these two companies with complementary network technologies, spectrum positions and operations. We are confident in our ability to execute a seamless integration, and with additional spectrum and network capabilities"
Stephenson also played to President Obama’s goals for nationwide mobile broadband.
Indeed, Sprint spokesman John Taylor already made it clear where his company stands:
"The combination of AT&T and T-Mobile USA, if approved by the Department of Justice (DOJ) and Federal Communications Commission (FCC), would alter dramatically the structure of the communications industry. AT&T and Verizon are already by far the largest wireless providers. A combined AT&T and T-Mobile would be almost three times the size of Sprint, the third largest wireless competitor. If approved, the merger would result in a wireless industry dominated overwhelmingly by two vertically-integrated companies that control almost 80% of the US wireless post-paid market, as well as the availability and price of key inputs such as backhaul and access needed by other wireless companies to compete. The DOJ and the FCC must decide if this transaction is in the best interest of consumers and the US economy overall, and determine if innovation and robust competition would be impacted adversely and by this dramatic change in the structure of the industry"

Key Notes:
  • AT&T will pay $25 billion in cash with the rest in stock.
  • Deutsche Telekom will own about 8 percent of AT&T when the deal is done.
  • AT&T’s purchase of T-Mobile is a spectrum play.
  • AT&T estimated that it would take five years to build out its network density to what T-Mobile and AT&T have today.

For more information on the transaction, including background information and factsheets, visit

Innovation in Wireless Industry

In today’s age asking question like “What can technology do for me?” seems irrelevant. It should be other way round - “What you want technology to do for you?” Open up your mind, think about your wildest imagination and let your creativity flow, nothing is impossible. This is the age of a revolution in technology that would be an outburst with the advent of 4G wireless speed. Smart phones are getting even smarter each day and it’s just a matter of time when your phones will control all your daily activities. It all started with human interaction with digital world and now the digital world is interacting with human. Let me show you this video from QUALCOMM’s “Augmented Reality”:

Now, this is a game application on phones that has a come out live in real life. This stuff is really amazing and will completely change gaming industry for sure. Now think about the possibilities where we can use this kind of technology, for presentations, classroom lecture,  and the list is endless. 

People have started realizing the power of wireless technology and they need to deliver innovations faster from their lab to the masses. Many telecom operators have taken an initiative to open an innovation center. AT&T Foundry is one such step towards bringing out innovative ideas to the market and to the users. It is a joint collaborative effort of industry leaders like Alcatel Lucent, Ericsson, Samsung and many more. Verizon too has its own innovation center opened in USA. This center provides a platform for local developer, small company who can use the wireless infrastructure of the telecom operator to test their products and applications.

 A quote from Verizon website:

‘If we gave you the ability to WIRELESSLY NETWORK ANYTHING what would YOU do?’

So what would you do? Think about it!

Saturday, March 12, 2011

What is VOIP ?

VoIP (Voice over Internet Protocol) is simply the transmission of voice traffic over IP-based networks.

VoIP services convert your voice into a digital signal that travels over the Internet. If you are calling a regular phone number, the signal is converted to a regular telephone signal before it reaches the destination. Three methods VOIP calls: PC to PC, PC to fixed-network lines, and telephone calls via IP-based internal networks. Some people use VOIP in addition to their traditional phone service, since VOIP service providers usually offer lower rates than traditional phone companies, but sometimes doesn't offer 911 service, phone directory listings, 411 service, or other common phone services. While many VoIP providers offer these services, consistent industry-wide means of offering these are still developing.

So! How VOIP Works?

As said, the interesting thing about VoIP is that there is not just one way to place a call. There are three ways one use VOIP service to make a phone call is.
  • ATA
  • The simplest and most common way is through the use of a device called an ATA (analog telephone adaptor). The ATA allows you to connect a standard phone to your computer or your Internet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes the analog signal from your traditional phone and converts it into digital data for transmission over the Internet. Providers like Vonage and AT&T CallVantage are bundling ATAs free with their service. You simply crack the ATA out of the box, plug the cable from your phone that would normally go in the wall socket into the ATA, and you're ready to make VoIP calls. Some ATAs may ship with additional software that is loaded onto the host computer to configure it; but in any case, it's a very straightforward setup.
  • IP Phones -- These specialized phones look just like normal phones with a handset, cradle and buttons. But instead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernet connector. IP phones connect directly to your router and have all the hardware and software necessary right onboard to handle the IP call. Wi-Fi phones allow subscribing callers to make VoIP calls from any Wi-Fi hot spot.
  • Computer-to-computer -- This is certainly the easiest way to use VoIP. You don't even have to pay for long-distance calls. There are several companies offering free or very low-cost software that you can use for this type of VoIP. All you need is the software, a microphone,speakers, a sound card and an Internet connection, preferably a fast one like you would get through a cable or DSL modem. Except for your normal monthly ISP fee, there is usually no charge for computer-to-computer calls, no matter the distance.

·         Low cost: VOIP services in the United States allow you to call anywhere in    North America at no extra charge.
·         Portability: you can take your phone converter and phone number and use them wherever you travel in the world, just as long as you have access to a high-speed Internet connection.
·         Features:  Call forwarding, call waiting, voicemail, caller ID and three-way calling are some of the many services included with VOIP telephone service at no extra charge.
·         Video-conferencing: User can enjoy the benefit of simultaneously Voice plus video call used for video conferencing.

·         Limited Emergency Calls: Traditional phone equipment can trace your location. Emergency calls are diverted to the nearest call center where the operator can see your location in case you can't talk. However, because a voice-over-IP call is essentially a transfer of data between two IP addresses, not physical addresses, with VOIP there is currently no way to determine where your VOIP phone call is originating from.
·         Sound Quality And Reliability: VOIP QoS(Quality of service) depends many factors say for example your broadband connection, your hardware, the service provided by your provider (ISP), the destination of your call etc.
·         Security: The most prominent security issues over VoIP are identity and service theft, viruses and malware, denial of service, spamming, call tampering and phishing attacks.

VOIP Protocol Stack:

Watch this out:

Friday, March 4, 2011

What is Erlang ?

·         An Erlang is a unit of telecommunications traffic measurement. An Erlang represents the continuous use of one voice path.  In practice, it is used to describe the total traffic volume of one hour. If one person makes one call and occupies one channel for an hour than we say that the system has 1 Erlang of traffic on it. 

  • Unit of traffic Erlang is name after Agner Krarup Erlang. He was born in 1878 in Lønborg, Denmark. He was a pioneer in the study of telecommunications traffic and, through his studies, proposed a formula to calculate the fraction of callers served by a village exchange who would have to wait when attempting to place a call to someone outside the village. 
  • In 1909, he published his first work: The Theory of Probabilities and Telephone Conversations. He gained worldwide recognition for his work, and his formula was accepted for use by the General Post Office in the UK. 
  • During the 1940s, the Erlang became the accepted unit of telecommunication traffic measurement, and his formula is still used today in the design of modern telecommunications networks.
·         For example, if a group of user made 30 calls in one hour, and each call had an average call duration of 5 minutes, then the number of Erlangs this represents is worked out as follows:
Minutes of traffic in the hour
number of calls x duration
Minutes of traffic in the hour
30 x 5
Minutes of traffic in the hour
Hours of traffic in the hour
150 / 60
Hours of traffic in the hour
Traffic figure
2.5 Erlangs  

It is obvious that you will need at least three lines to handle this traffic. But even then, due to the random nature of calls, you will still have a significant rate of callers who do not get through and hear a busy signal instead.

Erlang related to telecommunication:
Averaged over time, one erlang of telephone traffic occupies exactly one channel. However, the arrival and closing of telephone calls are random processes. As time elapses, one erlang of traffic may occupy zero, one or multiple channels. The definition of the unit erlang does not say anything about how the traffic behaves statistically about this average. Thus one erlang of traffic can be generated for instance by
  • One call of infinite duration, or
  • A random process of many calls arriving and closing, such that the average number of active calls is one.
 Erlang formula and symbol
It is possible to express the way in which the number of Erlangs are required in the format of a simple function or formula.
A     =     λ     x     h
λ = the mean arrival rate of new calls
h = the mean call length or holding time
A = the traffic in Erlangs.
Using this simple Erlang function or Erlang formula, the traffic can easily be calculated.
Erlang-B and Erlang-C
Erlang calculations are further broken down as follows:
  • Erlang B:   The Erlang B is used to work out how many lines are required from a knowledge of the traffic figure during the busiest hour. The Erlang B figure assumes that any blocked calls are cleared immediately. This is the most commonly used figure to be used in any telecommunications capacity calculations.
  • Extended Erlang B:   The Extended Erlang B is similar to Erlang B, but it can be used to factor in the number of calls that are blocked and immediately tried again.
  • Erlang C:   The Erlang C model assumes that not all calls may be handled immediately and some calls are queued until they can be handled. This model is mainly deployed in a call center environment.
These different models are described in further detail below.

Erlang B
It is particularly important to understand the traffic volumes at peak times of the day. Telecommunications traffic, like many other commodities, varies over the course of the day, and also the week. It is therefore necessary to understand the telecommunications traffic at the peak times of the day and to be able to determine the acceptable level of service required. The Erlang B figure is designed to handle the peak or busy periods and to determine the level of service required in these periods.

Erlang C
The Erlang C model is used by call centres to determine how many staff or call stations are needed, based on the number of calls per hour, the average duration of call and the length of time calls are left in the queue. The Erlang C figure is somewhat more difficult to determine because there are more interdependent variables. The Erlang C figure, is nevertheless very important to determine if a call centre is to be set up, as callers do not like being kept waiting interminably, as so often happens.

What is E1 / T1 ?

The PDH (plesiochronous Digital Hierarchy) has 2 primary communication systems as its foundation.

These are,
T1 system based on 1544kbit/s that is recommended by ANSI &
E1 system based on 2048kbit/s that is recommended by ITU-T.

Common Characteristics :- 
  1. Both are having Same Sampling Frequency i.e. 8kHz.
  2. In both (E1 & T1) Number of samples/telephone signal = 8000/sec.
  3. In both (E1 & T1) Length of PCM Frame = 1/8000s = 125µs.
  4. In both (E1 & T1) Number of Bits in each code word = 8.
  5. In both (E1 & T1) Telephone Channel Bit Rate = 8000/s x 8 Bit = 64 kbit/s.

Differing Characteristics :-
  1. In E1 Encoding/Decoding is followed by A-Law while in T1 Encoding/Decoding is followed by µ-Law.
  2. In E1 - 13 Number of Segments in Characteristics while in T1 - 15 Number of Segments in Characteristics.
  3. In E1 - 32 Number of Timeslots PCM Frame while in T1 - 24Number of Timeslots PCM Frame.
  4. In E1 - 8 x 32 = 256 number of bits / PCM Frame while in T1 - 8 x 24 + 1* = 193 number of bits / PCM Frame. (* Signifies an additional bit).
  5. In E1 - (125µs x  8)/256 = approx 3.9µs is the length of an 8-bit Timeslot while in T1 - (125µs x  8)/193 = approx 5.2µs is the length of an 8-bit Timeslot.
  6. In E1 - 8000/s x 256 bits = 2048kbit/s is the Bit Rate of Time-Division Multiplexed Signal while in T1 - 8000/s x 193 bits = 1544kbit/s is the Bit Rate of Time-Division Multiplexed Signal.
Reference Source: 

Wednesday, February 23, 2011

What's the difference between "dB", "dBm", and "dBi" ?

I keep seeing people using the terms "dB", "dBm", and "dBi" interchangeably, when they actually mean very different things. So, here's a little background on the correct usage of the terms.

A dB is a RELATIVE measure of two different POWER levels. There's also dB relative to VOLTAGE levels, but I won't go into those, as we're mostly concerned with POWER levels in our discussions here. 3dB is twice (or half) as much, 6dB is four times, 10dB is ten times, and so on. The formula for calculating gain or loss in dB is: 10log P1/P2. It's used for stating the gain or loss of one device (P1) IN RELATION to another (P2). Thus, I can say that an amplifier has “30 dB of gain”, or I have “6dB total feedline loss”. I CANNOT say, “My amp puts out 30 dB”, or “I have a 24dB antenna”, as you must state what you're referencing it to, which is where the subscript comes in. The dB by itself is not an absolute number, but a ratio.

For amplifiers, a common reference unit is the dBm, with 0dBm being equal to 1 milliwatt. Thus, an amp with an output of 30dBm puts out 1 Watt. How much gain it has is a different matter entirely, and you can have two different amps, each with an output of 30dBm (1Watt), that have different gains, and require different levels of drive power to achieve their outputs. You can also have two different amps with the same gain that have different output powers.

There's also dBW (Referenced to 1 WATT), but you generally only use those when dealing with “Big Stuff”, as 30dBW is 1000w, and way beyond what we deal with here!

For antennas, a common reference unit is the dBi, which states the gain of an antenna as referenced to an ISOTROPIC source. An Isotropic source is the perfect omnidirectional radiator, a true “Point Source”, and does not exist in nature. It's useful for comparing antennas, as since it’s theoretical, it’s always the same. It's also 2.41 dB BIGGER than the next common unit of antenna gain, the dBd, and makes your antennas sound better in advertising. The dBd is the amount of gain an antenna has referenced to a DIPOLE antenna. A simple dipole antenna has a gain of 2.41dBi, and a gain of 0dBd, since we're comparing it to itself. 

If I say I have a “24dB antenna”, it means nothing, as I haven't told you what I referenced it to. It could be a 26.41dBi antenna (also 24dBd), or a 21.59dBd (also 24dBi) antenna, depending on what my original reference was. The difference is 4.81dB, a significant amount. Most antenna manufacturers have gotten away from playing this game, but the reference will be different in different fields. 

Reference Link:

Explain SS7 Protocols Stack ?

SS7 Protocols Stack

The standard SS7 protocol has 4 levels (layers) as defined in the OSI 7 Layer Model. The levels 1 to 3 constitute the Message Transfer Part (MTP) and level 4 is the User Part (Transport Layer in OSI).

  • MTP1 = Message Transfer Part 1
  • MTP2 = Message Transfer Part 2
  • MTP3 = Message Transfer Part 3
  • SCCP = Signaling Connection Control Part
  • TCAP = Transaction Capabilities Application Part
  • MAP = Mobile Application Part
  • INAP = Intelligent Network Application Part
  • ISUP = ISDN User Part

1.) Message Transfer Part (MTP Level 1) Physical
  • Provides an interface to the actual physical channel over which communication takes place
  • CCITT recommends 64Kbps transmission whereas ANSI recommends 56 Kbps
    2.) MTP Level 2 (Data Link)

    • Ensures accurate end-to-end transmission of a message across a signaling link
    • Variable Length Packet Messages are defined here
    • Implements flow control, message sequence validation, error checking and message retransmission
    • Monitor links and reports their status
    • Test links before allowing their use

    3.) MTP Level 3 (Network)
    • Message routing between signaling points in the SS7 network
    • Signaling network management that provides traffic, links and routing management, as well as congestion (flow) control
    • Re-routes traffic away from failed links and signaling points, controls traffic when congestion occurs
      4.) Signaling Connection Control Part (SCCP)
      ·         Provides connectionless and connection-oriented network services
      ·         Provides global title translation (GTT) capabilities above MTP level 3; translates numbers to DPCs and subsystem numbers
      ·         Provides more detailed addressing information than MTPs
      ·         Used as transport layer for TCAP (Transaction capabilities applications part) based services

      5.) Transaction Capabilities Applications Part (TCAP)

      ·         Exchange of non-circuit related data
              Between applications across the SS#7 network
              Using the SCCP service
      ·         Queries and responses sent between Signaling Switching Point (SSPs)  and Signaling Control Point (SCPs)
      ·         Sends and receives database information
              Credit card validation
              Routing information

      6.) Telephone User Part (TUP)
      • Basic call setup and tear down
      • In many countries, ISUP has replaced TUP for call management
        7.) ISDN User Part (ISUP)
        • Necessary messaging for setup and tear down of all circuits (voice and digital)
        • Messages are sent from a switch, to the switch where the next circuit connection is required
        • Call circuits are identified using circuit identification code (CIC)

        Explain SS7 Links ?

        SS7 Links

        The figure shows the relationship between the link names and the link location (type). Signaling links are logically organized by link type ("A" through "F") according to their use in the SS7 signaling network. 
        The "A" (access) links connect the signaling end points (e.g., an SCP or SSP) to the STPs. Only messages originating from or destined to the signaling end point are transmitted on an "A" link. The "B" (bridge) links connect the STP to another STP. The "C" (cross) link connects STPs performing identical functions into a mated pair. "D" (diagonal) links connect the secondary (e.g., local or regional) STP pair to a primary (e.g., inter-network gateway) STP pair in a quad-link configuration. "E" (extended) links connect the SSP to an alternate STP. An "F" (fully associated) link is connected between two signaling end points (i.e., SSPs and SCPs).All SPs (signalling points) are connected using (typically) pairs of Links. 

        All links use the same physical connections (typically DS0A - 56K bit/s or DS1 (T1)).


        A link (access)
        Connects signaling end point (SCP or SSP) to STP
        B link (bridge)
        Connects an STP to another STP; typically, a quad of B links interconnect peer (or primary) STPs (STPs from a network connect to STPs of another network)
        C link (cross)
        Connects STPs performing identical functions, forming a mated pair (for greater reliability)
        D link (diagonal)
        Connects a secondary (local or regional) STP pair to a primary (inter-network gateway) STP pair in a quad-link configuration; the distinction between B and D links is arbitrary
        E link (extended)
        Connects an SSP to an alternate STP
        F link
        (fully associated)
        Connects two signaling end points (SSPs and SCPs) in the same local network