ALL ABOUT WIRELESS TELECOMMUNICATION

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

2011-08-31T08:43:00.001-07:00

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: http://news.cnet.com/8301-13506_3-20099776-17/justice-dept-to-block-at-ts-t-mobile-deal/#ixzz1WcbmOv2K

AT&T Buying T-Mobile For $39 Billion

2011-03-23T15:20:00.000-07:00

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.

In a statement, AT&T CEO Randall Stephenson positioned the deal this way.

"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 www.MobilizeEverything.com.

Innovation in Wireless Industry

2011-03-23T14:34:00.000-07:00

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!

What is VOIP ?

2011-03-12T19:44:00.000-08:00

Definition:

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

Introduction:

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.

Advantages:

· 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.

Disadvantages:

· 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:

What is Erlang ?

2011-03-04T17:35:00.000-08:00

Definition:

· 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.

History:

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.

Example:

· 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	=	150
Hours of traffic in the hour	=	150 / 60
Hours of traffic in the hour	=	2.5
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

Where:
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 ?

2011-03-04T17:11:00.000-08:00

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 :-

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

Differing Characteristics :-

In E1 Encoding/Decoding is followed by A-Law while in T1 Encoding/Decoding is followed by µ-Law.
In E1 - 13 Number of Segments in Characteristics while in T1 - 15 Number of Segments in Characteristics.
In E1 - 32 Number of Timeslots / PCM Frame while in T1 - 24Number of Timeslots / PCM Frame.
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).
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.
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:

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

2011-02-23T22:55:00.000-08:00

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 ?

2011-02-23T22:40:00.000-08:00

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 ?

2011-02-23T21:44:00.000-08:00

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)).

Summary:

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

Explain SS7 Architecture ?

2011-02-23T19:23:00.000-08:00

SS7 Architecture

SS7 Architecture consists of three different entities:

SSP (Service Switching Point)
STP (Signal Transfer Point)
SCP (Service Control Point)

1. 1.) Service Switching Point (SSP)

The Service Switching Point (SSP) is the local exchange in the telephone network. An SSP can be a combination voice switch and SS7 switch, or an adjunct computer connected to the local exchange's voice switch. The SSP provides the functionality of communicating with the voice switch via the use of primitives and creating the packets, or signal units, needed for transmission in the SS7 network. The SSP function is to use the information provided by the calling party (such as dialed digits) and determine how to connect the call. A routing table will identify which trunk circuit to use to connect the call, and which exchange this trunk terminates at. An SS7 message must be sent to this adjacent exchange requesting a circuit connection on the specified trunk. There are very few features required of an SSP. The ability to send messages using the ISDN User Part (ISUP) protocol and the Transaction Capabilities Application Part (TCAP) protocol is the only requirement, other than the network management.

Summary:

SSPs are switches in SS7 network.
SSPs convert a dialed number from a subscriber line to SS7 signaling messages.
SSPs setup, manage and release voice circuits required to make a call.
SSPs send messages using the ISDN User Part (ISUP) and Transaction Capabilities Application Part (TCAP) protocols
SSP's function is to use a global title to determine how to connect a call using its routing table.

2.) Signal Transfer Point (STP)

STPs are packet switches, and act like routers in the SS7 network. Routes each incoming message to an outgoing signaling link, based on routing information contained in the SS7 messages and a pre-defined route table. It does not offer termination services. STPs are paired to ensure redundancy

There are three levels of STPs.

· National Signal Transfer Point
· International Signal Transfer Point
· Gateway Signal Transfer Point

National STP exists within the national network

· Protocol converters often interconnect a National and an International STP by converting from ANSI to ITU-TS.

International STP functions within an international network.

· All nodes connecting to an International STP must use the ITU-TS protocol standard.

Gateway STP converts signaling data from one protocol to another.

· Gateway STPs are often used as an access point to the international network.

· Depending on its location, the Gateway STP must be able to use both the International and National protocol standards.

Summary:

An STP is a router and/or a gateway in the SS7 network.

Messages are not originated by an STP..

If an originating SSP does not know the address of a destination SSP, the STP must provide it using Global Title Translation.

Gateway STPs serve as the interface into another network and they can provide protocol conversion.

STPs also provide traffic and usage measurements.

3.) SCP (Service Control Point)

An SCP is usually a computer used as a front end to a database system. It is an interface to application-specific databases. The address of an SCP is a point code, and the address of the database it interfaces with is a subsystem number. The database is an application entity which is accessed via the TCAP protocol. It accepts a query for information from a subsystem at another node. SCP is used by STP to perform a function called global title translation

What is SS7 ?

2011-02-21T23:45:00.000-08:00

SS7 (Common Channel Signaling System No. 7/Short Signalling 7/Signalling System Number 7/ C7)

1.) SS7 Definition
2.) SS7 Functionality
2.) SS7 Architecture
3.) SS7 Link Interface
4.) SS7 Signalling Protocols

1.) SS7 Definition :
Signalling System Number 7 (SS#7 or C7) is the protocol used by the telephone companies for interoffice signalling. SS7 is a critical component of modern telecommunications systems. SS7 is a communications protocol that provides signaling and control for various network services and capabilities. Every call in every network is dependent on SS7. Likewise, every mobile phone user is dependent on SS7 to allow inter-network roaming. SS7 is also the "glue" that sticks together circuit switched (traditional) networks with Internet protocol based networks.

2.) SS7 Functionality:

The SS7 network and protocol are used for:

basic call setup, management, and tear down
wireless services such as personal communications services (PCS), wireless roaming, and mobile subscriber authentication
local number portability (LNP)
toll-free (800/888) and toll (900) wireline services
enhanced call features such as call forwarding, calling party name/number display, and three-way calling
efficient and secure worldwide telecommunication

3.) SS7 Architecture:

There are three kinds of signaling points in the SS7 network as shown

SSP (Service Switching Point)
STP (Signal Transfer Point)
SCP (Service Control Point)

4.) SS7 Link Interface

Access Link (A Link)
Bridge Link (B Link)
Cross Link (C Link)
Diagonal Link (D Link)
Extended Link (E Link)
Fully Associated Link (F Link)

5.) SS7 Signalling Protocols

This is just an brief overview of the SS7 network layout.

What is LTE ?

2011-02-20T23:37:00.000-08:00

LTE, or Long Term Evolution, is a a 4th generation (4G) mobile broadband standard and is aimed to be the successor to the 3G technologies GSM/UMTS. It is currently in development and is considered the competitor to WiMAX. Carriers will include Verizon, Vodafone, AT&T, and many more worldwide. Verizon launched their LTE service in 38 markets in late 2010.

Like WiMAX, this technology provides broadband services wirelessly (like EVDO), but instead of transmitting signals via microwaves, LTE utilizes a radio platform. Users need an LTE modem to access the network, which can be in USB format, ExpressCard, PCMCIA, or embedded in a laptop; it will also likely be featured as the internet connection on PDAs and phones.

This super-fast network, which is promising theoretical peak download rates of up 100Mbps (real-world speeds will vary, of course), provides an alternative to DSL, cable, satellite, and dial-up internet, which will be a big boon to people living in areas that aren't currently serviced by a high-speed network. It will also free people from the burden of having to find a WiFi hotspot when they are on the road - as long as you have an LTE modem, you can connect to the internet anywhere in the service provider's coverage area!

Key Details:

Theoretical downlink peak data rates up to 100 Mbps with 20 MHz bandwidth
Theoretical uplink peak data rates up to 50 Mbps with 20 MHz bandwidth
Reduced latency to 10 msec round-trip time between user equipment and base station

LTE deployment is considerably behind WiMAX, and it likely won't be widely available until about 2012 .

What Verizon has to say about its LTE technology have a look.

AT&T iPhone beats Verizon in nationwide 3G speed tests

2011-02-20T23:16:00.000-08:00

'CNN - The AT&T iPhone's average download speed was 1,769 Kbps, and the average upload speed was 730 Kbps. By way of comparison, the Verizon iPhone's average download speed was 848 Kbps, and the average upload speed was 506 Kbps.,

Well ! what difference does it makes if AT&t beats Verizon 3G Speed test by 300-400Kbs. I mean what difference it would make if you stream video faster than your friend ultimately you both gonna watch the same video. In my opinion the actual test should be of " Network Availability and Accessibility".

What I mean by Network Availability and Accessibility ?

'CNN- However, the AT&T iPhone sometimes could not complete tests because it did not have a connection, whereas the Verizon iPhone successfully completed every test. In short, I found the Verizon iPhone to be slower with network transfers but more reliable with coverage. Reviewers at other publications had the same results.'

Now, that is what I mean... Verizon has far more better network coverage than At&T. If a user is not able to access the network then what is the use of all that high speed. Give it a thought !

Types Of Interfaces: 2G

2011-02-20T19:20:00.000-08:00

Different Types of interfaces:

l A interface

l B interface

l C interface

l D interface

l zz interface

l 39/xx interface

l Q interface

l T1 interface

l ISUP interface

1. A Interface

The A interface exists between the circuit switched domain of the CDMA network subsystem (NSS) and the base station subsystem (BSS). In terms of entity, it is the interface between the MSCe/MGW and BSC.

The A interface complies with the base station subsystem application part (BSAP) of signaling system SS7. It includes voice channels and signaling channels. The A interface is used to transmit the following messages:

l Mobile station (MS) management messages

l Base station (BS) management messages

l Mobility management messages

l Call processing messages

2. B Interface

The B interface exists between MSCe and VLR. It adopts an internal protocol without a specific signaling transmission mode. The B interface is used in the following cases:

l The MSCe obtains user messages from the VLR.

l The MSCe notifies the VLR to record location information when an MS updates its location.

l The MSCe notifies the HLR to update data through VLR when an MS activates a supplementary service or modifies the relevant data.

3. C Interface

The C interface exists between the MSCe and HLR. It complies with the mobile application part (MAP) of the SS7. The C interface is used in the following cases:

l The HLR sends routing information to the MSCe when an MS is called.

l The short message service (SMS) is implemented.

4. D Interface

The D interface exists between VLR and HLR. It complies with the MAP of the SS7. The D interface is used to transmit the following messages:

l Authentication data

l Location update messages

l User data indexed during call connection

l Data of supplementary services

l VLR recovery messages

5. zz Interface

The zz interface exists between two MSCes. It complies with the SIP with encapsulated ISUP (SIPI).

The zz interface provides the inter-office call control function for the narrowband circuit-switched domain services that are independent of the user plane bearer technology and the control plane transmission technology.

6. 39/xx Interface

The 39/xx interface exists between an MGW and a media resource function processor (MRFP). It complies with the Megaco/H.248. The 39/xx interface is used in the following cases:

l The MSCe controls dynamic and static resources of the transmission modes (IP/TDM) in the MGW during call processing, including terminal attributes, terminal connectivity, and media streams.

l The MSCe maintains and manages the status of the MGW.

7. Q Interface

The Q interface exists between the MSCe and the MC. It complies with the MAP of the SS7 to support the short message service.

8. T1 Interface

The T1 interface exists between the MSCe and the SCP. It complies with the WIN of the SS7 to support the intelligent network service
9. ISUP Interface
When the MSC serves as a gateway office, it provides the interface between the PSTN and other mobile network devices. It also controls the incoming calls and outgoing calls through ISUP or TUP of the SS7

Mobile Originated Short Messages

2011-02-19T23:06:00.000-08:00

A short messages originated from an MS may either be originated from the control channel or be originated from the service channel. This is subject to the setting of the MS.

Figure shows Procedure for originating a short message from the control channel

1. The originating MS constructs a short message (SM) and the SM is sent from the BSC over the control channel in the ADDS Transfer message to the MSCe. The message carries parameters such as the called address, telecom parameter identifier, location area identifier (LAI) and service bearer.

2. The MSCe converts the received SMD-REQ message into an SMDPP message, and then sends the SMDPP message to the message center (MC) serving the originating MS.

3. The MC returns an smdpp message to the MSCe to acknowledge receiving the message.

4. If a response-requested flag is carried in the ADDS Transfer message sent from the BSC, while the setting of software parameter 38 allows the response to the MS, the MSCe will send an ADDS Page message to the BSC to acknowledge the result of sending the SM.

5. If the MC does not have the routing address of the short message, the MC sends an SMSREQ message to the HLR for the address.

6. If the HLR does not have the address, the HLR sends an SMSREQ message to the MSCe/VLR that serves the terminating MS.

7. The MSCe’/VLR’ returns an smsreq message to the HLR. The message contains the parameter SMSADDR that indicates the routing address of the SM.

8. The HLR returns to the MC a smsreq message that carries SMSADDR.

9. The MC uses the address to send an SMDPP message containing the SM to the serving MSCe of the terminating MS.

10. The MSCe’ converts the SMDPP message into SMD-REQ message and sends it to MS’.

11. The MS’ returns an SMD-ACK message through the BSC after receiving the SM.

12. Upon receiving the SMD-ACK message, the MSCe’ returns a smdpp message to the MC.

Interfaces and Protocols

2011-02-19T22:37:00.000-08:00

The Telecommunication system adopts the distributed network structure. Its functions are distributed among different physical entities. To accomplish a task, all devices must work together. All devices and subsystems are interconnected through interfaces based on specific protocols. In the communication system, the messages that are required to coordinate different entities are called signaling. The signaling system instructs every part of a system to work and coordinate together to accomplish tasks. Signaling messages are described in interface protocols and specifications. An interface refers to the connecting point between two adjacent network entities. A protocol defines rules to be complied with for exchanging information on the connecting point. Protocols are of the layer structure. The protocol of an interface is often described in the application layer.

Network Architecture:

Tables Below shows all interfaces its corresponding protocols used between different telecom entities.

Entities Connected	Interface Name	Signaling and Protocol
MSCe/MGW–BSC	A	BSAP
MSCe–VLR	B	Internal protocol
MSCe–HLR	C	MAP
VLR–HLR	D	MAP
MSCe–MGW	39/xx	H.248
MSCe–MSCe	zz	SIPI
MSCe–MC	Q	MAP
MSCe–SCP	T1	WIN
MSCe–SG	–	SIGTRAN

What is WiMax ?

2011-02-19T21:53:00.000-08:00

WiMAX is an IP based, wireless broadband access technology that provides performance similar to 802.11/Wi-Fi networks with the coverage and QOS (quality of service) of cellular networks. WiMAX is also an acronym meaning "Worldwide Interoperability for Microwave Access (WiMAX).WiMax is a 4G technology used by Sprint. Sprint threw down with Clearwire's WiMAX 802.16e network, which was first out of the gate stateside. Simply stated, WiMAX is a scalable wireless access technology designed to provide high throughput over long distances.It also has the potential to bring high-speed access to areas that have, thus far, been blocked from it due to a lack of fiber.
So, how will WiMAX make this possible?
A WiMAX deployment will essentially have two hardware components — a WiMAX tower and a WiMAX receiver. The tower is not unlike a cell tower, but can provide coverage to an estimated 30-mile radius. The tower connects to the Internet using a high-bandwidth wireline connection (e.g., T3), or it provide a wireless backhaul to other WiMAX towers with LOS (line of sight) microwave links.Ultimately, a network of connected WiMAX towers will drive the deployment of an 802.20-based Global Area Network (GAN), closely resembling cellular networks, but with far fewer tower required to provide the same coverage. This will allow true ubiquitous access across the country or region, providing bandwidth comparable to cable Internet service, at the very least.

Watch this :

What is the difference between 3G and 4G?

2011-02-18T16:08:00.000-08:00

3G vs 4G

3G is currently the world’s best connection method when it comes to mobile phones, and especially mobile Internet. 3G stands for 3rd generation as it is just that in terms of the evolutionary path of the mobile phone industry. 4G means 4th generation. This is a set of standard that is being developed as a future successor of 3G in the very near future.

The biggest difference between the two is in the existence of compliant technologies. There are a bunch of technologies that fall under 3G, including WCDMA, EV-DO, and HSPA among others. Although a lot of mobile phone companies are quick to dub their technologies as 4G, such as LTE, WiMax, and UMB, none of these are actually compliant to the specifications set forth by the 4G standard. These technologies are often referred to as Pre-4G or 3.9G.

4G speeds are meant to exceed that of 3G. Current 3G speeds are topped out at 14Mbps downlink and 5.8Mbps uplink. To be able to qualify as a 4G technology, speeds of up to 100Mbps must be reached for a moving user and 1Gbps for a stationary user. So far, these speeds are only reachable with wired LANs.

Another key change in 4G is the abandonment of circuit switching. 3G technologies use a hybrid of circuit switching and packet switching. Circuit switching is a very old technology that has been used in telephone systems for a very long time. The downside to this technology is that it ties up the resource for as long as the connection is kept up. Packet switching is a technology that is very prevalent in computer networks but has since appeared in mobile phones as well. With packet switching, resources are only used when there is information to be sent across. The efficiency of packet switching allows the mobile phone company to squeeze more conversations into the same bandwidth. 4G technologies would no longer utilize circuit switching even for voice calls and video calls. All information that is passed around would be packet switched to enhance efficiency.

Wireless network Generation evolution from 1G to 4G: Watch this out pretty cool stuff.

What happens when I press a calling key on my cell phone?

2011-02-17T13:28:00.000-08:00

The phenomena of you hitting your call button to place a call is known as "Call Flow Process" in technical term. So the better way of asking this question would be What is the Basic call flow in a wireless network?
Now to understand the very basic Call flow you need to have an idea of the network architecture of wireless network which I have explained in my previous post. But for a quick overview, wireless network consists of BTS, BSC and MSC.
Now take an example of Number A calling a Number B:
The moment Number A places a call "Call Initiate signal" reaches MSC (Mobile Switching Center) informing it about your intention to call number B.
MSC is a like brain of the telecommunication network which has its nerves connected to different other network entity.
Now MSC checks if Number A is authorize or in other words whether it has a permission to make call or not. If it does it analysis the Number B and finds out where the current location of Number B in the network.
It finds the location of Number B with the help of HLR (Home Location Register).
HLR is like a cerebrum cortex in the brain which stores the information of all the users who are currently active on the network.
Once MSC gets the location information of Number B it instructs BSC ( Base Station Controller) to sent a signal to Number B intimating it about the incoming call from Number A.
Now BSC is like a Spinal Cord which helps to transfer signals to and fro from MSC.
BSC sends the signal to the correct Number B with the help of BTS ( Base Transceiver station).
BTS is like the endpoint of your nerve from the spinal cord. All the Mobile Station ( i.e nothing but your cell phone) which are active on the network are connected to BTS via Air interface and thats where the name comes from ' Wireless Telecommunication'.
Once BTS send the signal to Number B its cell phone starts ringing. As soon as Number B hits green key to answer the call exactly opposite signal flow takes places from BTS to BSC and then to MSC and finally call gets connected. All this signal exchanges takes place in a fraction of a sec and the time from you placing the call and hearing a ring back tone is usually between 5 to 6 sec.

Have a look interesting stuff:

How does a wireless Network look like ?

2011-02-16T22:21:00.000-08:00

To start with, three basic entities used in a Wireless Telecommunication are:

1.) Base Transceiver Station (BTS)

2.) Base Station Controller (BSC)

3.) Mobile Switching Center (MSC)

Picture shows how are they interconnected with each other and this is how a basic Wireless architecture looks like. Each entity has a specific role to play when you place a call or when you receive a call on your MS i.e Mobile station which is nothing but your Cell phone.

The complete architecture consists of few other network entities which too have some specific role to play during your call setup procedure. The pictures below shows complete architecture of a GSM network.

4.) HLR : Home Location register

5.) VLR: Visitor Location register

6.) EIR: Equipment Identity register

7.) Auc: Authentication center

To understand their inter working and role each entity plays during a call setup you need to study Basic Call Flow Process. " What happens when I hit the calling key on my cell phone? "

Have a look at the presentation for an Overview of a Cellular System:

What the hell is 4G ?

2011-02-16T13:09:00.000-08:00

Have you ever wonder what all those 'G's' means in 1G-2G-3G and now the latest Buzzz in the market 4G !.
G is short of 'Generation' and it refers to nothing but the timeline of the wireless telecommunication. In a layman term it is similar to AD and BC we use to refer timeline of our Human race. 1G refers to the start of the wireless telecom world and 4G refers to the present era. So when next time some one ask you, do you have a 4G mobile all he is referring to " Do you have the latest mobile phone".

But to understand 4G in detail you need to first understand wireless technology and its evolution from 1G. So for now 4G refers to 4th Generation of wireless telecommunication and it promises to bring lightning data speed to the end user. So that a user can enjoy surfing the internet and other data services even while they are mobile.

Thats what 4G is all about !

Have a Look here...

Recommendations:

1.) What is LTE ?

2.) What is WiMax?

3) Evolution from 1G to 4G.

3.) How does a Cellular Network looks like?