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Virtual Private Network the Solution of Networking

2007-09-04T11:55:00.000+08:00

Tunneling's technological conjugate and enkripsi makes VPN( Virtual Private Network) as technology which admirably and helps user work myriad it.
Both of its technology non-negotiable and discomfited sue again in forms one VPN'S communication. Both of technology it shall be fused to get perfect result, which is safe data communication and efficient. Safe meaning Your data secrecies awake regular and its perfection. Are not bungling side get to catch and reads Your data, even that data then grass at public communication band. Constant perfection awake fathoms a meaning not bungling person get to confound content and Your data path. It needs to be looked after since if was passing public band, a great many fad person
one that maybe just destroy Your midway data. To that is, why technology second this really gets essential role in be formed VPN'S communication solution.

Any kind Technology Tunneling?
To make one tunnel, necessary one its regulator protocol so tunnel logically it gets to walk with every consideration such as connection point to point actually. Now, in store maker protocol myriad tunnel who can be utilized. But, tunneling protocol that commonest and at most is utilized consisting of three types hereunder:

• Layer 2 Tunneling Protocol (L2TP)

L2TP is one tunneling protocol who fuses and compounding two numbers tunneling protocol who gets proprietary's character, which is L2F (Layer 2 Forwarding) Cisco Systems's belonging with PPTP (Point to Point Tunneling Protocol) Microsoft's belonging.

Initially, all product Cisco utilizes L2F to manage that tunneling, meanwhile operating system is antecedent Microsoft just utilizes PPTP to service its user that wants to play by tunnel. But currently, Windows NT's Microsoft / 2000 got utilizes PPTP or L2TP in technological that VPN.

L2TP usually being utilized deep makes Virtual Private Dial Network (VPDN) one that gets to work takes in all communications protocol type at in it. Besides, L2TP also gets independent media character because get to above work any media. L2TP enables its user for regular can connected with local network theirs with policy same security and of whichever they lie, via VPN'S connection or VPDN. This connection oft is looked on as medium lengthen local network belongs to its user, but passes through public media.

But, this tunneling's technology have no mechanism to provide enkripsi's facility because really quite a pure just form tunnel's network. Besides, what last grass in tunnel this can be a prey to and be monitored by use of protocol analizer.

• Generic Routing Encapsulation (GRE)
tunneling's protocol this the one has ability to take in more than one addressing protocol type communication. Are not just package get even one internet protocol address get to be taken in it, but a lot of other protocol package as CNLP, IPX, and a lot of again. But, all that was packaged or to enkapsulation becomes one package which get IP addressing system. Then that package is distributed system thru also tunnel working above IP communications protocol.

By use of tunneling GRE, router is aught at the end tunnel does enkapsulation other protocol packages in header of IP protocol. It will make package previously get to be taken in to whichever by and method that exist on IP technology. With marks sense this ability, therefore protocol which took in by that IP package gets free more move to go to whichever location which is wended, provided that achievable IP addressing ala.

Quite a lot application utilize tunneling's protocol help this is merge local network that separatedly distance ala is back get to get communication. Or in other words, GRP there are many is utilized to lengthen and mengekspansi is proprietary local network the its user. Even quite a lot is utilized, GRE not also provide enkripsi's system grasses last data at its tunnel, so all its data activity can monitor to utilize protocol analyzer so-so.

• security Protocol's IP (IPSec)
IPSec is one for felicitous tunneling protocol option to be utilized deep VPN korporat's level. IPSec constitutes protocol that gets open's character default who can provide data security, data perfection, and autentikation is data among peer second that participates in it.

IPSec provides data security system as it by use of one peacemaker method that named Internet Goes To y. Exchange (IKE). IKE this on call to handle negotiation problem of protocol and security algorithm that is created bases from policy which is applied on network the user. IKE on eventually will result one enkripsi's system and its peacemaker key that will be utilized for autentikasi on this IPSec's system.

How with Enkripsinya's Technology?
Besides technology tunneling, enkripsi's technology in VPN also highly varied. Actually technological enkripsi is not just belonging VPN just, but far-flung its purpose. Enkripsi on call to look after privasi and that data secrecy can't with easy to read by side that don't deserve. Marginally tech enkripsi is divided up two types, which is:

Symmetric Encryption
Symmetric Encryption is known even with nickname one diarrhoea goes to y. encryption. Enkripsi is this type a lot of is utilized deep enkripsi's process data in volumed one outgrows. Up to data communication term, network peripheral that have enkripsi's ability this type will change data that as text of purification (cleartext) as gets text form already at random or its terminology is ciphertext. This random text obviously been made by use of algorithm. This random text really is not easily to be read, so Your data security awakes.

Succeeding question, how is that random data opened by really party be wended? To open this random data, seeker's algorithm previously also makes one key which can open all content in origin. This key proprietary by the consigner and also data consignee. Key this is that will be utilized deep enkripsi's process and dekripsi ciphertext it.

Digital Encryption Default (DES) constitute one standard algorithm that is utilized to make this symmetric encryption's process. This algorithm at claim as one commonest being utilized currently. DES'S algorithm operating deep measure 64 bit obstructs data. Fathom a meaning, this algorithm will carry on one series of randomization process 64 incoming data bits for then is issued as 64 random data bits. That process utilizes 64 bit key whereabouts 56 its bit be chosen at random, 8 its bit comes from parity bits of Your data. Bit eighth that was slipped bit 56th betwixt previous.

Resulting key then is sent to data consignee.
With enkripsi's system such, DES is not easy to be conquered But along with technology developing, DES can be uncovered by use of supercomputer in the period of few days only. Alternative for DES is triple DES (3DES) one that do process in DES as much thrice. So key which is resulted and is needed to open enkripsi is as much three numbers.

Asymmetric Encryption
Enkripsi is this type is frequent at conceive of system public goes to y. encryption. enkripsi's process this type can utilize algorithm any kind, but enkripsi's result of this algorithm will function as complement in to seeker and data collation. In enkripsi this type is needful two peacemaker keys that variably, but mutually gets bearing in processes its algorithm. Both of this peacemaker key is frequent so-called with Public's terminology Goes To y. and Private Goes To y..
For example it, Andi and Kindness wants to get safe communication by use of system enkripsi this. To it, both has to have public goes to y. and private goes to y. beforehand. Andi shall have public and private goes to y., so even with Kindness. While processes communication be begun, they will utilize keys that variably to enkrip and dekrip is data. Key may variably, but data get flawlessly been delivered same algorithm blessing.

Public's makings mechanism and private goes to y. this complex enough. Usually goes to y. to y. this at generate utilizes going generator RSA'S algorithm (Ron Rivest, Adi Shamir, Leonard Adleman) or EL Gamal. Result of this generator usually is two random numeral formations huge ones. One random number functions as public goes to y. and one again for private to go to y.. This random numbers really have as much been made and as random as maybe to strengthen uniquenesses of go to y. to y. You.
To genberationi goes to y. to y. this really need tall CPU process. Therefore that, this process can't be done every time You do data transactions. In other words, enkripsi is this type never been utilized to secure data truthfully because its complex character it. Even so, enkripsi this will so effective in autentikasi's process data and its application that involve signature's digital system and goes to y. management.

How Choose VPN'S Technology in point?
VPN'S technology so a lot of its option for You to utilize. How choose the best one for You? VPN'S technology the best one for Your really clings to traffic's requirement data that wants then grass at its settle.
IPSec's technology constitute main option and the most complete to give solution for VPN'S network enterprise's level. But unhappily, IPSec just backs up traffic that berbasiskan IP and package technology that gets unicast's characteristic only. So if characteristic Your data that wants to be overlooked by appropriate VPN with competence IPSec, therefore not necessarily again utilizes it because IPSec easier relative at configuration and at troubleshoot. But if traffic You consisting of protocols besides IP or IP communication get multicast's characteristics, therefore utilizes GRE or L2TP.
Well-matched GRE is utilized if You want to make site to site VPN's communication that will be passed by various communications protocol kind. Besides, GRE also well-matched being utilized deep through multicast's IP package as one a lot of is utilized deep routing protocol. So match is utilized as band of communication among router. GRE that to enkapsulation will all traffic without source and aim care it.
For network what do a lot of impassable by traffic for stationary networking Microsoft, L2TP really close-fitting to be utilized in here. Since its relationship that hand in glove with PPP protocol, L2TP also well-matched being utilized deep build access VPN's remote that need multiprotokol's support.
But one becomes constraint be good GRE and also L2TP no that have enkripsi's system and data perfection keeper. Therefore of that, usually deep implementation both of this VPN'S technology merged by its purpose with IPSec to get enkripsi's facility and integrity keeper mechanism its data.

Safe and Comfortable
VPN really molded of second conjugate technological already been enlightened in broad outline upon. There is one principle which amends among data communication practitioner that says that “ safe data communication will never cozy ”. That principle maybe available its scorpion is right, whereabouts You shall make policy policy that dazes to carry the wind, tunneling's teches and enkripsi what do You will utilize, and rule rule what do so tights and play by play to stop all rioter that don't deserve to access Your data. But, technology VPN may can be counted out deep that principle.

Really correctness, performa is network VPN won't can as good as tissue personal truthfully. Big latensi time must espouse to whichever VPN goes. Besides, this network so sensitif to happening trouble midway entah whereabouts. But, all that risk still maybe accepted since if was connected, tremendous convenience You can enjoy. Moreover, to You practitioner carry on business, a great many business applications which can be made by use of VPN.

Proxy For Sharing Internet

2007-07-26T13:34:00.000+08:00

Tech proxy is tech standard one for ala internet access goes together by severally computer at a swoop in one Local Network's Area (LAN) via one modem or one communication channel. Proxy's terminology own a lot of recognised / is utilized especially at the world / diplomatic circle. Classically proxy is someone / acting institute as intermediate or on behalf from other people / institute / other state.

This tech is recognised with severally name which is at marketing, e.g.:

• Connection Sharing's internet (ICS) – this terminology utilized by Microsoft on its Windows 2000.
• Proxy is server – it usually as software of affix that is assembled at acting computer as intermediate.
• Sharing's internet server (ISS) – usually as selfsupporting hardware furnished with modem, hub and proxy's software in it.
• Network Address Translation (NAT) – other terminology that is utilized for proxy's software server.
• IP Masquerade – tech that is utilized at NAT / Proxy's software server to do proxy's process.

Why is proxy's tech becomes to be of important for share internet access from one LAN ala goes together? As picture of common, in one computer network – included Internets, all network component at identifies by one number (at Internet is known as Protocol internet address, internet protocol address, IP address). Why utilized by number? Since IP's number purpose will make easy route's process & forwarding data – than if utilize no name its order. Approximately kindred concept ala by patterns that is used at phone number.

Nah is its hoodoo, (1 ) this IP's number its circumscribed amount and (2 ) we oftentimes not want persons to know from computer which / network which we access Internet in order not to be opened for attack cracker from Internet network that its public character.

Base two (2 ) main reasons upon, therefore developed private network's concept, private's network or then recognised with IntraNet (as foe from Internet). This IntraNet's network is next become basis for network at complex about office, plant wide, campus, Internet booth (WARNET) etcetera. Technologically no difference it among IntraNet & Internet, difference that significant is internet protocol address that is utilized. In Internet deal, one Intanet (private's network) can utilize internet protocol address in 192.168.x.x's region or 10.x.x.x. IP 192.168 & 10 not at all utilized by Internets because really is allocated for IntraNet's need only.

pengkaitan's process tissue typing second that variably it is done in a simple via one computer or going tool proxy's software upon. So on functioning computer as intermediate this, will ever have two (2 ) interface (among face), usually one as modem to tack on to Internet network, and one Ethernet Card to tack on to IntraNet's network that private's cipher.

To link network second that variably it, which is Internet & IntraNet, need to be done by translasi address / IP Address. proxy / Network Address Translation's tech own for that matter simple by use of table eight (8 ) columns, one that meaty information:

• workstation's internet protocol address that asks for relationship.
• workstation's application port that asks for relationship.
• proxy's internet protocol address server that accepts to see dammed hell first proxy.
• proxy's application port server that accepts to see dammed hell first proxy.
• proxy's internet protocol address server that keeps on to see dammed hell first proxy
• proxy's application port server that keeps on to see dammed hell first proxy.
• Intent server internet protocol address.
• Intent server application port.

In this way, package with internet protocol address couple information:port from workstation user what does ask for internet protocol address couple service:intent server port can be substituted that server intenting to suspect that service requisition its coming from internet protocol address couple:proxy's port server that keeps on to see dammed hell first proxy. Intent server will send all requested data to internet protocol address couple:proxy's port server that keeps on to see dammed hell first proxy – is next keep on it again to internet protocol address couple:workstation's port user that utilize 192.168.x.x's internet protocol address.

If we see blur ala, for that matter tech proxy this constitute simplest tech from one firewall. Why? With proxy's tech, intent server doesn't know that computer address that ask for that data for that matter is at turn back proxy server & utilize private's internet protocol address 192.168.x.x.

The concept of the IMS (IP Multimedia Service)Procedure

2007-04-09T15:34:00.000+08:00

The IMS foundation four safety the main specification.

One safety
Delivery the service of multimedia communication characteristically real time and person to person with the IP basis (like voice or videotelepony), likewise his matter with communication person to mechine (like the service gabling).

Two safety
Integrated the service of multimedia communication real time (like the video streaming and live chatting).

Three safety
Could serve and interact with the service and the application that were varied like combined presence and instant messaging.

Four safety
The Ease in melkukan the set up the multi-service in one sesion single or multi sesion simultaneously.

IMS could in toimplementasion wth GPR/EDGE, CDMA EV-DO, UMTS, xDSL or WLAN. To use IMS, the operator carried him out parallel from the available network.
When the operator wanted to place the service voice with communication IP, then IMS increased the application VoIP parallel.

As signaling protocol him, IMS used SIP (session initiation protocol) that was standardised by IETF (the internet engineering task force).
However, because initially standarisai this only was focussed to fixed the internet, the standard that was used by the network selullar will be different.
The theory SIP proxy server this that was worn 3GPP as the concept of the IMS foundation.

SIP that was used by IMS was as protocol application servers and softswitch that could interact with IAD or Access Gateway as well as developed communication between caller and that was called.

SIP could be used constructive caller, nerworking, as well as constructive the session of multimedia communication.
At this time was developed SIP especially to telepony that was mentioned SIP-T. the numbering System SIP-T fully pointed system cash that was used ITU-T, as well as to akomododir the mechanism interegrasi the service telepony with web like: UMS, the internet call waiting, click to dial and instant messanging

Wireless Communication technology for the Multimedia

2007-03-22T09:26:00.000+08:00

Bluetooth is Service was a communication technology wireless (without the cable) that operated in the frequency tape 2.4 of GHz unlicensed ISM (industrial, Scientific and Medical) by using one frequency hopping tranceiver that could provide the service and the voice of data communication in a manner real-time between host-host bluetooth and the distance of the range of the limited service (around 10 metre).
Bluetooth personally could take the form of card that the form and his function almost be the same as card that was used to wireless local the area network (WLAN) where using the frequency of IEEE standard radio 802.
11, only in bluetooth had the range of the distance of the shorter service and the transfer capacity of the lower data.

Basically bluetooth was created not only to replace or eliminated the use of the cable inside carried out the exchange of information, but also could offer fitur that was good for technology mobile wireless at a cost of that was relatively low, consumption of the low power, interoperability that was promising, was easy in the operation and could provide the service that various things.

To give the picture that was sharper concerning technology bluetooth that was relatively new this to the reader, along with was untangled about the history of the emergence bluetooth and his development, technology that was used in the system bluetooth and the aspect of the service that could be provided, as well as few analyses about the comparison of the modulation method spread spectrum FHSS (Frequency Hopping Spread Spectrum) that was used by bluetooth compared with the method spread spectrum DSSS (Direct Sequence Spread Spectrum).
In May 1998, the champion's 5 companies that is Ericsson, IBM, Intelligence, Nokia and Toshiba formed one of Special Interest Group (SIG) and began to make the specification that was named by them ‘bluetooth’.

In July 1999 the specification document bluetooth the version 1.
0 began to be launched.
In December 1999 was begun again by the production of the specification document bluetooth the version 2.
0 with the addition 4 new champions that is 3Com, Lucent Technologies, Microsoft and Motorola.
At this time, more than 1800 companies in various fields in part in the field semiconductor manufacture, PC manufacture, mobile network carrier, companies automobile and water lines bergambung in a consortium as adopter technology bluetooth.
These foremost companies in part like Compaq, Xircom, Phillips, Texas instruments, Sony, BMW, the Puma, NEC, Casio, Boeing, etc..
Although the Bluetooth SIG standard at this time ‘dimiliki’ by the group of the champion but he will it was hoped become a IEEE standard (802.15).

Protocol bluetooth used a combination between circuit switching and packet switching.
Bluetooth could support a data canal asinkron, three synchronous voice canals simultaneous or a canal where simultaneously supported the data service asinkron and the synchronous voice.
Each voice canal supported a synchronous voice canal 64 kb/s.
The canal asinkron could support the maximal speed 723.2 kb/s asymmetric, where for the direction was the reverse could support up to the speed 57.6 family planning/s.
Whereas to mode symmetrical could support up to the speed 433.9 kb/s.

An equipment that had technology wireless bluetooth will have the capacity to carry out the exchange of information with the distance of the range up to 10 metre (~30 feet).
The system bluetooth provided the communication service point to point and communication point to multipoint.
The product bluetooth could take the form of PC card or USB adapter that was put into equipment.
Equipment that could diintegerasikan with technology bluetooth in part: mobile PC, mobile phone, PDA (Personal Digital Assistant), headset, the camera, printer, router et cetera.
Applications that could be provided by the service bluetooth this in part: PC to PC file the transfer, PC to PC file synch (notebook to desktop), PC to mobile phone, PC to PDA, wireless headset, LAN connection via ethernet access point and etc.

Bluetooth FHSS vs WLAN DSSS
In fact why bluetooth more chose the FHSS method (Frequency Hopping Spread Spectrum) compared with DSSS (Direct Sequence Spread Spectrum).
The reason that made why bluetooth did not use DSSS in part as follows:

1.FHSS needed consumption of the power and the complexity that were lower compared with DSSS this was caused because DSSS used the speed chip (chip rate) compared with the speed of the symbol (symbol rate) that was used by FHSS, so as cost that was needed to use DSSS will be higher.

2.FHSS used FSK where endurance of the disturbance noise relative better compared with DSSS that usually uses QPSK (for IEEE 802.
11 2 Mbps) or CCK (IEEE 802.
11b 11 Mbps).

Although FHSS had the distance of the range and the transfer of the data that was lower compared with DSSS but for the service was supervised 2 Mbps FHSS could give the solution cost-effective that was better.

Seamless Mobility

2007-03-15T12:20:00.000+08:00

During him to no longer distinguished the function of the house telephone, the office telephone, personal enamel, and phonsel you.
The existence seamless mobility enabled anyone to terkoneksi by any, including communicating with plasama the house TV.

For the last three year you have enjoyed many benefits of cellular technology latest, from that was simplest sepertti voice calling, smsed, GPRS, MMS, the video streaming, the speed of data access via EDGE and CDMA-1X, to now that just just emerged like technology 3G and convergence between cellular and access broadband.

Further possibly you still do not think about many matters to terkoneksi with the digital thing thing in and around us.
The concept seamless mobility that was introduced by Avaya Technology and Motorola became one of the new solutions to speed up the process kovergensi between you, technology wired and wireless, and any available in and around us.

Alcatel and Ericsson mentioned him as mobile triple play tried to be visited by you http://www.freescape.com/seamless, there was shown that this concept enabled you to continue terkoneksi and synchronisation with the digital device that has disesuikan with your personal data.
For example when you were being in the carriage sembari saw the presentation of football from live the TV in the screen phonsel, and you too could continue him when arriving in the station and following the trip by the car.

You could listen to his direct report from car stereo radio or the screen plasmaTV in the car.
When until in home, then the presentation live this TV will continue to the personal TV in the house without did setting the manual to the channel TV. And like that henceforth.

This all was enabled by the blessing konfergensi the network that was high between wired and wireless, used internet technology protocol (IP), and interoperability and kontabilitythat was high.
The concept seamless mobility made the cellular phone the main key from all the activities, good in the house, in the office, and anywhere.
And as means to connect with the directory database in the office and most synchronous with digital equipment in the house as re-beads.

All was done in a real manner time by giving priority to the concept seamless that connected between humankind and equipment, humankind with humankind, equipment with humankind, and equipment with equipment.
Seamless mobility introduced in user a new convergence technology that connected between the function wired and wireless, voice and the data, as well as local and wide the area.
Wired voice and the network of the data in the function enterprise him normally went with fitur voip, in his convergence between keduan him, the data went together on the network of the cellular area, and in implemetasikan inside voice-enabled WLANs.
The intensity in this very convergent environment, the user only needed one kind handheld then.
To connect the two funsi this, then was created one single handheld
that very multimode, the multi-band, and multimedia.
One of them with made dual-network phone that could terkoneksi with IEEE 802.
11 standards of Wi-Fi to connect with FANTASTIC-based IP-PBX, and in the environment of the cellular phone industry could do seamless handoff between WLAN access point in one industrial environment, and outside between voice enabled-WLAN and the network selular.
This solution was very useful for enterprise market, because handheld the user apart from terkoneksi with the cellular channel, also could enjoy facilities that were given by the company, including the conference call, call hold and access voice inside database and the company's directory.

Basically with the existence of the solution seamless mobility, was hoped the profit that was obtained from convergence of several networks was to reduce the management cost from several networks.
With the technological emergence 3G this end end, was hoped for by all sitem multimedia as well as the implementation seamless mobility could real was felt in and around us, his matter with the existence 3G, technology seamless mobility can be realised.

Lightweight Directory Access Protocol

2007-02-13T15:43:00.000+08:00

The Lightweight Directory Access Protocol, or LDAP in computer networking,is a networking protocol for querying and modifying directory services running over TCP/IP.

A directory is a set of information with similar attributes organized in a logical and hierarchical manner. The most common example is the telephone directory, which consists of a series of names (either of a person or organization) organized alphabetically, with an address and phone number attached.

An LDAP directory often reflects various political, geographic, and/or organizational boundaries, depending on the model chosen. LDAP deployments today tend to use Domain Name System (DNS) names for structuring the topmost levels of the hierarchy. Deeper inside the directory might appear entries representing people, organizational units, printers, documents, groups of people or anything else which represents a given tree entry (or multiple entries).

Telecommunication companies introduced the concept of directory services to information technology and computer networking, as their understanding of directory requirements was well-developed after some 70 years of producing and managing telephone directories. The culmination of this input was the comprehensive X.500 specification, a suite of protocols produced by the International Telecommunication Union (ITU) in the 1980s.

X.500 directory services were traditionally accessed via the X.500 Directory Access Protocol (DAP), which required the Open Systems Interconnection (OSI) protocol stack. LDAP was originally intended to be a "lightweight" alternative protocol for accessing X.500 directory services through the simpler (and now widespread) TCP/IP protocol stack. This model of directory access was borrowed from the DIXIE and Directory Assistance Service protocols.

Standalone LDAP directory servers soon followed, as did directory servers supporting both DAP and LDAP. The latter has become popular in enterprises, as LDAP removed any need to deploy an OSI network. Today, X.500 directory protocols including DAP can also be used directly over TCP/IP.

The protocol was originally created by Tim Howes of the University of Michigan, Steve Kille of ISODE and Wengyik Yeong of Performance Systems International, circa 1993. Further development has been done via the Internet Engineering Task Force (IETF).

In the early engineering stages of LDAP, it was known as Lightweight Directory Browsing Protocol, or LDBP. It was renamed as the scope of the protocol was expanded to include not only directory browsing and searching functions, but also directory update functions.

LDAP has influenced subsequent Internet protocols, including later versions of X.500, XML Enabled Directory (XED), Directory Service Markup Language (DSML), Service Provisioning Markup Language (SPML), and the Service Location Protocol (SLP).

Protocol LDAP
A client starts an LDAP session by connecting to an LDAP server, by default on TCP port 389. The client then sends operation requests to the server, and the server sends responses in turn. With some exceptions the client need not wait for a response before sending the next request, and the server may send the responses in any order.

The basic operations are:

* Start TLS - optionally protect the connection with Transport Layer Security (TLS), to have a more secure connection
* Bind - authenticate and specify LDAP protocol version
* Search - search for and/or retrieve directory entries
* Compare - test if a named entry contains a given attribute value
* Add a new entry
* Delete an entry
* Modify an entry
* Modify DN - move or rename an entry
* Abandon - abort a previous request
* Extended Operation - generic operation used to define other operations
* Unbind - close the connection (not the inverse of Bind)

In addition the server may send "Unsolicited Notifications" that are not responses to any request, e.g. before it times out a connection.

A common alternate method of securing LDAP communication is using an SSL tunnel. This is denoted in LDAP URLs by using the URL scheme "ldaps". The default port for LDAP over SSL is 636. The use of LDAP over SSL was common in LDAP Version 2 (LDAPv2) but it was never standardized in any formal specification. This usage has been deprecated along with LDAPv2, which was officially retired in 2003.

LDAP is defined in terms of ASN.1, and protocol messages are encoded in the binary format BER. It uses textual representations for a number of ASN.1 fields/types, however.

The protocol accesses LDAP directories, which follow the 1993 edition of the X.500 model:

* A directory is a tree of directory entries.
* An entry consists of a set of attributes.
* An attribute has a name (an attribute type or attribute description) and one or more values. The attributes are defined in a schema (see below).
* Each entry has a unique identifier: its Distinguished Name (DN). This consists of its Relative Distinguished Name (RDN) constructed from some attribute(s) in the entry, followed by the parent entry's DN. Think of the DN as a full filename and the RDN as a relative filename in a folder.

Be aware that a DN may change over the lifetime of the entry, for instance, when entries are moved within a tree. To reliably and unambiguously identify entries, a UUID may be provided in the set of the entry's operational attributes.

An entry can look like this when represented in LDIF format (LDAP itself is a binary protocol):

dn: cn=John Doe,dc=example,dc=com
cn: John Doe
givenName: John
sn: Doe
telephoneNumber: +1 888 555 6789
telephoneNumber: +1 888 555 1234
mail: john@example.com
manager: cn=Barbara Doe,dc=example,dc=com
objectClass: inetOrgPerson
objectClass: organizationalPerson
objectClass: person
objectClass: top

dn is the name of the entry; it's not an attribute nor part of the entry. "cn=John Doe" is the entry's RDN, and "dc=example,dc=com" is the DN of the parent entry. The other lines show the attributes in the entry. Attribute names are typically mnemonic strings, like "cn" for common name, "dc" for domain component, and "mail" for e-mail address.

A server holds a subtree starting from a specific entry, e.g. "dc=example,dc=com" and its children. Servers may also hold references to other servers, so an attempt to access "ou=department,dc=example,dc=com" could return a referral or continuation reference to a server which holds that part of the directory tree. The client can then contact the other server. Some servers also support chaining, which means the server contacts the other server and returns the results to the client.

LDAP rarely defines any ordering: The server may return the values in an attribute, the attributes in an entry, and the entries found by a search operation in any order. This follows from the formal definitions - an entry is defined as a set of attributes, and an attribute is a set of values, and sets are inherently unordered.

GPS of Navigation Signals

2007-02-05T15:13:00.000+08:00

The user's GPS receiver is the user segment of the GPS system. In general, GPS receivers are composed of an antenna, tuned to the frequencies transmitted by the satellites, receiver-processors, and a highly-stable clock (often a crystal oscillator). They may also include a display for providing location and speed information to the user. A receiver is often described by its number of channels: this signifies how many satellites it can monitor simultaneously. Originally limited to four or five, this has progressively increased over the years such that, as of 2006, receivers typically have between twelve and twenty channels.

GPS receivers may include an input for differential corrections, using the RTCM SC-104 format. This is typically in the form of a RS-232 port at 4,800 bps speed. Data is actually sent at a much lower rate, which limits the accuracy of the signal sent using RTCM. Receivers with internal DGPS receivers can outperform those using external RTCM data. As of 2006, even low-cost units commonly include WAAS receivers.

Many GPS receivers can relay position data to a PC or other device using the NMEA 0183 protocol. NMEA 2000[9] is a newer and less widely adopted protocol. Both are proprietary and controlled by the US-based National Marine Electronics Association. References to the NMEA protocols have been compiled from public records, allowing open source tools like gpsd to read the protocol without violating intellectual property laws. Other proprietary protocols exist as well, such as the SiRF protocol. Receivers can interface with other devices using methods including a serial connection, USB or Bluetooth.

Navigation signals
GPS satellites broadcast three different types of data in the primary navigation signal. The first is the almanac which sends coarse time information along with status information about the satellites. The second is the ephemeris, which contains orbital information that allows the receiver to calculate the position of the satellite. This data is included in the 37,500 bit Navigation Message, which takes 12.5 minutes to send at 50 bps.

GPS satellites broadcast three different types of data in the primary navigation signal. The first is the almanac which sends coarse time information along with status information about the satellites. The second is the ephemeris, which contains orbital information that allows the receiver to calculate the position of the satellite. This data is included in the 37,500 bit Navigation Message, which takes 12.5 minutes to send at 50 bps.

The satellites also broadcast two forms of clock information, the Coarse / Acquisition code, or C/A which is freely available to the public, and the restricted Precise code, or P-code, usually reserved for military applications. The C/A code is a 1,023 bit long pseudo-random code broadcast at 1.023 MHz, repeating every millisecond. Each satellite sends a distinct C/A code, which allows it to be uniquely identified.

The P-code is a similar code broadcast at 10.23 MHz, but it repeats only once a week. In normal operation, the so-called "anti-spoofing mode", the P code is first encrypted into the Y-code, or P(Y), which can only be decrypted by units with a valid decryption key. Frequencies used by GPS include:
• L1 (1575.42 MHz) - Mix of Navigation Message, coarse-acquisition (C/A) code and encrypted precision P(Y) code.
• L2 (1227.60 MHz) - P(Y) code, plus the new L2C code on the Block IIR-M and newer satellites.
• L3 (1381.05 MHz) - Used by the Defense Support Program to signal detection of missile launches, nuclear detonations, and other high-energy infrared events.
• L4 (1379.913 MHz) - Being studied for additional ionospheric correction.
• L5 (1176.45 MHz) - Proposed for use as a civilian safety-of-life (SoL) signal (see GPS Modernization). This frequency falls into an internationally protected range for aeronautical navigation, promising little or no interference under all circumstances. The first Block IIF satellite that would provide this signal is set to be launched in 2008.

Calculating positions
The coordinates are calculated according to the World Geodetic System WGS84 coordinates system. To calculate its position, a receiver needs to know the precise time. The satellites are equipped with extremely accurate atomic clocks, and the receiver uses an internal crystal oscillator-based clock that is continually updated using the signals from the satellites.

The receiver identifies each satellite's signal by its distinct C/A code pattern, then measures the time delay for each satellite. To do this, the receiver produces an identical C/A sequence using the same seed number as the satellite. By lining up the two sequences, the receiver can measure the delay and calculate the distance to the satellite, called the pseudorange.

The orbital position data from the Navigation Message is then used to calculate the satellite's precise position. Knowing the position and the distance of a satellite indicates that the receiver is located somewhere on the surface of an imaginary sphere centered on that satellite and whose radius is the distance to it. When four satellites are measured simultaneously, the intersection of the four imaginary spheres reveals the location of the receiver. Earth-based users can substitute the sphere of the planet for one satellite by using their altitude. Often, these spheres will overlap slightly instead of meeting at one point, so the receiver will yield a mathematically most-probable position (and often indicate the uncertainty).

Calculating a position with the P(Y) signal is generally similar in concept, assuming one can decrypt it. The encryption is essentially a safety mechanism; if a signal can be successfully decrypted, it is reasonable to assume it is a real signal being sent by a GPS satellite. In comparison, civil receivers are highly vulnerable to spoofing since correctly formated C/A signals can be generated using readily available signal generators. RAIM features will not help, since RAIM only checks the signals from a navigational perspective.

Accuracy and Error Sources
The position calculated by a GPS receiver requires the current time, the position of the satellite and the measured delay of the received signal. The position accuracy is primarily dependent on the satellite position and signal delay.
To measure the delay, the receiver compares the bit sequence received from the satellite with an internally generated version. By comparing the rising and trailing edges of the bit transitions, modern electronics can measure signal offset to within about 1% of a bit time, or approximately 10 nanoseconds for the C/A code. Since GPS signals propagate nearly at the speed of light, this represents an error of about 3 meters. This is the minimum error possible using only the GPS C/A signal.

Position accuracy can be improved by using the higher-speed P(Y) signal. Assuming the same 1% accuracy, the faster P(Y) signal results in an accuracy of about 30 centimeters.
Electronics errors are one of several accuracy-degrading effects outlined in the table below. When taken together, autonomous civilian GPS horizontal position fixes are typically accurate to about 15 meters (50 ft). These effects also reduce the more precise P(Y) code's accuracy.

Atmospheric effects
Changing atmospheric conditions change the speed of the GPS signals as they pass through the Earth's atmosphere and ionosphere. Correcting these errors is a significant challenge to improving GPS position accuracy. These effects are minimized when the satellite is directly overhead, and become greater for satellites nearer the horizon, since the signal is affected for a longer time. Once the receiver's approximate location is known, a mathematical model can be used to estimate and compensate for these errors.

Because ionospheric delay affects the speed of radio waves differently based on frequency, a characteristic known as dispersion, both frequency bands can be used to help reduce this error. Some military and expensive survey-grade civilian receivers compare the different delays in the L1 and L2 frequencies to measure atmospheric dispersion, and apply a more precise correction. This can be done in civilian receivers without decrypting the P(Y) signal carried on L2, by tracking the carrier wave instead of the modulated code. To facilitate this on lower cost receivers, a new civilian code signal on L2, called L2C, was added to the Block IIR-M satellites, first launched in 2005. It allows a direct comparison of the L1 and L2 signals using the coded signal instead of the carrier wave.

The effects of the ionosphere are generally slow-moving, and can be averaged over time. The effects for any particular geographical area can be easily calculated by comparing the GPS-measured position to a known surveyed location. This correction is also valid for other receivers in the same general location. Several systems send this information over radio or other links to allow L1 only receivers to make ionospheric corrections. The ionospheric data are transmitted via satellite in Satellite Based Augmentation Systems such as WAAS, which transmits it on the GPS frequency using a special PRN, so only one antenna and receiver are required.

Humidity also causes a variable delay, resulting in errors similar to ionospheric delay, but occurring in the troposphere. This effect is much more localized, and changes more quickly than the ionospheric effects, making precise compensation for humidity more difficult. Altitude also causes a variable delay, as the signal passes through less atmosphere at higher elevations. Since the GPS receiver measures altitude directly, this is much simpler correction to apply.

Multipath effects
GPS signals can also be affected by multipath issues, where the radio signals reflect off surrounding terrain; buildings, canyon walls, hard ground, etc. These delayed signals can cause inaccuracy. A variety of techniques, most notably narrow correlator spacing, have been developed to mitigate multipath errors. For long delay multipath, the receiver itself can recognize the wayward signal and discard it. To address shorter delay multipath from the signal reflecting off the ground, specialized antennas may be used. Short delay reflections are harder to filter out since they are only slightly delayed, causing effects almost indistinguishable from routine fluctuations in atmospheric delay.

Multipath effects are much less severe in moving vehicles. When the GPS antenna is moving, the false solutions using reflected signals quickly fail to converge and only the direct signals result in stable solutions.

Ephemeris and clock errors
The navigation message from a satellite is sent out only every 12.5 minutes. In reality, the data contained in these messages tend to be "out of date" by an even larger amount. Consider the case when a GPS satellite is boosted back into a proper orbit; for some time following the maneuver, the receiver’s calculation of the satellite's position will be incorrect until it receives another ephemeris update. The onboard clocks are extremely accurate, but they do suffer from some clock drift. This problem tends to be very small, but may add up to 2 meters (6 ft) of inaccuracy.
This class of error is more "stable" than ionospheric problems and tends to change over days or weeks rather than minutes. This makes correction fairly simple by sending out a more accurate almanac on a separate channel.

Selective availability
The GPS includes a feature called Selective Availability (SA) that introduces intentional errors between 0 meters and up to a hundred meters (300 ft) into the publicly available navigation signals, making it difficult to use for guiding long range missiles to precise targets. Additional accuracy was available in the signal, but in an encrypted form that was only available to the United States military, its allies and a few others, mostly government users.

SA typically added signal errors of up to about 10 meters (30 ft) horizontally and 30 meters (100 ft) vertically. The inaccuracy of the civilian signal was deliberately encoded so as not to change very quickly, for instance the entire eastern U.S. area might read 30 m off, but 30 m off everywhere and in the same direction. In order to improve the usefulness of GPS for civilian navigation, Differential GPS was used by many civilian GPS receivers to greatly improve accuracy.

During the Gulf War, the shortage of military GPS units and the wide availability of civilian ones among personnel resulted in a decision to disable Selective Availability. This was ironic, as SA had been introduced specifically for these situations, allowing friendly troops to use the signal for accurate navigation, while at the same time denying it to the enemy. But since SA was also denying the same accuracy to thousands of friendly troops, turning it off or setting it to a error of 0 meters (effectively the same thing) presented a clear benefit.

In the 1990s, the FAA started pressuring the military to turn off SA permanently. This would save the FAA millions of dollars every year in maintenance of their own radio navigation systems. The military resisted for most of the 1990s, but SA was eventually "discontinued"; the amount of error added was "set to zero" in 2000 following an announcement by U.S. President Bill Clinton, allowing users access to an undegraded L1 signal. Per the directive, the induced error of SA was changed to add no error to the public signals (C/A code). Selective Availability is still a system capability of GPS, and error could be in theory reintroduced at any time. In practice, in view of the hazards and costs this would induce for US and foreign shipping, it is unlikely to be reintroduced, and various government agencies, including the FAA, have stated that it is not intended to be reintroduced.
The US military has developed the ability to locally deny GPS (and other navigation services) to hostile forces in a specific area of crisis without affecting the rest of the world or its own military systems.

GPS jamming
Jammingof any radio navigation system, including satellite based navigation, is possible. The U.S. Air Force conducted GPS jamming exercises in 2003 and they also have GPS anti-spoofing capabilities. In 2002, a detailed description of how to build a short range GPS L1 C/A jammer was published in Phrack issue 60 by an anonymous author. There has also been at least one well-documented case of unintentional jamming, tracing back to a malfunctioning TV antenna preamplifier. If stronger signals were generated intentionally, they could potentially interfere with aviation

GPS receivers within line of sight. According to John Ruley, of AVweb, "IFR pilots should have a fallback plan in case of a GPS malfunction". Receiver Autonomous Integrity Monitoring(RAIM), a feature of some aviation and marine receivers, is designed to provide a warning to the user if jamming or another problem is detected. GPS signals can also be interfered with by natural geomagnetic storms, predominantly at high latitudes.

The U.S. government believes that such jammers were also used occasionally during the 2001 war in Afghanistan. Some officials believe that jammers could be used to attract the precision-guided munitions towards non-combatant infrastructure; other officials believe that the jammers are completely ineffective. In either case, the jammers may be attractive targets for anti-radiation missiles. During the Iraq War, the U.S. military claimed to destroy a GPS jammer with a GPS-guided bomb.

Relativity
According to Einstein's Theory of relativity, because of their constant movement and height relative to the Earth Centered Inertial reference frame the clocks on the satellites are affected by their speed (special relativity) as well as their gravitational potential (general relativity). Friedwardt Winterberg predicted in 1955 that when observed from the Earth's reference frame, satellite clocks would be perceived as running at a slightly faster rate than clocks on the Earth's surface.

For GPS satellites, this discrepancy is 38 microseconds per day. To account for this, the frequency standard on-board the satellites are given a rate offset prior to launch, making it run slightly slower than its desired frequency on Earth, at 10.22999999543 MHz instead of 10.23 MHz, a difference of -4.465 parts in 1010. The atomic clocks on board the GPS satellites are precisely tuned, making this a practical engineering application of the scientific theory of relativity in a real-world system.

Another relativistic effect to be compensated for in GPS observation processing is the Sagnac effect. The GPS time scale is defined in an inertial system, but observations are processed in an ECEF (co-rotating) system, in which simultaneity is not uniquely defined. The Lorentz transformation between the two systems modifies the signal run time, a correction having opposite algebraic signs for satellites in the Eastern and Western celestial hemispheres. Ignoring this effect will produce an East-West offset in the absolute position solution on the order of tens of metres
Neil Ashby presented in Physics Today (May 2002) an account how these relativistic corrections are applied, and their orders of magnitude. The error introduced by relativistic effects can be as much as 15 meters. The GPS system also makes adjustments for the relativistic drift of the atomic clocks in the satellites. Parts of this correction are carried out in the satellites and parts in the receiver.

Techniques to improve accuracy
Augmentation methods of improving accuracy rely on external information being integrated into the calculation process. There are many such systems in place and they are generally named or described based on how the GPS sensor receives the information. Some systems transmit additional information about sources of error (such as clock drift, ephemeris, or ionospheric delay), others provide direct measurements of how much the signal was off in the past, while a third group provide additional navigational or vehicle information to be integrated in the calculation process.

Examples of augmentation systems include the Wide Area Augmentation System, Differential GPS, and Inertial Navigation Systems

Precise Monitoring
The accuracy of a calculation can also be improved through precise monitoring and measuring of the existing GPS signals in additional or alternate ways.
The first is called Dual Frequency monitoring, and refers to systems that can compare two or more signals, such as the L1 frequency to the L2 frequency. Since these are two different frequencies, they are affected in different, yet predictable ways by the atmosphere and objects around the receiver. After monitoring these signals, it is possible to calculate how much error is being introduced and then nullify that error.
Receivers that have the correct decryption key can relatively easily decode the P(Y)-code transmitted on both L1 and L2 to measure the error. Receivers that do not possess the key can still use a process called codeless to compare the encrypted information on L1 and L2 to gain much of the same error information. However, this technique is currently limited to specialized surveying equipment. In the future, additional civilian codes are expected to be transmitted on the L2 and L5 frequencies. When these become operational, non-encrypted users will be able to make the same comparison and directly measure some errors.

A second form of precise monitoring is called Carrier-Phase Enhancement (CPGPS). The error, which this corrects, arises because the pulse transition of the PRN is not instantaneous, and thus the correlation (satellite-receiver sequence matching) operation is imperfect. The CPGPS approach utilizes the L1 carrier wave, which has a period 1000 times smaller than that of the C/A bit period, to act as an additional clock signal and resolve the uncertainty. The phase difference error in the normal

GPS amounts to between 2 and 3 meters (6 to 10 ft) of ambiguity. CPGPS working to within 1% of perfect transition reduces this error to 3 millimeters (1 inch) of ambiguity. By eliminating this source of error, CPGPS coupled with DGPS normally realizes between 20 and 30 centimeters (8 to 12 inches) of absolute accuracy.
Relative Kinematic Positioning (RKP) is another approach for a precise GPS-based positioning system. In this approach, determination of range signal can be resolved to an accuracy of less than 10 centimeters (4 in). This is done by resolving the number of cycles in which the signal is transmitted and received by the receiver. This can be accomplished by using a combination of differential GPS (DGPS) correction data, transmitting GPS signal phase information and ambiguity resolution techniques via statistical tests — possibly with processing in real-time (real-time kinematic positioning, RTK).

GPS Time
Atomic clocks on the satellites are set to "GPS time", similar to most time standards, but not corrected to the rotation of the Earth, ignoring leap seconds and other corrections. GPS time was set to match Coordinated Universal Time (UTC) in 1980, but has since diverged as leap seconds were added to UTC.
The current date is expressed in the GPS signal as a week number and a day-of-week number. GPS week zero started at 00:00:00 UTC (00:00:19 TAI) on January 6, 1980. The week number is transmitted in a ten-bit field, and so it wraps round every 1,024 weeks, (19.7 years). The transmitted week number rolled over to zero at 00:00:19 TAI on August 22, 1999 (23:59:47 UTC on August 21, 1999). GPS receivers thus need to know the time to within 3,584 days in order to correctly interpret the GPS date signal. A new field is being added to the GPS navigation message that specifies the calendar year number exactly, in a sixteen-bit field.

The GPS navigation message includes the difference between GPS time and UTC, which is 14 seconds as of 2006. Receivers subtract this offset from GPS time to calculate UTC and 'local' time. New GPS units may not show the correct UTC time, or not attempt to show UTC time at all, until after receiving the UTC offset message for the first time. This is usually within 15 minutes after the unit achieves GPS lock. The GPS-UTC offset field is only eight bits, and so it wraps round every 256 leap seconds. At the current rate of change of the earth's rotation, the first wraparound of this field is projected to occur in the year 2330.

GPS Modernization
Having reached Fully Operational Capability on July 17, 1995, the GPS completed its original design goals. However, additional advances in technology and new demands on the existing system led to the effort to "modernize" the GPS system. Announcements from the Vice Presidential and the White House in 1998 heralded the beginning of these changes and in 2000, the U.S. Congress reaffirmed the effort; referred to it as GPS III.
The project aims to improve the accuracy and availability for all users and involves new ground stations, new satellites, and four additional navigation signals. New civilian signals are called L2C', L5 and L1C; The new military code is called M-Code. A goal of 2013 has been established with incentives offered to the contractors if they can complete it by 2011

Basics GPS

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GPS stands for Global Positioning System and it is a system that can provide a position at any point on the Earth's surface to a very high degree of accuracy. The Global Positioning System (GPS) uses 24 active Navstar satellites in orbit 11 000 miles above the surface of the Earth.

Using economic ground based receivers GPS is able to provide position information to within a number of metres. The economic costs have also meant that it is now fitted to many motor vehicles, while separate GPS receivers can be bought for a few hundred pounds or dollars. As a result it is widely used by private individuals, as well as many commercial and professional users. In fact the primary use for GPS is as a military navigation system. The fact that it is used so widely is a by product of its success.

Basic concept
GPS operates by being able to measure the distances from the satellites that are in orbit around the Earth. By knowing the distance from a number of satellites, it is possible to calculate the position on the Earth's surface and the height above it by a process of triangulation. This a great simplification, but this is essentially how it works.

The satellites all send timing information so the receiver knows when the message was sent. As radio signals travel at the speed of light they take a very short but finite time to travel the distance from the satellite to the receiver. The satellites also transmit information about their positions. In this way the receiver is able to calculate the distance from the satellite to the receiver. To obtain a full fix, four satellites are required, and when the receiver is in the clear, more than four satellites are in view all the time.

Satellites
The satellites are orbiting above the Earth. Their orbits are tightly controlled because errors in their orbit will translate to errors in the final positions. The time signals are also tightly controlled. The satellites contain an atomic clock so that the time signals they transmit are very accurate. Even so these clocks will drift slightly and to overcome this, signals from Earth stations are used to correct this.

The satellites themselves have a design life of ten years, but to ensure that there are no holes in service in the case of unexpected failures, spares are held in orbit and these can be brought into service at short notice.

The satellites are provide their own power through their solar panels. These extend to about 17 feet, and provide the 700 watts needed to power the satellite and its batteries when it is in sunlight. Naturally the satellite needs t remain operation when it is on the dark side of the Earth when the solar panels do not provide any power. This means that when in sunlight the solar panels need to provide additional power to charge batteries, beyond just powering the basic satellite circuitry.

Receivers
A large number of GPS receivers are available today. They make widespread use of digital signalling processing techniques. The transmissions from the satellites use spread spectrum technology, and the signal processors correlate the signals received to recover the data. As the signals are very weak it takes some time after the receiver is turned on to gain the first fix. This Time To First Fix (TTFF) may be as long as twelve minutes, although receivers that us a large number of correlators are able to shorten this.

When using a GPS receiver the receiver must be in the open. Buildings, or any structure will mask the signals and it may mean that few satellites can be seen. Thus the receivers will not operate inside buildings, and urban areas may often cause problems.

Applications
The primary use for GPS is as a military navigational aid. Run by the American Department of Defense its primary role is to provide American forces with an accurate means of navigation anywhere on the globe. However its use has been opened up so that commercial and private users have access to the signals and can use the system. Accordingly it is very widely used for many commercial applications from aircraft navigation, ship navigation to surveying, and anywhere where location information is required. For private users very cost effective receivers are available these days and may be used for applications including sailing. Even many motor vehicles have them fitted now to provide SatNav systems enabling them to navigate easily without the need for additional maps.

It can be said that GPS has revolutionised global navigation since it became available. Prior to this navigation systems were comparatively localised, and did not offer anything like the same degrees of accuracy, flexibility and coverage.

NAVSTAR Global Positioning System

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The Global Positioning System (GPS), is currently the only fully-functional satellite navigation system. More than two dozen GPS satellites are in medium Earth orbit, transmitting signals allowing GPS receivers to determine location, speed and direction.

Since the first experimental satellite was launched in 1978, GPS has become indispensable for navigation around the world, and an important tool for map-making and land surveying. GPS also provides a precise time reference used in many applications including scientific study of earthquakes, and synchronization of telecommunications networks.

Developed by the United States Department of Defense, it is officially named NAVSTAR GPS (Navigation Signal Timing and Ranging Global Positioning System). The satellite constellation is managed by the United States Air Force 50th Space Wing. Although the cost of maintaining the system is approximately US$400 million per year, including the replacement of aging satellites, GPS is free for civilian use as a public good.

A GPS receiver calculates its position by measuring the distance between itself and three or more GPS satellites. Measuring the time delay between transmission and reception of each GPS radio signal gives the distance to each satellite, since the signal travels at a known speed. The signals also carry information about the satellites' location. By determining the position of, and distance to, at least three satellites, the receiver can compute its location using trilateration. Receivers do not have perfectly accurate clocks, and must track one extra satellite to correct their clock error.

Technical description
System segmentation
The current GPS consists of three major segments. These are the space segment (SS), a control segment (CS), and a user segment (US).

Space segment
The space segment is composed of the orbiting GPS satellites, or Space Vehicles (SV) in GPS parlance. The GPS design calls for 24 SVs to be distributed equally among six circular orbital planes. The orbital planes are centered on the Earth, and not rotating with respect to the distant stars. The six planes have approximately 55° inclination (tilt relative to the equator) and are separated by 60° right ascension of the ascending node (angle along the equator).

Orbiting at an altitude of approximately 20,000 kilometers (11,000 nautical miles), each SV makes two complete orbits each sidereal day, so it passes over the same location on Earth once each day. The orbits are arranged so that at least six satellites are always within line of sight from almost anywhere on Earth.

As of January 2007, there are 29 actively broadcasting satellites in the GPS constellation. The additional satellites improve the precision of GPS receiver calculations by providing redundant measurements. With the increased number of satellites, the constellation was changed to a nonuniform arrangement. Such an arrangement was shown to improve reliability and availability of the system, relative to a uniform system, when multiple satellites fail.

The flight paths of the satellites are tracked by monitoring stations in Hawaii, Kwajalein, Ascension Island, Diego Garcia, and Colorado Springs, Colorado, along with monitor stations from other agencies. The tracking information is sent to the Air Force Space Command's master control station at Schriever Air Force Base, Colorado Springs, Colorado, which is operated by the 2d Space Operations Squadron (2 SOPS) of the United States Air Force (USAF). 2 SOPS contacts each GPS satellite regularly with a navigational update (using the ground antennas at Ascension Island, Diego Garcia, Kwajalein, and Colorado Springs). These updates synchronize the atomic clocks on board the satellites to within one microsecond and adjust the ephemeris of each satellite's internal orbital model. The updates are created by a Kalman Filter which uses inputs from the ground monitoring stations, space weather information, and other various inputs.

System Satellite Orbits

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The variety of different orbits that can be adopted for satellites. The ones that receive the most attention are the geostationary orbit used by many communications and direct broadcast satellites for satellite television and also the low earth orbit ones that travel around the global. Those used in the Navstar or Global Positioning (GPS) system occupy a relatively low earth orbit. There are also many other types of satellite from weather satellites to research satellites and many others.

The actual orbit that is chosen will depend on factors including its function, and the area it is to serve. In some instances the orbit may be as low as 100 miles (160 km) for a low earth orbit (LEO), whereas others may be over 22 000 miles (36000 km) high as in the case of a geostationary orbit. The satellite may even have an elliptical rather than a circular orbit.

Gravity
As satellites orbit the earth they are pulled back in by the force of the gravitational field. If they did not have any motion of their own they would fall back to earth, burning up in the upper reaches of the atmosphere. Instead the motion of the satellite rotating around the earth has a force associated with it pushing it away from the earth. For any given orbit there is a speed for which gravity and the centrifugal force balance each other and the satellite remains in a stable orbit, neither gaining height nor loosing it.

Obviously the lower the orbit, the stronger the gravitational pull, and this means that the satellite must orbit the earth faster to counteract this pull. Further away the gravitational field is less and the satellite velocities are correspondingly less. For a very low orbit of around 100 miles a velocity of about 17500 miles per hour is needed and this means that the satellite will orbit the earth in about 90 minutes. At an altitude of 22 000 miles a velocity of just less than 7000 miles per hour is needed giving an orbit time of about 24 hours.

Circular and elliptical orbits
A satellite can orbit the earth in one of two basic types of orbit. The most obvious is a circular orbit where the distance from the earth remains the same at all times. A second type of satellite orbit is an elliptical one.

When a satellite orbits the earth, either in a circular or elliptical orbit, the satellite orbit forms a plane that passes through the centre of gravity or geocentre of the Earth. The rotation around the earth is also categorised. It may be in the same direction as the earth's rotation when it is said to be posigrade, or it may be in the opposite direction when it is retrograde.

The track of the satellite around the globe is often defined as well. The point on the Earth's surface where the satellite is directly overhead moves around the globe. This is known as the ground track. This forms a circle which has the geocentre at its centre. It is worth noting that geostationary satellites are a special case as they appear directly over the same point of the earth all the time. This means that their ground track consists of a single point on the earth's equator. Also for satellites with equatorial orbits the ground track is along the equator.

Satellites may also be in other orbits. These will cross the equator twice, once in a northerly direction, and once in a southerly direction. The point at which the groundtrack crosses the equator is known as a node. There are two, and the one where the groundtrack passes from the southern hemisphere to the northern hemisphere is called the ascending node. The one where the groundtrack passes from the northern to the southern hemisphere is called the descending node. For these orbits it is usually found that the groundtrack shifts towards the west for each orbit because the earth is rotating towards the east underneath the satellite.

For many orbit calculations it is necessary to consider the height of the satellite above the geocentre. This is the height above the earth plus the radius of the earth. This is generally taken to be 3960 miles or 6370 km.

Velocity is another important factor as already seen. For a circular orbit it is always the same. However in the case of an elliptical one this is not the case as the speed changes dependent upon the position in the orbit. It reaches a maximum when it is closest to the earth and it has to combat the greatest gravitational pull, and it is at its lowest speed when it is furthest away.

Elliptical orbits are often used, particularly for satellites that only need to cover a portion of the Earth's surface. For any ellipse, there are two focal points, and one of these is the geocentre of the Earth. Another feature of an elliptical orbit is that there are two other major points. One is where the satellite is furthest from the Earth. This point is known as the apogee. The point where it is closest to the Earth is known as the perigee.

The plane of a satellite orbit is also important. Some may orbit around the equator, whereas others may have different orbits. The angle of inclination of a satellite orbit is shown in Figure 8.2. It is the angle between a line perpendicular to the plane of the orbit and a line passing through the poles. This means that an orbit directly above the equator will have an inclination of 0 degrees (or 180 degrees), and one passing over the poles will have an angle of 90 degrees. Those orbits above the equator are generally called equatorial obits, whilst those above the poles are called polar orbits.

A further feature of any satellite is the angle of elevation above the Earth's surface at a given position on the Earth and a given time. It is very important because the earth station will only be able to maintain contact with the satellite when it is visible. The angle of elevation is the angle at which the satellite appears above the horizontal. If the angle is too small then signals may be obstructed by nearby objects if the antenna is not very high. For those antennas that have an unobstructed view there are still problems with small angles of elevation. The reason is that signals have to travel through more of the earth's atmosphere and are subjected to higher levels of attenuation as a result. An angle of five degrees is generally accepted as the minimum angle for satisfactory operation.

In order that a satellite can be used for communications purposes the ground station must be able to follow it in order to receive its signal, and transmit back to it. Communications will naturally only be possible when it is visible, and dependent upon the orbit it may only be visible for a short period of time. To ensure that communication is possible for the maximum amount of time there are a number of options that can be employed. The first is to use an elliptical orbit where the apogee is above the planned earth station so that the satellite remains visible for the maximum amount of time. Another option is to launch a number of satellites with the same orbit so that when one disappears from view, and communications are lost, another one appears. Generally three satellites are required to maintain almost uninterrupted communication. However the handover from one satellite to the next introduces additional complexity into the system, as well as having a requirement for at least three satellites.

Circular orbits
Circular orbits are classified in a number of ways. Terms such as Low Earth orbit, Geostationary orbit and the like detail distinctive elements of the orbit:

* Low Earth Orbit (LEO: 200 - 1200km above the Earth's surface)
* Medium Earth Orbit (MEO or ICO: 1200 - 35790 km)
* Geosynchronous Orbit (GEO: 35790 km above Earth's surface)
* Geostationary Orbit (GSO)
* High Earth Orbit (HEO: above 35790 km)

The LEO and MEO are used for many types of satellite. As they are relatively close to the Earth's surface they orbit in times much shorter than those higher up. This is because there is a particular velocity required at any given altitude for the gravitational and centrifugal forces to balance. Also the path loss to and from the satellite is much lower in view of the shorter radio paths involved.

As the height of a satellite increases, so the time for the satellite to orbit increases. At a height of 35790 km, it takes 24 hours for the satellite to orbit. This type of orbit is known as a geosynchronous orbit, i.e. it is synchronized with the Earth.

One particular form of geosynchronous orbit is known as a geostationary orbit. In this type of orbit the satellite rotates in the same direction as the rotation of the earth and has a 24 hour period. This means that it revolves at the same angular velocity as the earth and in the same direction and therefore remains in the same position relative to the earth. Geostationary orbits are very popular because once the earth station is set onto the satellite it can remain in the same position, and no tracking is normally necessary. This considerably simplifies the design and construction of the antenna. For direct broadcast satellites it means that people with dishes outside the home do not need to adjust them once they have been directed towards the satellite.

Once in a geostationary orbit, the satellite needs to be kept in its position and not drift. Small rockets are installed on a satellite to ensure that any deviations can be corrected.

The path length to any geostationary satellite is a minimum of 22300 miles. This gives a small but significant delay of 0.24 seconds. For a communications satellite this must be doubled to account for the uplink and downlink times giving virtually half a second. This delay can make telephone conversations rather difficult when satellite links are used. It can also be seen when news reporters as using satellite links. When asked a question from the broadcasters studio, the reporter appears to take some time to answer. This delay is the reason why may long distance links use cables rather than satellites as the delays incurred are far less.

In some applications high earth orbits may be required. For these applications the satellite will take longer than 24 hours to orbit the Earth, and path lengths may become very long resulting in additional delays for the round trip from the Earth toth e satellite and back as well as increasing the levels of path loss.
The choice of the satellite orbit will depend on its applications. While geostationary orbits are popular for applications such as direct broadcasting and for communications satellites, others such as GPS and even those satellites used for mobile phones are much lower.

Applications of Satellite

2007-01-02T15:43:00.000+08:00

There are many applications for satellites in today's world. Ever since the first satellite, Sputnik 1, was launched in 1957, large numbers of satellites have been launched into space to meet a variety of needs. As satellite technology has developed over the years, so ahs the number of applications to which they can be put. Whatever the type of satellite it is necessary to be able to communicate with them, and in view of the large distances, the only feasible technology is radio. As such radio communication is an integral part of any satellite system, whatever its application.

Satellite applications

Astronomical satellites - these satellites are used for the observation of distant stars and other objects in space. Placing an observation point in space removes the unwanted effects of the atmosphere and enables far greater levels of detail to be seen than would be possible on earth where many observatories are placed on mountain tops that experience low levels of cloud. The most famous astronomical satellite is the Hubble Telescope. Although now reaching the end of its life it has enabled scientists to see many things that would otherwise not have been possible. Nevertheless it did suffer some major design setbacks that were only discovered once it was in orbit.

Communications satellites - these satellites possible form the greatest number of satellites that are in orbit. They are used for communicating over large distances. The height of the satellite above the Earth enables the satellites to communicate over vast distances, and thereby overcoming the curvature of the Earth's surface.
Even within the communications field there are a number of sub-categories. Some satellites are used for point to point telecommunications links, others are used for mobile communications, and there are those used for direct broadcast. There are even some satellites used for mobile phone style communications. Even though these satellites did not take the market in the way that was originally expected because terrestrial mobile phone networks spread faster than was originally envisaged, some mobile phone satellite systems still exist.

Earth observation satellites - these satellites are used for observing the earth's surface and as a result they are often termed geographical satellites. Using these satellites it is possible to see many features that are not obvious from the earth's surface, or even at the altitudes at which aircraft fly. Using these earth observation satellites many geographical features have become obvious and they have even been used in mineral search and exploitation.

Navigation satellites - in recent years satellites have been used for accurate navigation. The first system known as GPS (Global Positioning System) was set up by the US DoD and was primarily intended for use as a highly accurate military system. Since then it has been adopted by a huge number of commercial and private users. Small GPS systems are available at costs that are affordable by the individual and are used for car navigation, and they are even being incorporated into phones in a system known as A-GPS (Assisted GPS) to enable accurate location of the phone in case of emergency.
Further systems are planned for the future. The Russian system known as Glonass and the European and Chinese system Galileo are planned for the future.

Reconnaissance satellites - these satellites, are able to see objects on the ground and are accordingly used for military purposes. As such their performance and operation is kept secret and not publicized.

Weather satellites - as the name implies these satellites are used to monitor the weather. They have helped considerably in the forecasting of the weather and have helped provide a much better understanding not only of the underlying phenomena, but also in enabling predictions to be made. A variety of these satellites are in use and include the NOAA series.

There are now many thousands of satellites in orbit around the Earth. Many are in operations, while some that have not yet fallen out of orbit are still circling the Earth. The operational satellites provide many of the services on which we rely today. Without them many of the services which we have come to accept as normal would not be so nearly to achieve by other means.

Satellite facts and information

2007-01-02T15:33:00.000+08:00

Facts about numbers of satellites in orbit
There are over 2500 satellites in orbit around the Earth
There are also over 10 000 man made objects orbiting around the Earth. These include a variety of pieces of satellite debris ranging from panels to disused equipment.

Facts about satellite firsts
The first satellite named Sputnik 1 was launched by the Soviet Union on 4th October 1957. It was a football sized globe that transmitted a "beep beep" sound as it orbited the Earth. The word Sputnik means satellite. It continued transmitting for about 21 days. It was followed four months later by the US satellite Explorer 1 which was launched on 31st January 1958.

Possibly one of the best known satellites was Telstar 1. Built by AT&T it was launched on July 10, 1962, and on the same day live television pictures originating in the United States were received in France.

Facts about satellite orbits
Most communications satellites use what is termed a geostationary orbit. These are at an altitude of, around 22,000 miles and as a result of their speed and the circumference of the orbit they travel round the Earth above the equator in 24 hours. As they travel at the same rate that the Earth rotates, they stay above the same point on the Earth's surface all the time.

In contrast, Low Earth Orbits are just above the Earth's atmosphere and are typically between 100 and 800 miles in altitude. Orbiting at this altitude, an object may only take about 90 minutes to completely circle the Earth, travelling at around 17,000 miles per hour. Low Earth Orbit is used by manned vehicles such as the space shuttle and the International Space Station. It is also used for weather and remote sensing satellites. On a clear night it is usually possible to see with the naked eye several satellites in low earth orbit passing overhear.

Facts about the Global Positioning System (GPS)
The GPS system is run by the US Department of Defense. It consists of 24 operational satellites although there are some extra in orbit as spares in case of catastrophic failure even though each satellite is built to last for ten years. The satellites are named Navstar satellites and each one weighs around 1860 pounds. They are about 17 feet across with the solar panels extended, and they transmit about 50 watts, although the solar panels generate around 700 watts.

The satellites are in one of six orbits. These are in planes that are inclined at approximately 55 degrees to the equatorial plane and there are four satellites in each orbit. The orbits that are roughly 20200 km above the surface of the earth and the satellites travel at a speed of around 14000 km / hour (i.e. about 8500 mph) which means they complete each orbit in roughly 12 hours.

The superheterodyne radio receiver

2006-12-30T12:53:00.000+08:00

The superhet radio or to give it its full name the superheterodyne receiver is one of the most popular forms of receiver in use today. Virtually all broadcast radios, televisions and many more types of receiver use the superhet or superheterodyne principle. First developed at the end of the First World War, with its invention credited to the American Edwin Armstrong, the use of the superhet has grown ever since the concept was first discovered.

Mixing
The idea of the superhet revolves around the process of mixing. Here RF mixers are used to multiply two signals together. (This is not the same as mixers used in audio desks where the signals are added together). When two signals are multiplied together the output is the product of the instantaneous level of the signal at one input and the instantaneous level of the signal at the other input. It is found that the output contains signals at frequencies other than the two input frequencies. New signals are seen at frequencies that are the sum and difference of the two input signals, i.e. if the two input frequencies are f1 and f2, then new signals are seen at frequencies of (f1+f2) and (f1-f2). To take an example, if two signals, one at a frequency of 5 MHz and another at a frequency of 6 MHz are mixed together then new signals at frequencies of 11 MHz and 1 MHz are generated.

The signals generated by mixing or multiplying two signals together

Concept of the superheterodyne receiver
In the superhet or superheterodyne radio, the received signal enters one input of the mixed. A locally generated signal (local oscillator signal) is fed into the other. The result is that new signals are generated. These are applied to a fixed frequency intermediate frequency (IF) amplifier and filter. Any signals that are converted down and then fall within the passband of the IF amplifier will be amplified and passed on to the next stages. Those that fall outside the passband of the IF are rejected. Tuning is accomplished very simply by varying the frequency of the local oscillator. The advantage of this process is that very selective fixed frequency filters can be used and these far out perform any variable frequency ones. They are also normally at a lower frequency than the incoming signal and again this enables their performance to be better and less costly.

To see how this operates in reality take the example of two signals, one at 6 MHz and another at 6.1 MHz. Also take the example of an IF situated at 1 MHz. If the local oscillator is set to 5 MHz, then the two signals generated by the mixer as a result of the 6 MHz signal fall at 1 MHz and 11 MHz. Naturally the 11 MHz signal is rejected, but the one at 1 MHz passes through the IF stages. The signal at 6.1 MHz produces a signal at 1.1 MHz (and 11.1 MHz) and this falls outside bandwidth of the IF so the only signal to pass through the IF is that from the signal on 6 MHz.

The basic concept of the superhet radio

If the local oscillator frequency is moved up by 0.1 MHz to 5.1 MHz then the signal at 6.1 MHz will give rise to a signal at 1 MHz and this will pass through the IF. The signal at 6 MHz will give rise to a signal of 0.9 MHz at the IF and will be rejected. In this way the receiver acts as a variable frequency filter, and tuning is accomplished.

Images
The basic concept of the superheterodyne receiver appears to be fine, but there is a problem. There are two signals that can enter the IF. With the local oscillator set to 5 MHz and with an IF it has already been seen that a signal at 6 MHz mixes with the local oscillator to produce a signal at 1 MHz that will pass through the IF filter. However if a signal at 4 MHz enters the mixer it produces two mix products, namely one at the sum frequency which is 10 MHz, whilst the difference frequency appears at 1 MHz. This would prove to be a problem because it is perfectly possible for two signals on completely different frequencies to enter the IF. The unwanted frequency is known as the image. Fortunately it is possible to place a tuned circuit before the mixer to prevent the signal entering the mixer, or more correctly reduce its level to acceptable value.

Fortunately this tuned circuit does not need to be very sharp. It does not need to reject signals on adjacent channels, but instead it needs to reject signals on the image frequency. These will be separated from the wanted channel by a frequency equal to twice the IF. In other words with an IG at 1 MHz, the image will be 2 MHz away from the wanted frequency.

Using a tuned circuit to remove the image signal

Complete receiver
Having looked at the concepts behind the superheterodyne receiver it is helpful to look at a block diagram of a basic superhet. Signals enter the front end circuitry from the antenna. This contains the front end tuning for the superhet to remove the image signal and often includes an RF amplifier to amplify the signals before they enter the mixer. The level of this amplification is carefully calculated so that it does not overload the mixer when strong signals are present, but enables the signals to be amplified sufficiently to ensure a good signal to noise ratio is achieved.

The tuned and amplified signal then enters one port of the mixer. The local oscillator signal enters the other port. The local oscillator may consist of a variable frequency oscillator that can be tuned by altering the setting on a variable capacitor. Alternatively it may be a frequency synthesizer that will enable greater levels of stability and setting accuracy.

Once the signals leave the mixer they enter the IF stages. These stages contain most of the amplification in the receiver as well as the filtering that enables signals on one frequency to be separated from those on the next. Filters may consist simply of LC tuned transformers providing inter-stage coupling, or they may be much higher performance ceramic or even crystal filters, dependent upon what is required.

Once the signals have passed through the IF stages of the superheterodyne receiver, they need to be demodulated. Different demodulators are required for different types of transmission, and as a result some receivers may have a variety of demodulators that can be switched in to accommodate the different types of transmission that are to be encountered. The output from the demodulator is the recovered audio. This is passed into the audio stages where they are amplified and presented to the headphones or loudspeaker.

Block diagram of a basic superheterodyne receiver

The diagram above shows a very basic version of the superhet or superheterodyne receiver. Many sets these days are far more complicated. Some superhet radios have more than one frequency conversion, and other areas of additional circuitry to provide the required levels of performance. However the basic superheterodyne concept remains the same, using the idea of mixing the incoming signal with a locally generated oscillation to convert the signals to a new frequency.

The Overview Digital Radio Mondiale

2006-12-15T13:28:00.000+08:00

Digital Radio Mondiale (DRM) is set to revolutionise broadcasting on the long, medium and short wave bands. Since the very earliest days of broadcasting these wavebands have been filled with signals that are amplitude modulated. These transmissions are of low audio quality and particularly in recent years there has been a move away from these bands to find higher quality transmissions. Broadcasts in the VHF FM band have received far more listeners with the result that audience figures are dropping for AM broadcasting. Now DAB Digital Radio is available in many countries and this has set new standards in broadcasting. The next stage is to improve the transmissions on the long medium and short wave bands. As the requirements are very different to those experienced on the higher frequencies the DAB standard is not applicable and as a result a totally new system has been developed. Known as DRM it provides many of the improvements that are badly needed along with the flexibility to allow for future developments.

What is DRM?
DRM itself is a consortium of broadcasters, network operators, equipment manufacturers, broadcasting unions, regulatory bodies and other organisations representing 29 countries. It was founded in Guangzhou, China in 1998 and now has its headquarters in Geneva. Now with 82 members, the wide base of its membership has been part of the reason for its success. It has been able to draw on the experience of the membership to ensure that the resulting standard met the requirements, and it has also drawn on the experience gained by the Eureka project that was set up to develop DAB Digital Radio. As a result the new system has come to fruition remarkably swiftly. A preliminary system was designed and tested within a laboratory and this was later extended to include field trials on air to ensure that the new system would successfully meet all the requirements.

The system
When the specification for DRM was being drawn up there were a number of key requirements that needed to meet. The main thrust of the development was to ensure that far greater audio quality could be achieved, but this needed to be achieved whilst keeping the transmissions in a form where they could operate alongside the existing AM transmissions. This meant having the ability for the transmissions to occupy a variety of different bandwidths dependent up the location and frequencies in use. In the Americas a 10 kHz channel spacing is used on the medium wave band whilst in Europe there is a 9 kHz spacing. On the short wave bands a 5 KHz channel spacing has been adopted. It is necessary for the new standard to be able to be compatible with these whilst offering the possibility of other bandwidth options for the future.

Data can also transmitted. Not only does this supply information required for decoding the signal but it also allows data to be transmitted in support of the programme. One particularly useful feature for the short wave bands is that a list of alternative frequencies is transmitted so that listeners can be transferred to better channels very easily as conditions change.

Another advantage of the new system is that it can support what is termed a single frequency network (SFN). This allows a single frequency to be re-used even within the coverage area of the first transmitter without mutual interference. Currently frequencies can only be re-used used outside the coverage area of the first transmitter to avoid interference problems. By using an SFN, far more efficient use can be made of the available channels. With spectrum bandwidth always in short supply, this is another important feature.

DRM transmissions
There are two main elements to the new transmission system. These are the audio coding and the RF modulation used.

The main audio encoding system employs two main techniques. The first is called Advanced Audio Coding (AAC). It is found that the ear does not perceive all the sounds that are heard. A strong sound on one frequency will mask out others close in frequency that may be weaker. AAC, therefore, analyses each section of the spectrum and only encodes those sounds that will be perceived.

However AAC on its own does not provide sufficient compression of the data to enable the transmissions to be contained within the narrow transmission bandwidths required. To provide the additional data compression required a scheme known as Spectral Band Replication (SBR) is employed. This analyses the sounds in the highest octave which are normally from sounds such as percussion instruments of those that are harmonically related to other sounds lower in frequency. It analyses them and sends data to the receiver that will enable them to be reconstituted later.

Data channels
Data to provide the different functions on the transmission is organised into a number of channels that are then applied to the overall modulating signal. The main payload for the signal is known as the Main Service Channel (MSC) and this includes the audio signal data. Two subsidiary channels are also available. These are known as the Fast Access Channel (FAC) that provides the essential data required to fully decode the signal and the Service Description Channel (SDC).

RF Signal
The transmitted signal uses a form of modulation known as Coded Orthogonal Frequency Division Multiplex (COFDM). This form of modulation is being used more frequently is very resilient to many common forms of interference and fading. Its main drawback has been that it requires a significant level of signal processing to extract the data from the carriers and reassemble it in the correct fashion. However signal processing ICs are now sufficiently powerful and at a reasonable cost to make the use of this form of modulation viable. Interestingly COFDM is also used by DAB Digital Radio.

The signal consists of several carriers, across which the data is spread equally. The carriers are spaced equally apart where the spacing is equal to the inverse of the symbol period of the data applied to the carrier. With this spacing it is found that the energy density in the sidebands has nulls or minimum points that correspond with the position of the next carrier. In this way the interference between the nearby carriers is eliminated and they are said to be orthogonal.

Technology for HD Radio

2006-12-12T14:14:00.000+08:00

Digital technology is being applied to many areas of radio communication including radio broadcasting as it offers some significant advantages. While DAB digital radio is becoming established in some areas of the globe, the system that has been chosen for use in the USA is known as HD, or High Definition, Radio. Using HD Radio, will enable high quality audio to be received along with the ability to incorporate many new features and facilities.

The HD Radio system has been developed by iBiquity, and has now been selected by the FCC in the USA. It will take the place of both the existing AM and FM transmissions, and offers many advantages for both listeners and broadcasters alike:

* Improved audio quality - it is claimed that HD Radio broadcasts on the AM bands will be as good as current FM services and those on the FM band will offer CD quality audio.

* Reduced levels of interference. AM transmissions in particular are prone to static pops and bangs as well as high levels of background noise. HD Radio will almost eliminate this.

* Opportunity to use additional data services. By using digital technology, HD Radio provides the opportunity to add data services such as scrolling programme information, song titles, artist names, and much more.

* There is also the possibility of adding more advanced services such as surround sound, multiple audio sources, on-demand audio services, etc.

* Easy transition for broadcasters and listeners. Although new HD Radio receivers are required to receive the new transmissions in their digital format there is considerable re-use of infrastructure and spectrum.

HD Radio basics
HD Radio uses a variety of technologies to enable it to carry digital audio in an acceptable bandwidth and with the new high quality that is required. The transmission uses COFDM combined with specialised codec to compress the audio.

Orthogonal Frequency Division Multiplex (OFDM) is a form of transmission that uses a large number of close spaced carriers that are modulated with low rate data. Normally these signals would be expected to interfere with each other, but by making the signals orthogonal to each another there is no mutual interference. This is achieved by having the carrier spacing equal to the reciprocal of the symbol period.

This means that when the signals are demodulated they will have a whole number of cycles in the symbol period and their contribution will sum to zero - in other words there is no interference contribution. The data to be transmitted is split across all the carriers and this means that by using error correction techniques, if some of the carriers are lost due to multi-path effects, then the data can be reconstructed.

Additionally having data carried at a low rate across all the carriers means that the effects of reflections and inter-symbol interference can be overcome. It also means that single frequency networks, where all transmitters can transmit on the same channel can be implemented. Further information on OFDM can be found on this site under the Cellular telecoms section or by using the Search facility.

One of the requirements for HD Radio was that it would maintain compatibility with existing stations. To achieve this there are two versions; one HD Radio system for AM, and the other for FM.

In what is termed hybrid mode, the AM version has a data rate of 36 kbps for the main audio channel and the version of HD radio for the FM bands carries 96 kbps. In addition to this HD radio can also be used to carry multiple audio channels, and in addition to this secondary channels for services such as weather, traffic and the like may be added. However adding additional channels will reduce the available bandwidth for the primary channel and audio quality may be impaired.

In hybrid mode a radio receiver will first lock onto an analogue signal. If this is possible, then it will try to find a stereo component (FM only) and finally it will endeavour to decode a digital signal. If the digital signal is lost then it will fall back to the analogue signal. The success of this process depends upon the transmitting station being able to synchronise the digital and analogue signals. Often the digitisation process takes a noticeable amount of time and the digital and analogue signals may not be transmitted in time with each other.

Once HD Radio is fully established, the hybrid mode may be removed and at this point no analogue information will be transmitted. However it is envisaged that this will take some time as this can only be viable when very few analogue radios are in use.

Radio Frequencies Digital Audio Broadcasting

2006-12-09T09:34:00.000+08:00

DAB digital radio can be broadcast on a wide number of frequencies. There are both terrestrial and satellite allocations for Digital Audio Broadcasting (DAB). Currently the main frequencies where it is being deployed are within the Band III (Band 3) frequencies. Here a number of channels have been allocated. A complete table of the channels is given below, although in many countries the full number of channels is not available. Within the UK, the DAB multiplexes are being broadcast on channels 11B through to 12D inclusive.

Although it may appear that comparatively few channels are available, each multiplex is able to carry many stations. If high quality audio is required then fewer stations can be accommodated. However it is often possible to accommodate around four or five high quality broadcasts along with several lower quality ones. In addition to this data can also be carried.

Channel Frequency
MHz
5A 174.928
5B 176.640
5C 178.352
5D 180.064
6A 181.936
6B 183.648
6C 185.360
6D 187.072
7A 188.928
7B 190.640
7C 192.352
7D 194.064
8A 195.936
8B 197.648
8C 199.360
8D 201.072
9A 202.928
9B 204.640
9C 206.352
9D 208.064
10A 209.936
10B 211.648
10C 213.360
10D 215.072
11A 216.928
11B 218.640
11C 220.352
11D 222.064
12A 223.936
12B 225.648
12C 227.360
12D 229.072
13A 230.748
13B 232.496
13C 234.208
13D 235.776
13E 237.448
13F 239.200

How digital radio works

2006-12-07T09:32:00.000+08:00

DAB Digital Radio, which is also known as digital audio broadcasting, is an entirely new system for broadcasting and receiving radio stations. As the name indicates signals are broadcast in a digital format to enable CD quality to be achieved. People who have heard DAB digital radio have commented on the significantly better sound quality and "presence" of the new radio system. Also it does not suffer from the multipath effects often experienced on FM transmissions and as the system uses what it known as a single frequency network (SFN) there is no retuning required when moving from one coverage area to the next.

In addition to this many new services can be carried on these digital radio transmissions enabling the new system to be compatible with the 21st century. The digital radio signal carries data alongside the audio, and this enables text and images to be transmitted alongside the audio to enhance the listening experience. In this way it is possible to transmit the title of a track, and a picture of the artist whilst the some music is being transmitted. It is also possible to have news scrolling across the bottom of the screen used on the radio.

DAB digital radio is now well established in many countries around the world from the UK and Europe to Canada, Australia and many other countries. Wit the facilities that digital radio offers it is now being accepted and listeners are switching to these new digital radio transmissions in the areas where they are available.

To produce a digital system that operates satisfactorily under the conditions required for digital radio a large amount of work was undertaken in the development stages. Some existing digital techniques were investigated but it was realised these had significant limitations for this application. One of the major problems was that many receivers would use non-directional antennas and as a result they would pick up reflected signals. These would be delayed sufficiently for the data to become corrupted. Additionally the bandwidth required to accommodate a full stereo signal would need to be reduced to ensure efficient use of the spectrum. The technical standards for digital radio were developed under the auspices of the European Eureka Project 147. This consortium consisted of manufacturers, broadcasters research bodies and network operators.

There are two main areas of the system that are of interest in digital radio: namely the modulation system and the audio digital encoding and compression system.

The encoding and compression system is of paramount importance. For the system to be viable the data rate has to be considerably reduced from that of a standard CD. The digital radio system adopted reduces the data rate down to 128 kbits / sec, a sixth of the bit rate for a similar quality linearly encoded signal. To achieve these reductions the incoming audio signal is carefully analysed. It is found that the ear has a certain threshold of hearing. Below this the signals are not heard. Additionally if a strong sound is present on one frequency then weaker sounds close to it may not be heard because the threshold of hearing is modified. By analysing the incoming audio and only encoding those constituents that the ear will hear the significant reductions can be made. Further reductions in data rate can be achieved by reducing the audio bandwidth. This is implemented on some channels such as those used only for speech.

The other key to the operation of digital radio is the modulation system. Called Coded Orthogonal Frequency Division Multiplex (COFDM) it is a form of spread spectrum modulation that provides the robustness required to prevent reflections and other forms of interference from disrupting reception.

The system uses about 1500 individual carriers that fill around 1.5 MHz of spectrum. The carriers are spaced very close to one another. Interference between the carriers is prevented by making the individual signals orthogonal to each other. This is done by spacing each one by a frequency equal to the data rate being carried. In this way the nulls in the modulation sidebands fall at the position where the next carrier is located. The audio data is then spread across the carriers so that each carrier takes only a small proportion of the data rate. This has the advantage that if interference is encountered in one area then sufficient data is received to reconstitute the required signal. Guard bands are also introduced at the beginning of each symbol, and the combined effect is such that the system is immune to delays consistent with signals 60 km further away than the primary source.

With this level of immunity, the system can operate with other digital radio transmitters operating on the same frequency without any ill effects. This means that it is possible to set up a system where all the transmitters for a network operate on the same frequency. This means that it is possible to set up single frequency networks throughout an area in which a common "multiplex" is used. Even though it may appear that this is a recipe for poorer reception caused by several transmitters using the same frequency, the opposite is actually true. It is found that out of area signals tend to augment the required signal. It also means that small areas of poor coverage can have a small transmitter on exactly the same frequency filling in the hole and further improving reception in adjacent areas.

A further advantage of this digital radio system is that it requires less power than the more traditional transmitters. For example those that carry the main BBC FM networks from the main transmitting sites like Wrotham in the South East of England run at powers of around 100 kW for each of the four main services that are transmitted. The cost of the electricity alone is a significant factor in the BBC's running costs and the power reductions will bring huge savings, not to mention the environmental benefits.

Digital radio band allocations
In the UK a spectrum allocation between 217.5 and 230 MHz has been reserved for digital radio transmissions. This gives a total of seven blocks of 1.55 MHz, each able to carry a multiplex of services. In other countries as well spectrum is being made available. Within Europe spectrum is being made available either in Band III as in the UK or in L band between 1452 and 1467 MHz. The upper part of the band between 1467 and 1492 will be reserved for satellite delivery of digital radio.

Digital radio equipment
One of the main problems with the initial launch of digital radio was the availability of the equipment. A large investment had been required from the equipment manufacturers. The heavy reliance on digital signal processing techniques meant large development programmes were needed to develop the equipment. There were also problems with the fact that the early implementations required high current levels. These solutions would not have been suitable for portable receivers, and for in car and home applications heat dissipation was a problem. Furthermore the multi-chip solutions made the equipment large and bulky as well as making the manufacturing costs high.

Manufacturers soon solved the problem. Specific chip sets for DAB were developed and these enabled costs to be reduced dramatically from the initial ones that were seen so that DAB is no where near as high as it was when compared to FM receivers.

Many people now comment on the significant enhancements that DAB digital radio brings. One typical example was when a friend walked into a shop and noticed the music playing had an increased presence. He assumed it must be DAB, and this was confirmed when he asked. Others have noticed the seamless performance when in a car. None of the intermittent hissing when travelling through a marginal area between the two transmitters.

The Radio Data System ( RDS )

2006-12-04T13:18:00.000+08:00

RDS or Radio Data System is standard on most car radios and hi-fi tuners today. RDS is used on VHF FM radio broadcast transmissions and provides a number of facilities that are of great use to all radio listeners, but particularly to those radio listeners in cars. RDS enables traffic reports to be received more easily, and provides many facilities including enabling the radio station name to be displayed on the radio display.

The system has gained a considerable amount of popularity and is widely used in Europe where it has been established for a number of years.

How RDS Works
RDS operates by adding data to the baseband signal that is used to modulate the radio frequency carrier. The baseband signal consists of a number of components. Firstly there is the mono audio consisting of the left plus right (L+R) component that is transmitted at the normal audio frequencies up to 15 kHz. The stereo difference signal is then amplitude modulated as a double sideband suppressed carrier signal at 38 kHz. A pilot tone at 19 kHz (half the frequency of the stereo difference signal subcarrier) is also transmitted and this is used to enable the receiver demodulator to exactly recreate the 38 kHz subcarrier to decode the stereo difference signal.

The stereo difference signal is above the audio hearing range and as a result it does not detract from the normal mono signal. When adding anything new to a transmission, compatibility must be maintained with existing radios.

The RDS information is placed above the stereo difference signal on a 57 kHz subcarrier as shown. This happens to be three times the stereo pilot tone frequency. For stereo transmissions the RDS subcarrier is locked onto the pilot tone. It can either be in-phase with the third harmonic of the tone, or as in the case of the BBC it can be in quadrature.

The actual subcarrier that is used to carry the information is phase modulated to carry the data. It uses a form of modulation called Quadrature Phase Shift Keying (QPSK). This gives good immunity to data errors caused by noise whilst still allowing the data to be transmitted at a suitable rate. Combined with the fact that the subcarrier operates at a harmonic of the pilot tone, these facts minimise the possibility of interference to the audio signals.

Baseband Coding
The rate at which data is transmitted is 1187.5 bits per second. This is equal to the frequency of the RDS subcarrier divided by 48. By adopting this data rate the decoding circuits to operate synchronously. This reduces problems with spurious signals in the decoding circuits.

Data is transmitted in groups consisting of four blocks. Each block contains a 16 bit information word and a 10 bit check word as shown. This means that with the data rate of 1187.5 bit per second approximately 11.4 groups can be transmitted each second.

A 10 bit check word may seem to be long. However it is very important in view of the poor signal conditions which can exist. This can be particularly true for car or portable radios. The check word enables the radio decoder to detect and correct errors. It also provides a method for synchronisation.

The data groups are structured so that data can be transmitted as efficiently as possible. Different stations will want to transmit different types of data at different times. To cater for this there are a there are a total of 16 different group structures. Their applications are outlined in Figure 3.

Mixing of different types of data within groups is kept to a minimum. However the coding structure is such that messages which need repeating most frequently normally occupy the same position within groups. For example the first block in a group always contains the PI code and PTY and TP are to be found in block 2.

In order that a radio knows how to decode the data correctly, each type of group has to be identified. This function is performed by a four bit code occupying the first four bits in the second block.

Once generated the data is coded onto the subcarrier in a differential format. This allows the data to be decoded correctly whether the signal is inverted or not. When the input data level is "0" the output remains unchanged but when a "1" appears at the input the output changes its state.

With the basic signal generated the spectrum has to be carefully limited. This has to be done to avoid any cross talk in phase locked loop decoders. The power density close to 57 kHz is limited by the encoding each bit as a biphase signal. In addition to this the coded data is passed through a low pass filter.

Facilities
The RDS system offers a wide range of very useful facilities. The most widely publicised one is that of being able to provide travel news. This is available on most local radio stations. All of these stations transmit the TP code to identify that travel messages are flagged by RDS. When the radio is set for travel news it will only tune to stations which carry the TP indication. As the station is about to broadcast a travel announcement the TA code is transmitted. If a CD or cassette is being played then most sets will actually pause the CD or tape and then allow the travel announcement to be heard. In addition to this the volume may also be set slightly higher to allow the announcement to be heard more easily.

Autotuning
RDS brings intelligence into the tuning of a radio. The autotuning facility comes into its own on long journeys when the car moves from the service area of one transmitter to the next. Without RDS the radio has to be manually tuned to the next station. This is not always easy because it is difficult to reliably detect which is the strongest station.

An RDS set will look for the Programme Identification or PI code. A national network will be broadcast from a large number of different transmitters around the country. The station or network eg Radio 4 will have its own PI code. When the radio moves out of the range of one transmitter the radio will seek the strongest signal which has the same PI code, allowing the radio to remain tuned to the same programme.

When radios fitted with RDS store a station frequency, they also store the PI code along side it. This has the advantage that when the radio is turned on in a place outside the coverage area for the transmitter frequency which is stored then the radio will seek the strongest signal which has the correct PI code.

Local radio stations also have a PI code. In view of the local nature of these stations the PI code works slightly differently.

If the station has two or more transmitters then the PI code will operate in the normal way when it is range of these transmitters. However when the radio moves outside this coverage area it will retune to the strongest signal of the same type of station.

The PI code consists of four characters. The first indicates the country of origin and for the UK this is C. The next one indicates the type of coverage. The figure "2" indicates a national station, and the final two characters are the programme reference. For example Radio 3 has the PI code C203 and BBC GLR has C311.

Instant Tuning
It takes a number of seconds for the radio to search for the strongest signal with the correct PI code. During this time the radio would mute itself and the listener would have an annoying gap in listening. To enable the set to tune itself very quickly from one transmission to the next each transmitter broadcasts a short list of frequencies of adjacent transmitters. This vastly reduces the amount of seeking which the radio set has to perform. In addition to this a second front end is often employed to constantly detect the strength of the alternative frequency transmissions. This results in much faster changes in setting - to the extent that the listener should not be able to detect when the radio changes from one transmitter to another.

Another facility associated with tuning is called the Programme Service Name (PS). This enables the set to display the station name. This normally takes a second or two to come up on the display after the station has been tuned in. However it is a most useful facility with the ever-increasing number of stations on the air

New Facilities
A new feature which has been added to RDS is called Enhanced Other Networks (EON). This allows the set to listen to one station like a national network, but still be interrupted by travel news from a local radio station. This feature even allows announcements to be heard whilst travelling in silence or listening to a tape.

EON requires a large amount of co-ordination between the different stations. To achieve this, the BBC have a central computer specifically for this purpose. When a local radio station is about to transmit a traffic message the fact is flagged to the computer. In turn this directs the relevant national radio transmitters to indicate this fact, thereby enabling the radios to change frequency to the local radio station to receive the message. Once the message is complete the radio will return to its original station.

EON is relatively new and the first sets to have it included only appeared in 1991. Although it is being introduced on more sets, the majority still does not have it. However with manufacturers constantly bringing new sets onto the market EON should be included on far more sets in a year or two.

Broadcast VHF FM

2006-12-02T08:38:00.000+08:00

When broadcasting first started in the 1920s amplitude modulation was used because it was the obvious and the easiest way to transmit sound. However as radio technology developed its shortcomings became more obvious and the quest for higher quality transmissions lead to the introduction of wideband frequency modulation. Although the first commercial stations were set up in the USA around 1939, it was not until the 1950s that FM started to become really accepted. It was in 1954 that the BBC announced their intention to start FM broadcasting. Now VHF FM is the accepted medium for high quality transmissions, and stations that use AM on the medium and long wave bands have to work hard to retain listeners who prefer the higher quality of VHF FM.

What is FM?
Amplitude modulation, which is the simplest and most obvious form of modulation varies the amplitude of the carrier so that it carries the sound information. Frequency modulation is slightly more subtle and as the name indicates it varies the frequency of the carrier in line with the variations in the modulating audio signal. This as the modulating waveform increases in voltage, so the carrier will swing in one direction and as it decreases it will move in the other direction.

One of the important factors of FM is the degree by which the carrier changes. This deviation is usually expressed in kilohertz variation either side of the centre (no modulation) frequency. Typically a signal may have a deviation of +/- 3kHz if it varies up and down by 3 kHz. There are two main categories on FM. The first is called narrow band FM, and this is where the deviation is relatively small, possibly 5 kHz. This type of transmission is used mainly by VHF / UHF point to point mobile communications. To appreciate the full benefits of FM, wideband FM is used having a greater level of deviation. The standard for broadcasting is +/- 75 kHz. To fully accommodate these transmissions a bandwidth of 200 kHz is used.

The advantage of FM is that as the modulation is carried solely as frequency variations, much noise, which appears mainly as amplitude variations can be discarded in the receiver. Accordingly it is possible to achieve much better noise performance using FM. The upper audio frequency limit is generally taken as 15 kHz for these transmissions. This is quite adequate for most high quality transmissions.

Pre-emphasis and de-emphasis
One of the problems with the high quality VHF FM transmissions is that the increased audio bandwidth means that background noise can often be perceived. Even then it is considerably better than that obtained using and AM system. It is particularly noticeable towards the treble end of the audio spectrum, where it can be heard as a background hiss. To overcome this it is possible to increase the level of the treble frequencies at the transmitter. At the receiver they are correspondingly attenuated to restore the balance. This also has the effect of reducing the treble background hiss which is generated in the receiver. The process of increasing the treble signals is called pre-emphasis, and reducing the in the receiver is called de-emphasis. The rate of pre-emphasis and de-emphasis is expressed as a time constant. It is the time constant of the capacitor-resistor network used to give the required level of change. In the UK, Europe and Australia the time constant is 50 uS whereas in North America it is 75 uS.

Stereo
In recent years stereo transmission has become an accepted part of VHF FM transmissions. The system that is used maintains compatibility with mono only receivers without any noticeable degradation in performance. The system that is used is quite straightforward.

A stereo signal consists of two channels that can be labelled L and R, (Left and Right), providing one channel for each of the two speakers that are needed. An ordinary mono signal consists of the summation of the two channels, i.e. L + R, and this can be transmitted in the normal way. If a signal containing the difference between the left and right channels, i.e. L - R is transmitted then it is possible to reconstitute the left only and right only signals. Adding the sum and difference signals, i.e. (L + R) + (L - R) gives 2L, i.e. the left signal, and subtracting the two signal, i.e. (L + R) - (L - R) gives 2R, i.e. the right signal. This can be achieved relatively simply by adding and subtracting the two signals electronically. It only remains to find a method of transmitting the stereo difference signal in a way that does not affect any mono receivers.

This is achieved by transmitting the difference signal above the audio range. It is amplitude modulated onto a 38 kHz subcarrier. Both the upper and lower sidebands are retained, but the 38 kHz subcarrier itself is suppressed to give a double sideband signal above the normal audio bandwidth as shown below. This whole of the baseband is used to frequency modulate the final radio frequency carrier. It is the baseband signal that is regenerated after the signal is demodulated in the receiver.

To regenerate the 38 kHz subcarrier, a 19 kHz pilot tone is transmitted. The frequency of this is doubled in the receiver to give the required 38 kHz signal to demodulate the double sideband stereo difference signal.

The presence of the pilot tone is also used to detect whether a stereo signal is being transmitted. If it is not present the stereo reconstituting circuitry is turned off. However when it is present the stereo signal can be reconstituted.

To generate the stereo signal, a system similar to that shown in Fig. 8.5 is used. The left and right signals enter the encoder where they are passed through a circuit to add the required pre-emphasis. After this they are passed into a matrix circuit. This adds and subtracts the two signals to provide the L + R and L - R signals. The L + R signal is passed straight into the final summation circuit to be transmitted as the ordinary mono audio. The difference L - R signal is passed into a balanced modulator to give the double sideband suppressed carrier signal centred on 38 kHz. This is passed into the final summation circuit as the stereo difference signal. The other signal entering the balanced modulator is a 38 kHz signal which has been obtained by doubling the frequency of the 19 kHz pilot tone. The pilot tone itself is also passed into the final summation circuit. The final modulating signal consisting of the L + R mono signal, 19 kHz pilot tone, and the L - R difference signal based around 38 kHz is then used to frequency modulate the radio frequency carrier before being transmitted.

Reception of a stereo signal is very much the reverse of the transmission. A mono radio receiving a stereo transmission will only respond to the L + R signal. The other components being above 15 kHz are above the audio range, and in any case they will be suppressed by the de-emphasis circuitry.

For stereo receivers the baseband signal consisting of the stereo sum signal (L+R) and the difference signal (L-R) centred around 38 kHz and the pilot 19kHz tone are obtained directly from the FM demodulator. The decoder then extracts the Left only and Right only signals.

The block diagram of one type of decoder is shown below. Although this is not the only method which can be used it shows the basic processes that are required. The signal is first separated into its three constituents. The L + R mono signal between 0 and 15 kHz, the pilot tone at 19 kHz, and the stereo difference signal situated between 23 and 53 kHz. First the pilot tone at 19 kHz is doubled in frequency to 38 kHz. It is then fed into a mixer with the stereo difference signal to give the L - R signal at audio frequencies. Once the L + R and L - R signals are available they enter a matrix where they are added and subtracted to regenerate the L and R signals. At this point both signals are amplified separately in the normal way in a stereo amplifier before being converted into sound by loudspeakers or headphones.

Radio Broadcast Technology

2006-12-02T08:34:00.000+08:00

Radio broadcasting is an established use of radio technology. The first organised broadcasts taking place in the 1920s. Now there are many radio stations broadcasting all over the world using a variety of differernt types of transmission. Today, radio broadcast equipment from transmitters and receivers to antennas, studios and relay links are widely available, although with the new standards for transmission including DAB Digital Radio and DRM, new equipment is required. Nevertheless AM as well as FM with its RDS capability are still the most widely used.

VHF FM broadcasting
VHF FM is the most widely used form of broadcasting in areas of the world where the population is relatively high. Its bandwidth enables it to carry high quality transmissions, stereo, and other services such as RDS.
Broadcast VHF FM
RDS - Radio Data Service

Digital Audio Broadcasting (DAB)
DAB digital radio is now widely deployed in many countries around the globe, and now that the cost of radios has fallen, listener figures are rising. Although not available in many countries, it is certainly making a significant impact where it has been deployed, adding more flexibility and the possibility of near CD quality.
DAB digital radio
DAB digital radio Band III channel numbers and frequencies
HD Radio - the new digital radio system for the USA.

Digital Radio Mondiale
While DAB digital radio is focussed at bands at VHF and above there have been developments for the lower frequencies. Digital Radio Mondiale (DRM) is now being deployed and is a replacement for the amplitude modulation (AM) transmissions that have been on the airwaves for over 100 years. The new technology has now been extended for use up to 100 MHz and with many developments under way radios should soon be available.
Digital Radio Mondiale (DRM) - the new standard to replace AM broadcasting

Overview of TETRA Private Mobile Radio (PMR)

2006-11-29T09:11:00.000+08:00

TETRA is a modern standard for digital Private Mobile Radio (PMR) and Public Access Mobile Radio (PAMR). It offers many advantages including flexibility, security, ease of use and offers fast call set-up times. This makes it an ideal choice for many business communications requirements.

The name TETRA stands for TErrestrial Trunked RAdio. Aimed at a variety of users including the police, ambulance and fire services, it is equally applicable for utilities, public access, fleet management, transport services, and many other users. It offers the advantages of digital radio whilst still maintaining the advantages of a PMR system.

Tetra radio beginnings
Work started on the development of the TETRA standards in 1990 and has relied on the support of the European Commission and the ETSI members. Experience gained in the development of the highly successful GSM cellular radio standard, as well as experience from the development and use of trunked radio systems has also been used to fashion the TETRA standard. In addition to this the process has gained from the co-operation of manufacturers, users, operators and industry experts. With this combined expertise the first standards were ready in 1995 to enable manufacturers to design their equipment to interoperate successfully.

Tetra radio features
TETRA radio offers many new and valuable features. These include a fast call set-up time, which is a particularly important requirement for the emergency services. It also has excellent group communication support, direct mode operation between individual radios, packet data and circuit data transfer services, better economy of frequency spectrum use than the previous PMR radio systems and in addition to this it provides advanced security features. The system also supports a number of other features including call hold, call barring, call diversion, and ambience listening.

The TETRA radio system uses Time Division Multiple Access (TDMA) technology with 4 user channels on one radio carrier and 25 kHz spacing between carriers. This makes it inherently more efficient than its predecessors in the way that it uses the frequency spectrum. Data can be transmitted at 7.2 kbits per second for a single channel. This can be increased four fold to 28.8 kbits per second when multi-slot operation is employed.

For emergency services in Europe the frequency bands 380-383 MHz and 390-393 MHz have been allocated. These bands can be expanded to cover all or part of the spectrum from 383-395 MHz and 393-395 MHz should this be needed. For civil systems in Europe the frequency bands 410-430 MHz, 870-876 MHz / 915-921 MHz, 450-470 MHz, 385-390 MHz / 395-399,9 MHz, have been allocated.

TETRA radio trunking facility provides a pooling of all radio channels that are then allocated on demand to individual users, in both voice and data modes. By the provision of national and multi-national networks, national and international roaming can be supported, the user being in constant communication. TETRA supports point-to-point, and point-to-multipoint communications both by the use of the TETRA infrastructure and by the use of Direct Mode without infrastructure.

In addition to this it is possible for TETRA radio to operate in a secure format. The digital data can be encrypted before transmissions, making the system inherently secure. This may be required for some covert operations or for the police services.

TETRA radio operation
There are three different modes in which TETRA can be run:

* Voice plus Data (V+D)

* Direct Mode Operation (DMO)

* Packet Data Optimised (PDO)

The most commonly used mode is V+D. This mode allows switching between speech and data transmissions, and can even carry both by using different slots in the same channel. Full duplex is supported with base station and mobile radio units frequencies normally being offset by about 10 MHz to enable interference levels between the transmitter and receiver in the station to be reduced to an acceptable level.

DMO is used for direct communication between two mobile units and supports both voice and data, however full duplex is not supported in this mode. Only simplex is used. This is particularly useful as it allows the mobile stations to communicate with each other even when they are outside the range of the base station.

The third mode, PDO is optimised for data only transmissions. It has been devised with the idea that much higher volumes of data will be needed in the future and it is anticipated that further developments will be built upon this standard.

Data structures
TETRA radio uses TDMA techniques. This enables much greater spectrum efficiency than was possible with previous PMR systems because it allows several users to share a single frequency. As the speech is digitised, both voice and data are transmitted digitally and multiplexed into the four slots on each channel. Digitisation of the speech is accomplished using a system that enables the data to be transmitted at a rate of only 4.567 kbits/second. This low data rate can be achieved because the process that is used takes into account the fact that the waveform is human speech rather than any varying waveform. The digitisation process also has the advantage that it renders the transmission secure from casual listeners. For greater levels of security that might be required by the police or other similar organisations it is possible to encrypt the data. This would be achieved by using an additional security or encryption module.

The data transmitted by the base station has to allow room for the control data. This is achieved by splitting what is termed a multiframe lasting 1.02 seconds into 18 frames and allowing the control data to be transmitted every 18th frame. Each frame is then split into four time slots. A frame lasts 56.667 mS. Each time slot then takes up 14.167 mS. Of the 14.167mS only 14 milliseconds is used. The remaining time is required for the transmitter to ramp up and down. The data structure has a length of 255 symbols or 510 modulation bits. It consists of a start sequence that is followed by 216 bits of scrambled data, a sequence of 52 bits of what is termed a training sequence. A further 216 bits of scrambled data follows and then the stream is completed by a stop sequence. The training sequence in the middle of the data is required to allow the receiver to adjust its equaliser for optimum reception of the whole message.

The data is modulated onto the carrier using differential quaternary phase shift keying. This modulation method shifts the phase of the RF carrier in steps of ± pi /4 or ±3 pi /4 depending upon the data to be transmitted. Once generated the RF signal is filtered to remove any sidebands that extend out beyond the allotted bandwidth. These are generated by the sharp transitions in the digital data. A form of filter with a root raised cosine response and a roll off factor of 0.35 is used. Similarly the incoming signal is filtered in the same way to aid recovery of the data.

Additionally, TETRA radio uses error tolerant modulation and encoding formats. The data is prepared with redundant information that can be used to provide error detection and correction. The transmitter of each mobile station is only active during the time slot that the system assigns it to use. As a result the data is transmitted in bursts. The fact that the transmitter is only active for part of the time has the advantage that the drain on the battery of the mobile station is not as great as if the transmitter was radiating a signal continuously. The base station however normally radiates continuously as it has many mobile stations to service.

One important feature of TETRA is that the call set up time is short. It occurs in less than 300 mS and can be as little as 150 mS when operating in DMO. This is much shorter than the time it takes for a standard cellular telecommunications system to connect. This is very important for the emergency services where time delays can be very critical.

Further TETRA radio developments
While TETRA radio is a major improvement over the previous PMR systems in operation, additional data capacity is always needed. In view of the higher data capabilities now being offered by the cellular services, the TETRA radio standard is being updated to enable it to keep pace with other comparable technologies. In this way, TETRA will be able to offer commercial users the advantages of a PMR service alongside the data capabilities of a cellular network.

PMR Trunking using MPT1327

2006-11-28T09:02:00.000+08:00

A trunked version of the Private Mobile Radio (PMR) concept that is defined under the standard MPT 1327 (MPT1327) is widely used and provides significant advantages over the simpler single station systems that are in use. MPT1327 enables stations to communicate over wider areas as well as having additional facilities.

In view of the very high cost of setting up trunked networks, they are normally run by large leasing companies or consortia that provide a service to a large number of users. In view of the wider areas covered by these networks and the greater complexity, equipment has to be standardised so that suppliers can manufacture in higher volumes and thereby reduce costs to acceptable levels. Most trunked radio systems follow the MPT1327 format.

To implement trunked PMR a network of stations is set up. These stations are linked generally using land lines, although optical fibres and point to point radio are also used. In this way the different base stations are able to communicate with each other.

In order to be able to carry the audio information and also run the variety of organisational tasks that are needed the system requires different types of channel to be available. These are the control channels of which there is one in each direction for each base station or Trunking System Controller (TSC).

A number of different control channels are used so that adjacent base stations do not interfere with one another, and the mobile stations scan the different channels to locate the strongest control channel signal. In addition to this there are the traffic channels. The specification supports up to 1024 different traffic channels to be used. In this way a base station can support a large number of different mobile stations that are communicating at the same time. However for small systems with only a few channels, the control channel may also act as a non-dedicated traffic channel.

The control channels use signalling at 1200 bits per second with fast Frequency Shift Keying (FFSK) subcarrier modulation. It is designed for use by two-frequency half duplex mobile radio units and a full duplex TSC.

For successful operation it is essential that the system knows where the mobiles are located so that calls can be routed trough to them. This is achieved by base stations polling the mobile stations using the control channel.

To make an outgoing call the mobile transmits a request to the base station as requested in the control channel data stream from the base station. The mobile transmits its own code along with that of the destination of the call, either another mobile or a control office. The control software and circuitry within the base station and the central control processing area for the network sets up the network so that a channel is allocated for the audio (the traffic channel). It also sets up the switching in the network to route the call to the required destination.

To enable the mobile station to receive a call, it is paged via the incoming control channel data stream to indicate that there is an incoming call. Channels are allocated and switching set up to provide the correct routing for the call.

There is no method to "handover" the mobile from one base station to the next if it moves out of range of the base station through which a call is being made. In this way the system is not a form of cellular telephone. It is therefore necessary for the mobile station to remain within the service area of the base station through which any calls are being made.

The control channel signalling structure has to be defined so that all mobiles know what to expect and what data is being sent. Signalling on the forward control channel is nominally continuous with each slot comprising 64 bit code words. The first type is the Control Channel System Codeword (CSCC). This identifies the system to the mobile radio units and also provides synchronisation for the following address codeword. As mentioned the second type of word is the address codeword. It is the first codeword of any message and it defines the nature of the message. It is possible to send data over the control channel. When this occurs, botht he CSCC and the address codewords are displaced with the data appended to the address codeword. The mobile radio unit data structure is somewhat simpler. It consists fundamentally of synchronism bits followed by the address codeword.

There are a number of different types of control channel messages that can be sent by the base station to the mobiles:

Aloha messages -- Sent by the base station to invite and mobile stations to access the system.

Requests -- Sent by radio units to request a call to be set up.

"Ahoy" messages -- Sent by the base station to demand a response from a particular radio unit. This may be sent to request the radio unit to send his unique identifier to ensure it should be taking traffic through the base station.

Acknowledgements -- These are sent by both the base stations and the mobile radio units to acknowledge the data sent.

Go to channel messages -- These messages instruct a particular mobile radio unit to move to the allocated traffic channel.

Single address messages -- These are sent only by the mobile radio units.

Short data messages -- These may be sent by either the base station or the mobile radio unit.

Miscellaneous messages -- Sent by the base station for control applications.

One of the problems encountered by mobile signalling systems is that of clashes when two or more mobile radio units try to transmit at the same time on the control channel. This factor is recognised by the system and is overcome by a random access protocol that is employed. This operates by the base station transmitting a synchronisation message inviting the mobile radio units to send their random access message. The message from the base station contains a parameter that indicates the number of timeslots that are available for access. The mobile radio unit will randomly select a slot in which to transmit its request but if a message is already in progress then it will send its access message in the next available slot. If this is not successful then it will wait until the process is initiated again.

Although the data is transmitted as digital information, the audio or voice channels for the system are analogue, employing FM. However some work has been carried out to develop completely digital systems. The main systems are by Motorola, by Ericsson (EDACS) and Johnson (LTR). These systems have not gained such widespread acceptance.

Overview of Private Mobile Radio

2006-11-27T09:32:00.000+08:00

Private Mobile Radio (PMR) or as it is sometimes called Professional Mobile Radio is widely used for businesses as a very convenient way of communicating. The basic concept has been in use for many years and was firmly established prior to the introduction of the first cell phone systems, although systems including MPT1327 that provide trunking and TETRA enable far greater facilities.

The first PMR systems were initially set up to enable a set of mobile business users to maintain contact with a base. Organisations such as taxi firms, utility workers and the like all used these systems as they enabled them to maintain contact with their office. Additionally the emergency services used their own systems.

Initially the systems consisted of a base station with a number of mobile stations. Communication used a single frequency, with simplex push to talk transmissions. As pressure rose on the frequency allocations, often frequencies had to be shared. As the systems almost invariably used frequency modulation, squelch was employed so that the audio from the received was only switched on when a signal was present. Developments of this known as DTMF (dual tone multiple frequency) and CTCSS (continuous tone, coded squelch system) were used to enable only the required users to hear the call.

These systems were only able to communicate over relatively short distances. They used a single central base station to communicate with all the mobile stations. This considerably reduced their coverage area. To overcome this a system known as trunking was devised whereby several transmitters could be used and the signal was “trunked” to the correct station. Several systems are available for this but the one that has gained by far the widest use is specified as MPT 1327.

All the standards mentioned so far are analogue systems. The cellular telecommunications industry moved to digital technology to provide improved efficiency of spectrum usage along with a variety of new facilities. So too did the PMR industry with the induction of a system known as TETRA. . Originally the letters stood for Trans European Trunked RAdio, but as the system is now being used beyond Europe the abbreviation now stands for TErrestrial Trunked RAdio. This system provided a far more flexible service along with all the other advantages of using a digital system.

Overview of the DMB system, and in particular the T-DMB version to be used for mobile video broadcasts

2006-11-21T15:01:00.000+08:00

Digital Multimedia broadcasting, DMB is based on the Eureka 147 Digital Audio Broadcast or DAB system that is widely deployed in the UK and many other countries around the world for audio broadcasting. One of the advantages of using DMB is that it can be rolled out and used without much modification for mobile video applications, simply increasing the level of error correction to cope with the mobile environment.

In view of the different broadcasting platforms that could be used account needs to be taken of this. Eureka 147 allows for broadcasts both from terrestrial transmitters and from satellite based transmitters. For DMB both platforms are possible, but in view of the differing platforms and transmission requirements there would need to be some modifications between the two systems. For terrestrial based transmissions a flavour of the system designated as T-DMB (Terrestrial Digital Multimedia Broadcasting) is used, whereas for satellite broadcasting S-DMB (Satellite Digital Multimedia Broadcasting) is used.

RF signal characteristics
Like many other broadcasting systems, DMB and DAB use a form of transmission known as Orthogonal Frequency Division Multiplex (OFDM). This has been adopted because of its high data capacity and suitability for applications such as broadcasting. It also offers a high resilience to interference, can tolerate multi-path effects and is able to offer the possibility of a single frequency network, SFN.

DMB format
The transmissions for the form of DMB being deployed in many countries occupy approximately 1.5 MHz bandwidth and for the VHF broadcasts the transmission contains 1536 Carriers. However it is possible to use a variety of modes:

* 2K mode 1536 carriers

* 1K mode 768 carriers

* 0.5K mode 384 carriers

* 0.25K mode 192 carriers

Frequency allocations
It would be possible to utilise the DAB transmission system within the UK for DMB, however much of the capacity has been taken up, although some reserve capacity has been retained for future data transmissions of which DMB could be part.

A more likely solution for DMB is to use frequencies within the L-Band DAB allocation (1452 - 1467.5 MHz). This might be possible in some countries where the use of this broadcasting allocation could be used for this purpose with little legislation.

Using a new band it will not only be possible to use smaller antennas, an important element for mobile phones and PDAs, but it will also be possible to tailor the transmission to accommodate the Doppler shifts likely to be encountered by small mobile devices. This can be achieved by reducing the number of carriers. Despite the carrier number reduction, the maximum data rate of 1.152 Mbps is still retained. The drawback of using the L band frequencies is that they would require a much higher density of transmitters to provide the required coverage.

Battery consumption
One of the major requirements for any mobile video system such as DMB is that it shall not place a major load on the battery of the handheld device. With user expectations requiring that battery life shall be several days between recharges, this is a major consideration. While battery technology is improving, and IC technology has enabled current consumption of chips to be reduced, the basic technology can also play its part.

DMB is also ideally suited to the delivery of material to handheld devices. DAB inherently includes a technique known as time slicing by using an effectively using a Time Division Multiplexing delivery method. In this way the receiver is only switched on when it is required, thereby saving battery power.

It remains to be seen whether DMB or DVB-H will be the major standard that is adopted for mobile video. Some indicate that both schemes may be used in different countries around the world Accordingly many chip manufacturers whoa re addressing this market are catering for both schemes and developing systems that will be able to switch between the variety of bands that will be used around the globe.

In addition to this DMB trials are well advanced, particularly in Korea where it appears DMB will be adopted. For other countries, it remains to be seen what happens.

Overview of the DVB-H system to be used for mobile video broadcasts

2006-11-18T08:09:00.000+08:00

DVB-H or Digital Video Broadcast - Handheld, is one of the major systems to be used for mobile video and television for cellular phones and handsets. DVB-H has been developed from the DVB-T (Terrestrial) television standard that is used in many countries around the globe including much of Europe including the UK, and also other countries including the USA. The DVB-T standard has been shown to be very robust and in view of its widespread acceptance it forms a good platform for further development for handheld applications.

DVB-H development requirements
The environment for handheld devices is considerably different to that experienced by most televisions. Normally domestic televisions have good directional antenna systems and in addition to this the reception conditions are fairly constant. Additionally most televisions receiving DVB-T will be powered by mains supplies. As a result current consumption is not a major issue.

The conditions for handheld receivers are very different. In the first instance the antennas will be particularly poor because they will need to be small, and integrated into the handset in such a way that they either appear fashionable, or they are not visible. Additionally they will obviously be mobile, and this will entail receiving signals in a variety locations, many of which will not be particularly suitable for video reception. Not only will be signal be subject to considerable signal variations and multi-path effects, but it may also experience high levels of interference. Also some difficulties are presented by the fact that the handset could be in a vehicle and actually on the move. The operation of DVB-H has to be sufficiently robust to accommodate all these requirements.

"Note on multi-path effects:

Multi-path effects occur when signals reach the receiver via several different paths from the transmitter. This occurs because the signals leave the transmitter in a variety of directions - typically the transmitter may have an omni-directional radiation pattern so that it radiates signals equally in all directions. Accordingly some of the signal may travel directly to the receiver in what is termed the direct path, but some of the radiated may be reflected off a nearby hill, building or other object. In fact the received signal will consist of components reaching the receiver from the transmitter via a large number of paths. As the path length travelled by each of these components will be slightly different, each component will arrive at a slightly different time. If there are significant differences, this can cause the data being transmitted to be corrupted under some circumstances, although many modern receiver technologies can accommodate this and use the different signals travelling over different paths to reinforce one another."

While DVB-T proved to be remarkably robust under many circumstances, one of the major problems was that of current consumption. Battery life for handsets is a major concern where users anticipated the life between charges will be several days.

Operation of DVB-H
The DVB-H standard has been adopted by ETSI, European Telecom Standards Institute, and in this way the system can be truly international, and this will prevent compatibility problems caused by different countries and operators using different variants of the same system. The documents for the physical layer were ratified in 2004, with the upper layers defined in 2005.

DVB-H (Digital Video Broadcast Handheld) is based on the very successful DVB-T (Digital Video Broadcast Terrestrial) standard that is now used in many countries for domestic digital television broadcasts. DVB-H has taken the basic standard and adapted so that it is suitable for use in a mobile environment, particularly with the electronics incorporated into a mobile phone.

The DVB-H standard like DVB-T uses a form of transmission called Orthogonal Frequency Division Multiplex (OFDM). This has been adopted because of its high data capacity and suitability for applications such as broadcasting. It also offers a high resilience to interference, can tolerate multi-path effects and is able to offer the possibility of a single frequency network, SFN.

"Note on OFDM:

Orthogonal Frequency Division Multiplex (OFDM) is a form of transmission that uses a large number of close spaced carriers that are modulated with low rate data. Normally these signals would be expected to interfere with each other, but by making the signals orthogonal to each another there is no mutual interference. This is achieved by having the carrier spacing equal to the reciprocal of the symbol period. This means that when the signals are demodulated they will have a whole number of cycles in the symbol period and their contribution will sum to zero - in other words there is no interference contribution. The data to be transmitted is split across all the carriers and this means that by using error correction techniques, if some of the carriers are lost due to multi-path effects, then the data can be reconstructed. Additionally having data carried at a low rate across all the carriers means that the effects of reflections and inter-symbol interference can be overcome. It also means that single frequency networks, where all transmitters can transmit on the same channel can be implemented. Further information on OFDM can be found on this site under the Cellular telecoms section or by using the Search facility."

There are a variety of modes in which the DVB-H signal can be configured. These are conform to the same concepts as those used by DVB-T. These are 2K, 4K, and 8K modes, each having a different number of carriers as defined in the table below. The 4K mode is a further introduction beyond that which is available for DVB-T.

Parameter 2K mode 4K mode 8K mode
Number of active carriers 1705 3409 6817
Number of data carriers 1512 3024 6048
Individual carrier spacing 4464 Hz 2232 Hz 1116 Hz
Channel width 7.61 MHz 7.61 MHz 7.61 MHz

Signal parameters for DVB-H OFDM Signal (8MHz Channel)

The different modes balance the different requirements for network design, trading mobility for single frequency network size, with the 4K mode being that which is expected to be most widely used.

The standard will support a variety of different types of modulation within the OFDM signal. QPSK (Quadrature Phase Shift Keying), 16QAM (16 point Quadrature Amplitude Modulation), and 64QAM (64 point Quadrature Amplitude Modulation) will all be supported, chipsets being able to detect the modulation and receive the incoming signal. The choice of modulation is again a balance, QPSK offering the best reception under low signal and high noise conditions, but offering the lowest data rate. 64QAM offers the highest data rate, but requires the highest signal level to provide sufficiently error free reception.

Time slicing
One of the key requirements for any mobile TV system is that it should not give rise to undue battery drain. Mobile handset users are used to battery life times extending over several days, and although battery technology is improving, the basic mobile TV technology should ensure that battery drain is minimised.

There is a module within the standard and hence the software that enables the receiver to decode only the required service and shut off during the other service bits. It operates in such a way that it enables the receiver power consumption to be reduced while also offering an uninterrupted service for the required functions.

The time slicing elements of DVB-H enable the power consumption of the mobile TV receiver to be reduced by 90% when compared to a system not using this technique. Although the receiver will add some additional power drain on the battery, this will not be nearly as much as it would have been had the TV reception scheme not employed the time slicing techniques.

Interleaving
Interleaving is a technique where sequential data words or packets are spread across several transmitted data bursts. In this way, if one transmitted burst or group is lost as a result of noise or some other drop-out, then only a small proportion of the data in each original word or packet is lost and it can be reconstructed using the error detection and correction techniques employed.

Further levels of interleaving have been introduced into DVB-H beyond those used for DVB-T. The basic mode of interleaving used on DVB-T and which is also available for DVB-H is a native interleaver that interleaves bits over one OFDM symbol. However DVB-H provides a more in-depth interleaver that interleaves bits over two OFDM symbols (for the 4K mode) and four bits (for the 2K mode).

Using the in-depth interleaver enables the noise resilience performance of the 2K and 4K modes to be brought up to the performance of the 8K mode and it also improves the robustness of the reception of the transmissions in a mobile environment.

MPE-FEC
In view of the particularly difficult reception conditions that may occur in the mobile environment, further error correction schemes are included. A scheme known as MPE-FEC provides additional error correction to that applied in the physical layer by the interleaving. Tjis is a forward error correction scheme that is applied to the transmitted data and after reception and demodulation, allows the errors to be detected and corrected.

Compatibility with DVB-T
DVB-H is a development of DVB-T and as a result it shares many common components. It has also been designed so that it can be used in 6, 7, and 8 MHz channel schemes although the 8MHz scheme will be the most widely used. There is also a 5MHz option that may be used for non-broadcast applications.

In view of the similarities between DVB-H and DVB-T it is possible for both forms of transmission to exist together on the same multiplex. In this way a broadcaster may choose to run two DVB-T services and one DVB-H service on the same multiplex. This feature may be particularly attractive in the early days of DVB-H when separate spectrum is not available.

DVB-H has been used in a number of trials and appear to perform well. It ahs support from a number of the major industry players and is likely to achieve a considerable degree of acceptance world-wide. Accordingly it is likely to be one of the major standards, if not the major standard used for mobile video.

Video for Mobile Phones

2006-11-17T08:38:00.000+08:00

Mobile video for cell phones promises to be a major force in the broadcasting and cellular industries over the coming years. With the functionality in phones increasing, along with people's expectations, placing mobile video or TV into a phone enables its use to be maximized.

Advantages of broadcast
The concept of broadcasting offers many advantages for mobile video applications. Using the mobile phone infrastructure has many advantages it is what may be termed a one to one communications system. However the costs of downloading videos will need to be paid by the user, and therefore may be large. If a broadcast style model is used, which may be thought of as a one to many communications link, then the delivery costs are much lower. The disadvantage is that the level of choice is reduced to what is being broadcast and there is no interactive operation. Nevertheless it is this business model that looks the more attractive and the one that will succeed for video and general mobile content. The high data rates of 3G being reserved for content such as data downloads, data communications, video conferencing and the like.

Business models
The exact way in which mobile video will be implemented as far as revenue is concerned will depend on the operators. There may be subscription services and there may also be services that are supported by advertising. The huge advantage that placing video onto mobile phones is that they are an accessory that is already in people's pockets. It is then possible to extend their functionality to include video, and this means that another unit is not required for this functionality. Additionally the billing infrastructure is already in place, so this too can be extended without the need to start again.

It is possible that the mobile video transmissions could also be used for other services. These could include traffic and weather reports that could be broadcast in the background and brought up as required on the mobile. Additionally they could be used for software upgrades. In this way mobiles could be upgraded online, and new features added if necessary.

Spectrum considerations
One of the big issues surrounding the mobile video transmissions is that of radio spectrum. As these transmissions will not utilize the bands already allocated for cellular communications, further bandwidth will be needed. This is likely to delay the introduction of mobile video services in some countries.

In an ideal world it would be advantageous to allocate the same bands for mobile video broadcasting worldwide. In reality this is unlikely to happen totally, although there will undoubtedly be a large degree of commonality, although it is likely that not all countries will be able to adopt the same bands. In addition to this there are no bands set aside for DVB-H broadcasting, although for DMB, the technology is sufficiently similar to utilize the bands allocated for broadcasting using DAB.

For the long term it is anticipated that as the UHF analogue transmissions are closed this will release vast amounts of valuable spectrum and some of this could be allocated to mobile video broadcasts and in particular for DVB-H that ahs no allocations that can currently be used.

Main systems
As might be expected, there are several systems being developed. Around the world there are four major standards that are appearing, two of which appear to be open international standards.

* T-DMB Terrestrial Digital Multimedia Broadcasting

* DVB-H Digitla Multimedia Broadcasting Handheld

* ISDB-T Integrated Services Digital Broadcasting Terrestrial

* MediaFLO

Of these T-DMB is based on the DAB system that is currently gaining significant support in the UK and other countries around the world for audio broadcasting. DVB-H is based upon the DVB-T terrestrial television broadcasting system that is used in the UK and many other countries. ISDB-T is a standard that is only being used in Japan, and finally Mediaflow is a scheme that is being developed by Qualcomm. Mediaflow is a registered trade name of Qualcomm

System comparisons
It is worth looking at the comparisons between the various systems that are being trialled and utilized for downloading video.

A number of trials of both DMB and DVB-H have taken place in a number of countries around the globe. It appears that in the future both systems may be used, and additionally a variety of different frequency bands may be used. To combat this and enable phones to provide mobile video roaming, some manufacturers are developing multi-band multi-standard chipsets. Although the risk is that these will consume more battery power, careful design has ensured that this may not be the case.

i-mode for cell phone emails and surfing

2006-11-15T08:42:00.000+08:00

i-mode (imode) is the platform for mobile phone communications that has had an astounding success in Japan. Now the company that launched the system in February 1999, NTT DoCoMo, is launching other i-mode cell phone systems in other countries around the world.

i-mode is an information service, and this give rise to its name. It is provided as a premium add-on service to the basic cellular phone system and provides many facilities including e-mail, internet surfing, and picture mailing. Requiring special i-mode terminals to be purchased by the user, the system operates as a packet based network overlaid on the basic cellular system.

i-mail
One of the most popular aspects of the i-mode service is i-mail that enables users to send e-mail messages to others on the same service, or to anyone with an e-mail account.

There are limits to the number of characters that can be sent and received, but these are much greater than the limits that apply to the SMS service that has become so popular on GSM. For i-mode users can send messages up to 500 characters in length and can receive up to 4000 characters.

On opening an i-mode account, users are given an e-mail address that consists of a random mix of characters. This can be changed once the account has been set up to personalise it for the user, and to make it more memorable.

Internet access
The other major element of i-mode is its ability to surf the internet and access internet sites. Specially developed websites using a cut down version of HTML known as cHTML is used to enable sites to be downloaded more quickly whilst providing content that can be satisfactorily viewed on the phones. The i-mode menu on the phone enables the user to select one of four zones: namely Lifestyle (for sports, weather local events etc); Transaction (for facilities including banking, shopping, credit card information and the like); Database (for services including traffic updates, TV and radio schedules as well as cinema information); and Entertainment (where games and music downloads are available along with screen savers, ring tones and hobby information).

One of the major incentives to the development of the special i-mode sites is that the operators have been investing in the content developers to develop official i-mode sites. Rather than splitting the revenues 50/50 as in the case of other similar systems, a revenue share of 90/10 in favour of the content developer has been adopted. This has stimulated a healthy growth and there are many thousands of official i-mode sites, with countless thousands more unofficial ones that are i-mode compatible. This means that the user has a great degree of choice and the usage has risen. In this way the operator has been able to see considerably increased revenues.

On top of these services there is i-shot for taking and sending pictures as well as i-appli for running applications such as downloaded games, and i-area for location based services.

The System
The i-mode system was originally run on the PDC system that is found in Japan, however it can also be applied to other cellular systems as well.

Based on a packet transmission to the mobile phones, i-mode uses a protocol known as PDC-P (Personal Digital Cellular Packet) for the interchange of data packets. The service is based on a 3 channel TDMA model to provide a common access system that can be shared by multiple users on a random access basis. Using multi-slot transmissions across the three channels data speeds of up to 28.8 kbps can be achieved.

CDMA2000 1X is widespread in Japan, and i-mode system is also used with this system. As data speeds are very much higher for 1X, this enables faster and easier access of the data. Services are also available on the faster data only CDMA2000 1xEV-DO system that enables data transfer at rates up to 2.4 Mbps.

The future of i-mode looks bright. While it may have been originally seen as a rising sun in the East, it is now appearing in many countries and competing with other technologies and systems.

Overview UMTS / W-CDMA Part 5

2006-11-11T14:45:00.000+08:00

This final page of the UMTS / WCDMA tutorial looks at three elements of the system, namely the way packet data is carried, the way speech coding is accomplished and handover, including hard, soft and softer handover.

Packet data
Packet data is an increasingly important element within mobile phone applications. WCDMA is able to carry data in this format in two ways. The first is for short data packets to be appended directly to a random access burst. This method is called common channel packet transmission and it is used for short infrequent packets. It is preferable to transmit short packets in this manner because the link maintenance needed for a dedicated channel would lead to an unacceptable overhead. Additionally the delay in setting up a packet data channel and transferring the operational mode to this format is avoided.

Larger or more frequent packets have to be transmitted on a dedicated channel. A large single packet is transmitted using a single-packet scheme where the dedicated channel is released immediately after the packet has been transmitted. In a multipacket scheme the dedicated channel is maintained by transmitting power control and synchronization information between subsequent packets.

Speech coding
Speech coding in UMTS uses a variety of source rates. As a result, a variety of vocoders are employed including the GSM EFR vocoder. When a variety of rates are available, a system known as Adaptive Multi-Rate (AMR) may be employed where rate is chosen according to the system capacity and requirements. This scheme is the same as that used on GSM. The actual vocoder that is chosen is governed by the system.

The speech coding process can be combined with a voice activity detector. This is particularly useful because during normal conversations there are long periods of inactivity. In the same way that discontinuous transmission is applied to GSM, the same is also true for UMTS. It employs the same technique of inserting background noise when there is no speech as when the discontinuous transmission cuts out the transmission no background noise would otherwise be heard and this can be very disconcerting for the listener.

Discontinuous reception
One of the big issues with mobile phones in general is that of battery life. It is one of the key differentiators that people take into account when buying a phone and this gives a measure of its importance. Taking this into consideration when developing the UMTS / WCDMA standard a discontinuous reception or sleep mode was introduced. This mode allows several non-essential segments of the phone circuitry to power down during periods when paging messages will not be received.

To enable this facility to be introduced into the UMTS UE circuitry the paging channel is divided into groups or subchannels. The actual number of the paging subchannel to be used by a particular UE is assigned by the network. In this way the UE only has to listen for part of the time. To achieve this the Paging Indicator Channel (PICH) is split into 10 ms frames, each of which comprises 300 bits - 288 for paging data and 12 idle bits. At the beginning of each paging channel frame there is a Paging Indicator (PI) that identifies the paging group being transmitted. By synchronising with the paging channels being transmitted it is able to turn the receiver on only when it needs to monitor the paging channel. As the receiver, with its RF circuitry, will consume power, savings can be made by switching it off.

Handover
Within UMTS, handover follows many of the similar concepts to those used for other CDMA systems. There are three basic types of handover: hard, soft and softer. All three types are used but under different circumstances.

Hard handover is like that used for the previous generations of systems. Here, as the UE moves out of range of one node B, the call has to be handed over to another frequency channel. In this instance simultaneous reception of both channels is not possible and there must be a physical break.

Soft handover is a technique that was not available on the previous generations of mobile phone systems. With CDMA systems it is possible to have adjacent cell sites on the same frequency, and as a result it is possible for the UE to receive the signals from two adjacent cells at once, and they are also able to receive the signals from the UE. When this occurs and handover is affected it is known as soft handover.

The decisions about handover are generally handled by the RNC. It continually monitors information regarding the signals being received by both the UE and node B and when a particular link has fallen below a given level and another better radio channel is available, it initiates a handover. As part of this monitoring process, the UE measures the Received Signal Code Power (RSCP) and Received Signal Strength Indicator (RSSI) and the information is then returned to the node B and hence to the RNC on the uplink control channel.

If a hard handover is required then the RNC will instruct the UE to adopt a compressed mode, allowing short time intervals in which the UE is able to measure the channel quality of other radio channels.

Overview UMTS / W-CDMA Part 4

2006-11-09T07:41:00.000+08:00

The data carried by the UMTS / WCDMA transmissions is organised into frames, slots and channels. In this way all the payload data as well as the control data can be carried in an efficient manner.

UMTS uses CDMA techniques (as WCDMA) as its multiple access technology, but it additionally uses time division techniques with a slot and frame structure to provide the full channel structure.

A channel is divided into 10 ms frames, each of which has fifteen time slots each of 666 microseconds length. On the downlink the time is further subdivided so that the time slots contain fields that contain either user data or control messages.

On the uplink dual channel modulation is used so that both data and control are transmitted simultaneously. Here the control elements contain a pilot signal, Transport Format Combination Identifier (TFCI), FeedBack Information (FBI) and Transmission Power Control (TPC).

The channels carried are categorised into three: logical, transport and physical channels. The logical channels define the way in which the data will be transferred, the transport channel along with the logical channel again defines the way in which the data is transferred, the physical channel carries the payload data and govern the physical characteristics of the signal.

The channels are organised such that the logical channels are related to what is transported, whereas the physical layer transport channels deal with how, and with what characteristics. The MAC layer provides data transfer services on logical channels. A set of logical channel types is defined for different kinds of data transfer services.

Logical Channels:

Broadcast Control Channel (BCCH), (downlink). This channel broadcasts information to UEs relevant to the cell, such as radio channels of neighbouring cells, etc.

Paging Control Channel (PCCH), (downlink). This channel is associated with the PICH and is used for paging messages and notification information.

Dedicated Control Channel (DCCH), (up and downlinks) This channel is used to carry dedicated control information in both directions.

Common Control Channel (CCCH), (up and downlinks). This bi-directional channel is used to transfer control information.

Shared Channel Control Channel (SHCCH), (bi-directional). This channel is bi-directional and only found in the TDD form of WCDMA / UMTS, where it is used to transport shared channel control information.

Dedicated Traffic Channel (DTCH), (up and downlinks). This is a bidirectional channel used to carry user data or traffic.

Common Traffic Channel (CTCH), (downlink) A unidirectional channel used to transfer dedicated user information to a group of UEs.

Transport Channels:

Dedicated Transport Channel (DCH), (up and downlink). This is used to transfer data to a particular UE. Each UE has its own DCH in each direction.

Broadcast Channel (BCH), (downlink). This channel broadcasts information to the UEs in the cell to enable them to identify the network and the cell.

Forward Access Channel (FACH),(down link). This is channel carries data or information to the UEs that are registered on the system. There may be more than one FACH per cell as they may carry packet data.

Paging Channel (PCH) (downlink). This channel carries messages that alert the UE to incoming calls, SMS messages, data sessions or required maintenance such as re-registration.

Random Access Channel (RACH), (uplink). This channel carries requests for service from UEs trying to access the system

Uplink Common Packet Channel (CPCH), (uplink). This channel provides additional capability beyond that of the RACH and for fast power control.

Downlink Shared Channel (DSCH) (downlink).This channel can be shared by several users and is used for data that is "bursty" in nature such as that obtained from web browsing etc.

Physical Channels:

Primary Common Control Physical Channel (PCCPCH) (downlink). This channel continuously broadcasts system identification and access control information.

Secondary Common Control Physical Channel (SCCPCH) (downlink) This channel carries the Forward Access Channel (FACH) providing control information, and the Paging Channel (PACH) with messages for UEs that are registered on the network.

Physical Random Access Channel (PRACH) (uplink). This channel enables the UE to transmit random access bursts in an attempt to access a network.

Dedicated Physical Data Channel (DPDCH) (up and downlink). This channel is used to transfer user data.

Dedicated Physical Control Channel (DPCCH) (up and downlink). This channel carries control information to and from the UE. In both directions the channel carries pilot bits and the Transport Format Combination Identifier (TFCI). The downlink channel also includes the Transmit Power Control and FeedBack Information (FBI) bits.

Physical Downlink Shared Channel (PDSCH) (downlink). This channel shares control information to UEs within the coverage area of the node B.

Physical Common Packet Channel (PCPCH). This channel is specifically intended to carry packet data. In operation the UE monitors the system to check if it is busy, and if not it then transmits a brief access burst. This is retransmitted if no acknowledgement is gained with a slight increase in power each time. Once the node B acknowledges the request, the data is transmitted on the channel.

Synchronisation Channel (SCH) The synchronisation channel is used in allowing UEs to synchronise with the network.

Common Pilot Channel (CPICH) This channel is transmitted by every node B so that the UEs are able estimate the timing for signal demodulation. Additionally they can be used as a beacon for the UE to determine the best cell with which to communicate.

Acquisition Indicator Channel (AICH) The AICH is used to inform a UE about the Data Channel (DCH) it can use to communicate with the node B. This channel assignment occurs as a result of a successful random access service request from the UE.

Paging Indication Channel (PICH) This channel provides the information to the UE to be able to operate its sleep mode to conserve its battery when listening on the Paging Channel (PCH). As the UE needs to know when to monitor the PCH, data is provided on the PICH to assign a UE a paging repetition ratio to enable it to determine how often it needs to 'wake up' and listen to the PCH.

CPCH Status Indication Channel (CSICH) This channel, which only appears in the downlink carries the status of the CPCH and may also be used to carry some intermittent, or "bursty" data. It works in a similar fashion to PICH.

Collision Detection/Channel Assignment Indication Channel (CD/CA-ICH) This channel, present in the downlink is used to indicate whether the channel assignment is active or inactive to the UE.

Overview UMTS / WCDMA Part 3

2006-11-04T14:34:00.000+08:00

Physical layer within UMTS / WCDMA is totally different to that employed by GSM. It employs a spread spectrum transmission in the form of CDMA rather than the TDMA transmissions used for GSM. Additionally it currently uses different frequencies to those allocated for GSM.

Frequencies
There are currently six bands that are specified for use for UMTS / WCDMA although operation on other frequencies is not precluded. However much of the focus for UMTS is currently on frequency allocations around 2 GHz. At the World Administrative radio Conference in 1992, the bands 1885 - 2025 and 2110 - 2200 MHz were set aside for use on a world wide basis by administrations wishing to implement International Mobile Telecommunications-2000 (IMT-2000). The aim was that allocating spectrum on a world wide basis would facilitate easy roaming for UMTS / WCDMA users.

Within these bands the portions have been reserved for different uses:

* 1920-1980 and 2110-2170 MHz Frequency Division Duplex (FDD, W-CDMA) Paired uplink and downlink, channel spacing is 5 MHz and raster is 200 kHz. An Operator needs 3 - 4 channels (2x15 MHz or 2x20 MHz) to be able to build a high-speed, high-capacity network.

* 1900-1920 and 2010-2025 MHz Time Division Duplex (TDD, TD/CDMA) Unpaired, channel spacing is 5 MHz and raster is 200 kHz. Transmit and receive transmissions are not separated in frequency.

* 1980-2010 and 2170-2200 MHz Satellite uplink and downlink.

Carrier frequencies are designated by a UTRA Absolute Radio Frequency Channel Number (UARFCN). This can be calculated from:

UARFCN = 5 x (frequency in MHz)

UMTS uses wideband CDMA as the radio transport mechanism. The channels are spaced by 5 MHz. The modulation that is used is different on the uplink and downlink. The downlink uses quadrature phase shift keying (QPSK) for all transport channels. However the uplink uses two separate channels so that the cycling of the transmitter on and off does not cause interference on the audio lines, a problem that was experienced on GSM. The dual channels (dual channel phase shift keying) are achieved by applying the coded user data to the I or In-phase input to the DQPSK modulator, and control data which has been encoded using a different code to the Q or quadrature input to the modulator.

Spreading
The data to be transmitted is encoded using a spreading code particular to a given user. In this way only the desired recipient is able to correlate and decode the signal, all other signals appearing as noise. This allows the physical RF channel to be used by several users simultaneously.

The data of a CDMA signal is multiplied with a chip or spreading code to increase the bandwidth of the signal. For WCDMA, each physical channel is spread with a unique and variable spreading sequence. The overall degree of spreading varies to enable the final signal to fill the required channel bandwidth. As the input data rate may vary from one application to the next, so the degree of spreading needs to be varied accordingly.

For the downlink the transmitted symbol rate is 3.84 M symbols per second. As the form of modulation used is QPSK this enables two bits of information to be transmitted for every symbol, thereby enabling a maximum data rate of twice the symbol rate or 7.68 Mbps. Therefore if the actual rate of the data to be transmitted is 15 kbps then a spreading factor of 512 is required to bring the signal up to the required chip rate for transmission in the required bandwidth. If the data to be carried has a higher data rate then a lower spreading rate is required to balance this out. It is worth remembering that altering the chip rate does alter the processing gain of the overall system and this needs to be accommodated in the signal processing as well. Higher spreading factors are more easily correlated by the receiver and therefore a lower transmit power can be used for the same symbol error rate.

The codes required to spread the signal must be orthogonal if they are to enable multiple users and channels to operate without mutual interference. The codes used in W-CDMA are Orthogonal Variable Spreading Factor (OVSF) codes, and they must remain synchronous to operate. As it is not possible to retain exact synchronisation for this, a second set of scrambling codes is used to ensure that interference does not result. This scrambling code is a pseudo random number (PN) code. Thus there are two stages of spreading. The first using the OSVF code and the second using a scrambling PN code. These codes are used to provide different levels of separation. The OVSF spreading codes are used to identify the user services in the uplink and user channels in the downlink whereas the PN code is used to identify the individual node B or UE.

On the uplink there is a choice of millions of different PN codes. These are processed to include a masked individual code to identify the UE. As a result there are more than sufficient codes to accommodate the number of different UEs likely to access a network. For the downlink a short code is used. There are a total of 512 different codes that can be used, one of which will be assigned to each node B.

Synchronisation
The level of synchronisation required for the WCDMA system to operate is provided from the Primary Synchronisation Channel (P-SCH) and the Secondary Synchronisation Channel (S-SCH). These channels are treated in a different manner to the normal channels and as a result they are not spread using the OVSFs and PN codes. Instead they are spread using synchronisation codes. There are two types that are used. The first is called the primary code and is used on the P-SCH, and the second is named a secondary code and is used on the S-SCH.

The primary code is the same for all cells and is a 256 chip sequence that is transmitted during the first 256 chips of each time slot. This allows the UE to synchronise with the base station for the time slot.

Once the UE has gained time slot synchronisation it only knows the start and stop of the time slot, but it does not know information about the particular time slot, or the frame. This is gained using the secondary synchronisation codes.

There is a total of sixteen different secondary synchronisation codes. One code is sent at the beginning of the time slot, i.e. the first 256 chips. It consists of 15 synchronisation codes and there are 64 different scrambling code groups. When received, the UE is able to determine before which synchronisation code the overall frame begins. In this way the UE is able to gain complete synchronisation.

The scrambling codes in the S-SCH also enable the UE to identify which scrambling code is being used and hence it can identify the base station. The scrambling codes are divided into 64 code groups, each having eight codes. This means that after achieving frame synchronisation, the UE only has a choice of one in eight codes and it can therefore try to decode the CPICH channel. Once it has achieved this it is able to read the BCH information and achieve better timing and it is able to monitor the P-CCPCH.

Power Control
As with any CDMA system it is essential that the base station receives all the UEs at approximately the same power level. If not, the UEs that are further away will be lower in strength than those closer to the node B and they will not be heard. This effect is often referred to as the near-far effect. To overcome this the node B instructs those stations closer in, to reduce their transmitted power, and those further away to increase theirs. In this way all stations will be received at approximately the same strength.

It is also important for node Bs to control their power levels effectively. As the signals transmitted by the different node Bs are not orthogonal to one another it is possible that signals from different ones will interfere. Accordingly their power is also kept to the minimum required by the UEs being served.

To achieve the power control there are two techniques that are employed: open loop; and closed loop.

Open loop techniques are used during the initial access before communication between the UE and node B has been fully established. It simply operates by making a measurement of the received signal strength and thereby estimating the transmitter power required. As the transmit and receive frequencies are different, the path losses in either direction will be different and therefore this method cannot be any more than a good estimate.

Once the UE has accessed the system and is in communication with the node B, closed loop techniques are used. A measurement of the signal strength is taken in each time slot. As a result of this a power control bit is sent requesting the power to be stepped up or down. This process is undertaken on both the up and downlinks. The fact that only one bit is assigned to power control means that the power will be continually changing. Once it has reached approximately the right level then it would step up and then down by one level. In practice the position of the mobile would change, or the path would change as a result of other movements and this would cause the signal level to move, so the continual change is not a problem.

UMTS TDD (Universal mobile telecommunications system - time division duplex)

2006-11-04T10:27:00.000+08:00

UMTS TDD (Universal mobile telecommunications system - time division duplex) is a growing standard. Although UMTS TDD is not as widely deployed as the more popular UMTS FDD which is being deployed for the 3G mobile phone systems, UMTS TDD is nevertheless being used and providing a viable service for many applications. In particular it is being used to provide mobile broadband data services, and other applications may include its use in providing mobile TV applications.

TDD - time division duplex
A communications system requires that communication is possible in both directions: to and from the base station to the remote station. There are a number of ways in which this can be achieved. The most obvious is to transmit on one frequency and receive on another. The frequency difference between the two transmissions being such that the two signals do not interfere. This is known as frequency division duplex (FDD) and it is one of the most commonly used schemes, and it is used by most cellular schemes.

It is also possible to use a single frequency and rather than using different frequency allocations, use different time allocations. If the transmission times are split into slots, then transmissions in one direction take place in one time slot, and those in the other direction take place in another. It is this scheme that is known as time division duplex, TDD, and it is used for UMTS-TDD.

One of the major advantages of TDD systems such as UMTS TDD is that it is possible to vary the capacity in either direction. By altering the proportion of time allocated for transmission in each direction (downlink and uplink) it is possible to enable it to match the traffic load in each direction.

Typically there is more traffic in the downlink (network to the mobile) than in the uplink (mobile to network). Accordingly the operator is able to allocate more time to the downlink transmission than the uplink. This is not possible with the paired spectrum required for FDD systems where it is not possible to re-allocate the use of the different bands. As a result of this, it is possible to make very efficient use of the available spectrum.

UMTS TDD within 3GPP
Al the standards for UMTS 3G systems have been defined under the auspices of 3GPP - the third generation partnership project. The standards not only define the FDD systems, but also the TDD system.

In these specifications, it was the original intent of UMTS that the TDD spectrum would be used to provide high data rates in selected areas forming what could be termed 3G hot zones.

UMTS TDD details
UMTS TDD uses many of the same basic parameters as UMTS FDD. The same 5 MHz channel bandwidths are used. UMTS TDD also uses direct sequence spread spectrum and different users and what can be termed "logical channels" are separated using different spreading codes. Only when the receiver uses the same code in the correlation process, is the data recovered. In W-CDMA all other logical channels using different spreading codes appear as noise on the channel and ultimately limit the capacity of the system. In UMTS TDD, a scheme known as multi user detection (MUD) is employed in the receiver and improves the removal of the interfering codes, allowing higher data rates and capacity.

In addition to the separation of users by using different logical channels as a result of the different spreading codes, further separation between users may be provided by allocating different time slots. There are 15 time slots in UMTS TDD. Of these, three are used for overhead such as signalling, etc and this leaves twelve time slots for user traffic. In each timeslot there can be 16 codes. Capacity is allocated to users on demand, using a two dimensional matrix of timeslots and codes.

In order for UMTS TDD to achieve the best overall performance, the transport format, i.e. the modulation and forward error correction can be altered for each user. The schemes are chosen by the network, and will depend on the signal characteristics in both directions. Higher order forms of modulation enable higher data speeds to be accommodated, but they are less resilient to noise and interference, and this means that the higher data rate modulation schemes are only used when signal strengths are high. Additionally the levels of forward error correction can be changed. When errors are likely, i.e. when signal strengths are low or interference levels are high, Similarly higher levels of forward error correction are needed under low require additional data to be sent and this slows the payload transfer rate. Thus it is possible to achieve much higher data transfer rates when signals are strong and interference levels are low.

Spectrum allocations
Standard allocations of radio spectrum have been made for 3G telecommunications systems in most countries around the globe. In Europe and many other areas spectrum has been allocated for UMTS FDD between 1920MHz to 1980MHz and 2110MHz to 2170MHz. For UMTS TDD spectrum is primarily located between 1900MHz and 1920MHz and between 2010MHz and 2025MHz. In addition to this there are some other allocations around 3 GHz.

UMTS TDD performance
UMTS TDD is able to support high peak data rates. Release 5 of the UMTS standard provides HSDPA (high-speed downlink packet access). The scheme allows the use of a higher order modulation scheme called 16-QAM (16 point quadrature amplitude modulation), which enables peak rates of 10 Mbps per sector in commercial deployments. The next release increases the modulation to 64-QAM, and introduces intercell interference cancellation (called Generalized MUD) and MIMO (multiple in, multiple out). In combination, these increase the peak rate to 31 Mbps per sector.

Future
UMTS TDD, while not as widely deployed as UMTS FDD nevertheless offers significant advantages for a number of applications. While currently being used for mobile broadband, it appears as if it could serve to provide mobile TV, and other data in a filed where new methods of transport are being sought.

High Speed Packet Uplink Access (HSUPA)

2006-11-03T09:20:00.000+08:00

Work is now staring on developing the standards for High Speed Uplink Packet Access ( HSUPA ) to improve the data rates on the 3G W-CDMA mobile or cell phone standard. With the cellular telecommunications standards established and work progressing to introduce the equipment for High Speed Downlink Packet Access ( HSDPA ), the standards are now starting to be developed to enable the uplink from the mobile handset or User Equipment (UE) to the base station (Node B) to be able to handle data at similar speeds. This is known as HSUPA and it will enable new features including full video conferencing to be introduced.

For most applications including internet surfing, emails, video downloads and the like, data flowing in the downlink is far greater than the uplink. However for applications such as video conferencing, data flows equally in both directions. It is anticipated that video conferencing will become an increasing requirement, and a significant revenue generator for the operators in the near future. To enable high quality video to be passed, it is therefore essential to ensure that the uplink performs as fast as the downlink.

Although it is very early days for the standards, work on HSUPA has already started under the auspices of 3GPP, the body that controls the Wideband CDMA (W-CDMA) standards.

Technologies used
It is anticipated that many of the same techniques used in HSDPA will be used for HSUPA, but these still need to be formalised. Accordingly it is expected that adaptive modulation, along with HARQ (hybrid automatic repeat request) will be used. Improvements in the base station similar to those employed on HSDPA are also likely.

Originally W-CDMA had used only QPSK as the modulation scheme, however under the new HSUPA system,16-QAM which can carry a higher data rate, but is less resilient to noise is also used when the link is sufficiently robust. The robustness of the channel and its suitability to use 16-QAM instead of QPSK is determined by analysing information fed back about a variety of parameters. These include details of the channel physical layer conditions, power control, Quality of Service (QoS), and information specific to HSDPA.

Fast HARQ (hybrid automatic repeat request), has also been implemented along with multi-code operation and this eliminates the need for a variable spreading factor. By using these approaches all users, whether near or far from the base station are able to receive the optimum available data rate.

It is also likely that within the HSUPA upgrades there will be an additional uplink data channel introduced comparable to that in the downlink.

The future
Many manufacturers are working on implementing HSDPA, with initial equipment deliveries anticipated in 2005. Now with HSUPA in people's sights this should be implemented in the following years, making a far faster 3G system than is currently available.

High Speed Packet Downlink Access ( HSDPA )

2006-11-02T14:30:00.000+08:00

Improvements and enhancements are being made to the Wideband CDMA or UMTS 3G telecommunications system. Called High Speed Downlink Packet Access ( HSDPA ) the new technology promises to increase the download data rate five fold. In addition to this HSDPA also provides a two fold increase in base station capacity.

The introduction of HSDPA technology has come about as a result of the need to drive down costs as well as increasing the data rates possible. Current trends show the volume of packet switched data rising and overtaking the more traditional circuit switched traffic. By adopting a packet based approach to the delivery of digital content as well as IP based person to person digitized voice, a single session can be used for multiple purposes and this can be used to drive revenues upwards. With this approach in mind the use of HSDPA is a key element in providing the user with a better service as well as increasing revenues as a result of increased capacity and usage for the service providers.

Standards
The new high speed technology part of the W-CDMA evolution. Release 4 of the 3GPP W-CDMA standard provided the efficient IP support to enable provision of services through an all IP core network. Then Release 5 included HSDPA itself with support for the packet-based multimedia services. A further enhancement known as MIMO (Multiple Input Multiple Output) will then be contained within Release 6. As HSDPA needs to work alongside the original Release 99 systems, the new technology is completely backwards compatible.

Key technologies
One of the keys to the operation of HSDPA is the use of an additional form of modulation. Originally W-CDMA had used only QPSK as the modulation scheme, however under the new system16-QAM which can carry a higher data rate, but is less resilient to noise is also used when the link is sufficiently robust. The robustness of the channel and its suitability to use 16-QAM instead of QPSK is determined by analyzing information fed back about a variety of parameters. These include details of the channel physical layer conditions, power control, Quality of Service (QoS), and information specific to HSDPA.

Fast HARQ (hybrid automatic repeat request), has also been implemented along with multi-code operation and this eliminates the need for a variable spreading factor. By using these approaches all users, whether near or far from the base station are able to receive the optimum available data rate.

Further advances have been made in the area of scheduling. By moving more intelligence into the base station, data traffic scheduling can be achieved in a more dynamic fashion. This enables variations arising from fast fading can be accommodated and the cell is even able to allocate much of the cell capacity for a short period of time to a particular user. In this way the user is able to receive the data as fast as conditions allow.

A further channel known as the High Speed Downlink Shared Channel (HS-DSCH) has been introduced. W-CDMA normally carries data over dedicated transport channels (DCHs), several of which are multiplexed onto one RF carrier. This approach has been adopted because it provides the optimum performance with continuous user data. Under the new scheme the "bursty" nature of the data has been accounted for and more efficient use of the available spectrum has been made.

Performance
Using the new HSDPA scheme it will be possible to achieve peak data rates of 10 Mbps within the 5 MHz channel bandwidth offered under W-CDMA. The new scheme has a number of benefits. It improves the overall network packet data capacity, improves the spectral efficiency and will enable networks to achieve a lower delivery cost per bit. Users will see higher data speeds as well as shorter service response times and better availability of services. However new mobile designs will need to be able to handle the increased data throughput rates. Reports indicate that handsets will need to have at least double the memory currently contained within handsets. Nevertheless the advantages of HSDPA mean that it will be widely used as networks are upgraded and new phones introduced.

Overview UMTS / WCDMA Part 2

2006-10-30T08:32:00.000+08:00

UMTS, the Universal Mobile Telecommunications System is the third generation (3G) successor to the second generation GSM based technologies including GPRS, and EDGE. Although UMTS uses a totally different air interface, the core network elements have been migrating towards the UMTS requirements with the introduction of GPRS and EDGE. In this way the transition from GSM to the 3G UMTS architecture does not require such a large instantaneous investment.

UMTS uses Wideband CDMA (WCDMA or W-CDMA) to carry the radio transmissions, and often the system is referred to by the name WCDMA. It is also gaining a third name. Some are calling it 3GSM because it is a 3G migration for GSM.

Specifications and Management
In order to create and manage a system as complicated as UMTS or WCDMA it is necessary to develop and maintain a large number of documents and specifications. For UMTS or WCDMA, these are now managed by a group known as 3GPP - the Third Generation Partnership Programme. This is a global co-operation between six organisational partners - ARIB, CCSA, ETSI, ATIS, TTA and TTC.

The scope of 3GPP was to produce globally applicable Technical Specifications and Technical Reports for a 3rd Generation Mobile Telecommunications System. This would be based upon the GSM core networks and the radio access technologies that they support (i.e., Universal Terrestrial Radio Access (UTRA) both Frequency Division Duplex (FDD) and Time Division Duplex (TDD) modes).

Capabilities
UMTS uses Wideband CDMA - WCDMA - as the radio transmission standard. It employs a 5 MHz channel bandwidth. Using this bandwidth it has the capacity to carry over 100 simultaneous voice calls, or it is able to carry data at speeds up to 2 Mbps in its original format. However with the later enhancements of HSDPA and HSUPA (described in other articles accessible from the cellular telecommunications menu page ) included in later releases of the standard the data transmission speeds have been increased to 14.4 Mbps.

Many of the ideas that were incorporated into GSM have been carried over and enhanced for UMTS. Elements such as the SIM have been transformed into a far more powerful USIM (Universal SIM). In addition to this, the network has been designed so that the enhancements employed for GPRS and EDGE can be used for UMTS. In this way the investment required is kept to a minimum.

A new introduction for UMTS is that there are specifications that allow both Frequency Division Duplex (FDD) and Time Division Duplex (TDD) modes. The first modes to be employed are FDD modes where the uplink and downlink are on different frequencies. The spacing between them is 190 MHz for Band 1 networks being currently used and rolled out.

However the TDD mode where the uplink and downlink are split in time with the base stations and then the mobiles transmitting alternately on the same frequency is particularly suited to a variety of applications. Obviously where spectrum is limited and paired bands suitably spaced are not available. It also performs well where small cells are to be used. As a guard time is required between transmit and receive, this will be smaller when transit times are smaller as a result of the shorter distances being covered. A further advantage arises from the fact that it is found that far more data is carried in the downlink as a result of internet surfing, video downloads and the like. This means that it is often better to allocate more capacity to the downlink. Where paired spectrum is used this is not possible. However when a TDD system is used it is possible to alter the balance between downlink and uplink transmissions to accommodate this imbalance and thereby improve the efficiency. In this way TDD systems can be highly efficient when used in picocells for carrying Internet data. The TDD systems have not been widely deployed, but this may occur more in the future. In view of its character, it is often referred to as TD-CDMA (Time Division CDMA).

UMTS or as it is often termed, Wideband CDMA, WCDMA is being widely deployed. It offers many advantages over GSM, GPRS, and EDGE in terms of much higher data rates and greater flexibility. These basic technical abilities reflect as a much richer number of applications and features that the 3G phones can be used to perform. This not only gives the user a much more useful 'phone', but this also translates into higher revenues for the operator.

Overview of UMTS / WCDMA Part 1

2006-10-21T10:50:00.000+08:00

UMTS, the Universal Mobile Telecommunications System is the third generation (3G) successor to the second generation GSM based technologies including GPRS, and EDGE. Although UMTS uses a totally different air interface, the core network elements have been migrating towards the UMTS requirements with the introduction of GPRS and EDGE. In this way the transition from GSM to the 3G UMTS architecture does not require such a large instantaneous investment.

UMTS uses Wideband CDMA (WCDMA or W-CDMA) to carry the radio transmissions, and often the system is referred to by the name WCDMA. It is also gaining a third name. Some are calling it 3GSM because it is a 3G migration for GSM.

Specifications and Management
In order to create and manage a system as complicated as UMTS or WCDMA it is necessary to develop and maintain a large number of documents and specifications. For UMTS or WCDMA, these are now managed by a group known as 3GPP - the Third Generation Partnership Programme. This is a global co-operation between six organisational partners - ARIB, CCSA, ETSI, ATIS, TTA and TTC.

The scope of 3GPP was to produce globally applicable Technical Specifications and Technical Reports for a 3rd Generation Mobile Telecommunications System. This would be based upon the GSM core networks and the radio access technologies that they support (i.e., Universal Terrestrial Radio Access (UTRA) both Frequency Division Duplex (FDD) and Time Division Duplex (TDD) modes).

Capabilities
UMTS uses Wideband CDMA - WCDMA - as the radio transmission standard. It employs a 5 MHz channel bandwidth. Using this bandwidth it has the capacity to carry over 100 simultaneous voice calls, or it is able to carry data at speeds up to 2 Mbps in its original format. However with the later enhancements of HSDPA and HSUPA (described in other articles accessible from the cellular telecommunications menu page ) included in later releases of the standard the data transmission speeds have been increased to 14.4 Mbps.

Many of the ideas that were incorporated into GSM have been carried over and enhanced for UMTS. Elements such as the SIM have been transformed into a far more powerful USIM (Universal SIM). In addition to this, the network has been designed so that the enhancements employed for GPRS and EDGE can be used for UMTS. In this way the investment required is kept to a minimum.

A new introduction for UMTS is that there are specifications that allow both Frequency Division Duplex (FDD) and Time Division Duplex (TDD) modes. The first modes to be employed are FDD modes where the uplink and downlink are on different frequencies. The spacing between them is 190 MHz for Band 1 networks being currently used and rolled out.

However the TDD mode where the uplink and downlink are split in time with the base stations and then the mobiles transmitting alternately on the same frequency is particularly suited to a variety of applications. Obviously where spectrum is limited and paired bands suitably spaced are not available. It also performs well where small cells are to be used. As a guard time is required between transmit and receive, this will be smaller when transit times are smaller as a result of the shorter distances being covered. A further advantage arises from the fact that it is found that far more data is carried in the downlink as a result of internet surfing, video downloads and the like. This means that it is often better to allocate more capacity to the downlink.

Where paired spectrum is used this is not possible. However when a TDD system is used it is possible to alter the balance between downlink and uplink transmissions to accommodate this imbalance and thereby improve the efficiency. In this way TDD systems can be highly efficient when used in picocells for carrying Internet data. The TDD systems have not been widely deployed, but this may occur more in the future. In view of its character, it is often referred to as TD-CDMA (Time Division CDMA).

The Basics of CDMA2000 1xEV-DO

2006-10-20T08:59:00.000+08:00

CDMA2000 1x EV-DO cell phone system is a standard that has evolved from the CDMA2000 mobile phone system and it is now firmly established in many areas of the world. The letters EV-DO stand for Evolution Data Only or Data Optimised. From the title it can be seen that it is a data only mobile telecommunications standard that can be run on CDMA2000 networks.

The EV-DO cell phone system is capable of providing the full 3G data rates up to 3.1 Mbps now that release A of the standard has been issued. The first commercial CDMA2000 1xEV-DO network was deployed by SK Telecom (Korea) in January 2002. Now operators in Brazil Ecuador, Indonesia, Jamaica, Puerto Rico, Taiwan and the USA to mention but a few have all launched networks and more are to follow.

Basics
Th CDMA2000 1xEV-DO cell phone system is defined under IS-856 rather than IS-2000 that defines the other CDMA2000 standards, and as the name indicates it only carries data. In Release 0 of the standard the maximum data rate was 2400 Mbps in the forward (downlink) with 153 kbps in the reverse (uplink) direction, the same as CDMA2000 1X. However in the later release of the standard, Release A, the forward data rate rises to 3.1 Mbps, and 1.2 Mbps in the reverse direction.

The forward channel forms a dedicated variable-rate, packet data channel with signalling and control time multiplexed into it. The channel is itself time-divided and allocated to each user on a demand and opportunity driven basis. A data only format was adopted to enable the standard to be optimised for data applications. If voice is required then a dual mode phone using separate 1X channel for the voice call is needed. In fact the "phones" used for data only applications are referred to as Access Terminals or ATs.

Air interface
The EV-DO RF transmission is very similar to that of a CDMA2000 1X transmission. It has the same final spread rate of 1.228 Mcps and it has the same modulation bandwidth because the same digital filter is used. Although 1xEV-DO has many similarities with 1X transmissions, it cannot occupy the same channels simultaneously, and therefore requires dedicated paired channels for its operation. Accordingly new bands, often in the new 3G allocations are being dedicated for EV-DO in some areas.

Forward link
The forward link possesses many features that are specific to EV-DO, having been optimised for data transmission, particularly in the downlink direction. Average continuous rates of 600 kbps per sector are possible. This is a six fold increase over CDMA2000 1X and is provided largely by the ability of 1xEV-DO to negotiate increased data rates for individual ATs because only one user is served at a time.

The forward link is always transmitted at full power and uses a data rate control scheme rather than the power control scheme used with 1X, and the data is time division multiplexed so that only one AT is served at a time.

In order to be able to receive data, each EV-DO AT measures signal-to-noise ratio (S/N) on the forward link pilot every slot, i.e. 1.667 ms. Based on the information this provides the AT sends a data rate request to the base station. The AN receives requests from a variety of ATs, and decisions have to be made regarding which ATs are to be served next. The AN endeavours to achieve the best data transfer, and this is done by serving those ATs offering a good signal to noise ratio. This is achieved at the expense of users at some distance from the AN's antenna.

Accurate time synchronisation is required between the EV-DO Access Nodes. To achieve this time information is taken from the Global Positioning System as this is able to provide an exceedingly accurate time signal.

Forward link channels
A number of channels are transmitted in the forward direction to enable signalling, data and other capabilities to be handled. These channels include the Traffic channel, MAC channel, Control channel and Pilot. These are time division multiplexed.

Traffic Channel - This channels uses Quadrature Phase Shift Keying (QPSK) modulation for data rates up to 1.2288 Mbps. For higher data rates, higher order modulation techniques are used in the form of 8PSK with 3 bits per symbol or 16QAM with 4 bits per symbol. The levels of the I and Q symbols are chosen so that the average power becomes 1.

The Incoming data to be used as the modulation comes from the from the turbo coder and is scrambled by mixing it with a Pseudo Random Number (PN) sequence. The initial state of the PN is derived from known parameters, and is unique for each user. Every packet starts at the same initial value of the PN sequence.

At the beginning of the transmission to each user, there is a preamble that contains the user ID for the data. Its repeat rate is determined by the data rate because lower data rates require higher repeat values. However even at its largest, the preamble will fill no more than half the first slot.

Control Channel - This channel carries the signalling and overhead messages.

Pilot - The differentiator between the cell and the sector is still the PN offset of the pilot channel and the pilot signal is only gated on for 192 chips per slot.

Medium Access Control (MAC) Channel - This channel carries a number of controls including the Reverse Power Control (RPC), the Data Rate Control (DRC) Lock, and the reverse activity (RA) channels.

Reverse Link
The reverse link for 1xEV-DO has a structure similar to that for CDMA2000. In EV-DO all signalling is performed on the data channel and this means that there is no Dedicated Control Channel. The data channel can support 5 data rates which are separated in powers of 2 from 9.6 to 153.6 kbps. These rates are achieved by varying the repeat factor. The highest rate uses a Turbo coder with lower gain. The following channels are transmitted in addition to those used with 1X:

Reverse Rate Indicator (RRI) Channel - This indicates the data rate of the Reverse Data Channel.

Acknowledgement (Ack) Channel - This channel is transmitted after the AT detects a frame with the preamble detailing it to be the recipient of the data.

Data Rate Control (DRC) Channel - This channel contains a four bit word in each slot to allow the choice of 12 different transmission rates.

cdmaOne Mobile Phone System"IS-95"

2006-10-14T08:39:00.000+08:00

IS-95 was the first CDMA mobile phone system to gain widespread use and it is found widely in North America. Its brand name is cdmaOne and the initial specification for the system was IS95A, but its performance was later upgraded under IS-95B. It is this later specification that is synonymous with cdmaOne. Apart from voice the mobile phone system is also able to carry data at rates up to 14.4 kbps for IS-95A and 115 kbps for IS-95B.

The IS-95 system was introduced by Qualcomm. They had been investigating the use of direct sequence spread spectrum techniques for military use when it was realised that it could be used as a multiple access technology for mobile communications. Previous systems had used frequency division multiple access (FDMA) to time division multiple access (TDMA). The principle of FDMA is that different users use different frequencies. This techniques was used for the analogue systems such as AMPS, TACS and NMT. The TDMA principle is used in GSM. Here in different users are allocated different time slots on a given channel.

CDMA
The CDMA or code division multiple access system used for IS-95 is very different. Although a complete summary of CDMA will not be included here, the basic principle of CDMA is that different codes are used to distinguish between the different users. CDMA uses a form of modulation known as direct sequence spread spectrum. Here a signal is generated that spreads out over a wide bandwidth. A code known as a spreading code is used to perform this action. By using a group of codes known as orthogonal codes, it is possible to pick out a signal with a given code in the presence of many other signals with different orthogonal codes. In fact many different baseband "signals" with different spreading codes can be modulated onto the same carrier to enable many different users to be supported. By using different orthogonal codes interference between the signals is minimal. Conversely when signals are received from several mobile stations, the base station is able to isolate each one as they have different orthogonal spreading codes. In fact the system has been likened to hearing many people in a room speaking different languages. Despite a very high noise level it is possible to pick out the person speaking your own language - English for example.

The advantage of using CDMA over FDMA and TDMA is that it enables a greater number of users to be supported. The improvement in efficiency is hard to define as it depends on many factors including the size of the cells and the level of interference between cells and several other factors.

Unlike the more traditional cellular systems where neighbouring cells use different sets of channels, a CDMA system re-uses the same channels. Signals from other cells will be appear as interference, but the system is able to extract the required signal by using the correct code in the demodulation and signal extraction process. Often more than one channel is used in each cell, and this provides additional capacity because there is a limit to the amount of traffic that can be supported on each channel.

Downlink signal
The downlink transmission (i.e. base station to the mobile) within IS-95 consists of a number of elements. There is the pilot channel and other further channels each with their own functions. The pilot channel corresponds to the control channel in GSM and enables the mobile to estimate the path loss and as a result of this to set its power level accordingly. In addition to this there are other channels for paging, speech, data etc. The speech is encoded using a voice encoder. Error correction is then applied to this data to enable it to be carried even under poor conditions. This brings the data rate up to 19.2 kbps. The next stage in the generation of the signal is to multiple the data by a Walsh code - the form of orthogonal code used to spread the signal when generating the CDMA signal itself. As this is a 64 bit code, this multiplies the data rate by 64 to bring the overall data rate to 1.228 Mbps. This signal is then transmitted.

Uplink signal
The uplink signal for IS-95 is generated in a different way. Although the same voice encoder is used, the resulting data has a greater degree of error correction or protection applied. Accordingly the resulting data rate is brought up to 28.8 kbps. A more complicated method of spreading using a Walsh code is used. However this only results in 307 kbps data stream. Further spreading is required. This is provided by using a different form of orthogonal spreading code known as a PN code. This is multiplied with the signal to increase its data rate by four to bring it up to the final data rate of 1.228 Mbps, the same as the downlink signal.

Soft handover
The reason that the uplink and downlink transmissions for IS-95 are generated in a different way results from the fact that it is difficult to synchronise the mobile handsets. Each one is a different distance away from the base station and the time delays will be different. As a result synchronisation is not possible. For the Walsh codes to maintain their orthogonality and to operate correctly they must be properly synchronised. PN codes do not require synchronisation and can be used more successfully under these circumstances.

One of the advantages of CDMA is the fact that handover can be made easier and more reliable. Normally when handing over from one from a base station in one cell to the base station in the next, it is necessary for the system to arrange for a new channel to be used. The mobile then changes channel and hopes to be able to receive the signal on the new one satisfactorily. Obviously there is a degree of risk, and occasionally a hand over does not proceed smoothly. With CDMA it is possible to use what is termed a soft hand over. As transmissions from the base stations in adjacent cells may be made on the same frequency, it is possible for a mobile to receive signals from two base stations at once. Normally the mobile would reject the signal from the second base station, but it is possible to arrange for it to receive signals from the two stations and this proves to be very useful during handover. During the period of the handover the two base stations transmit the same signal enabling the mobile to receive the signal via two routes at the same time. This means that during this handover phase the mobile should not loose the signal. Then as the mobile moves further into the second cell and the signal is firm, it can rely on one station only and the handover is complete.

This approach considerably reduces the risk of loosing the connection during handover, and it also minimises the risk of a short break in the speech during this period. However it is not free and there is an associated cost. The mobile needs two decoders to monitor and decode the two signals and this increases the complexity of the mobile. On the network side it means that two channels are used instead of one and this reduces the overall capacity. Some estimate this could be as high as 40%. This is dependent upon the speed of handover and the degree of overlap in the cells. The figure given is obviously a worst case scenario, but despite this the advantages are deemed to outweigh the reduction in capacity and increased mobile complexity.

IS-95 has been successfully installed in many areas of the world, chiefly in North America. IS 95 also has the advantage that it has an evolutionary migration path to 3G with CDMA2000 to give the higher data rates that are needed for video streaming and data transfer whilst retaining compatibility with the existing networks.

CDMA2000 / cdmaOne

2006-10-11T11:12:00.000+08:00

One of the major cell / mobile phone or cellular telecommunications technologies today is the cdmaOne / CDMA2000 system. One of its strengths is that it has focussed on being an evolutionary technology moving from standards such as IS-95 (IS-95A and IS-95B) for cdmaOne through to standards including IS-2000 and IS-856 for CDMA2000 1X, 1xEv, 1xEV-DO and 1xEV-DV. Currently the standard uses one standard channel under a system known as 1X RTT, although for the future three channels (3X RTT) may be used).

In view of the fact that the CDMA2000 system has been designed to be an evolutionary standard, it is relatively easy to introduce upgrades to the system. This has made it particularly popular with operators because the cost of upgrading to the new standards is much less, and they can have users with a variety of types of phone on the same network. Thus users may operate cdmaOne phones on the same network as CDMA2000 1X or CDMA2000 1xEV-DV phones.

The story of how the system was developed is particularly interesting, and it reveals much about the nature of the system as well as telling its significant successes.

The idea for using the form of modulation known as direct sequence spread spectrum (DSSS) for a multiple access system for mobile telecommunications came from a California based company called Qualcomm in the 1980s. Previously DSSS had been mainly used for military or covert communications systems as the transmissions were hard to detect, jam and eavesdrop.,span class="fullpost">

The system involved multiplying the required data with another data stream with a much higher data rate. Known as a spreading code, this widened the bandwidth required for the transmission, spreading it over a wide frequency band. Only when the original spreading code was used in the reconstruction of the data, would the original information be reconstituted. It was reasoned that by having different spreading codes, a multiple access system could be created for use in a mobile phone system.

In order to prove that the new system was viable a consortium was set up and Qualcomm was joined by US network operators Nynex and Ameritech to develop the first experimental code division multiple access (CDMA) system. Later the team was expanded as Motorola and AT&T (now Lucent) joined to bring their resources to speed development. As a result the new standard was published as IS-95A in 1995 under the auspices of the Cellular Telecommunications Industry Association (CTIA) and the Telecommunications Industry Association (TIA). As part of the development of CDMA an organisation called the CDMA Development Group (CDG) was formed from the founding network and manufacturers. Its purpose is to promote CDMA and evolve the technology and standards, although today most of the standards work is carried out by 3GPP2.

It then took a further three years before Hutchison Telecom became the first organisation to launch a system. It is now widely deployed in North America, and the Asia Pacific region, but there are also networks in South America, Africa, and the Middle East as well as some in Eastern Europe.

System Basics
The CDMA system was totally unlike any system used before. In the UK the original TACS system had used a channel spacing of 25 kHz and AMPS in the US had used 30 kHz. The new GSM system used 200 kHz channels whilst the US -TDMA standard kept compatibility with AMPS and was based around 30 kHz channels. CDMA, IS-95A, used a 1.25 MHz bandwidth and this was much wider than anything that had been used before. CDMA operates well with a wide bandwidth, but it was limited to 1.25MHz to remain compatible with the spectrum allocations that were available.

There were other differences as well. CDMA mobiles did not have SIM cards, although recently this has changed. Instead the subscriber data has simply been stored in memory of the mobile with a method of over-the-air programming of this data being available.

cdmaOne
The first offerings of CDMA under the guise of IS-95 catered for voice as well as data up to a speed of 14.4 kbps. However with the market moving towards data applications, the IS-95 specification was upgraded to IS-95B to cater for the needs of operators. This new specification allowed packet switched data transmission up to a speed of 64 kbps. IS-95B was first deployed in September 1999 in Korea and has since been adopted by operators in Japan and Peru.

Often IS-95 A and B versions are marketed under the brand name cdmaOne. This is a registered trademark of the CDMA Development Group.

CDMA2000 1X
cdmaOne had been very successful and was introduced into many countries, but with operators seeing revenue from voice services levelling off, the pressure to migrate to 3G technologies where data speeds were higher and revenue growth could be maintained. As a result of this the IS-2000 standard was developed to enable the higher 3G data rates to be provided.

Within IS-2000 a number of further developments were included. It was envisaged that with many more areas moving towards 3G standards and the old AMPS systems being made obsolete it would be possible to have systems operating on a wider bandwidth. As a result of this the new standards allowed for systems that would use the single channel bandwidth (1X or 1X RTT) and also ones that would use three times the bandwidth (3X). Currently all work is focussed on the 1X systems, with the idea for the 3X (or 3X RTT) systems to be used some time in the future.

CDMA2000 1X can double the voice capacity of cdmaOne networks and delivers peak packet data speeds of 307 kbps in mobile environments although today's commercial CDMA2000 1X networks (phase 1) support a peak data rate of 153.6 kbps. CDMA2000 1X has been designated a 3G standard and it is now widely deployed.

Evolution
CDMA2000 1X is the basic 3G standard, in fact some people only consider it a 2.75G system, and it is being developed beyond this. In what is termed CDMA2000 1xEv, there are further developments to bring it in line with the UMTS or Wideband CDMA system that is being deployed in Western Europe and many other areas.

The first of these known as CDMA2000 1xEV-DO (EVolution Data Only) is something of a sideline from the main evolutionary development of the standard. It is defined under IS-856 rather than IS-2000, and is as the name indicates is only carries data, but at speeds up to 2.4Mbps in the forward direction and the same as 1X in the reverse direction.

The forward channel forms a dedicated variable-rate, packet data channel with signalling and control time multiplexed into it. The channel is itself time-divided and allocated to each user on a demand and opportunity driven basis. A data only format was adopted so that the system could be optimised for data applications, and if voice is required then a dual mode phone using separate 1X channel for the voice call is required. In fact the "phones" used for data only applications are referred to as Access Terminals or ATs.

The first commercial CDMA2000 1xEV-DO network was deployed by SK Telecom (Korea) in January 2002. Now operators in Brazil Ecuador, Indonesia, Jamaica, Puerto Rico, Taiwan and the USA to mention but a few have all launched networks and more are to follow.

Data and voice
The next logical evolution of the system is to incorporate both data and voice into the standard. This is exactly what CDMA2000 1xEV-DV achieves. This is catered for under Release C of the IS-2000 standard. And is effectively 1X with additional high speed data channels. In this way it is able to provide complete backward compatibility with both CDMA2000 1X and cdmaOne. In addition to this the migration requires comparatively few upgrades to a 1X system and as such it is a very attractive option for network operators. Further developments are available under Release D of the IS-2000 standard. This provides for 3.1 Mbps data in both directions as well as many other upgrades.

The first CDMA networks in the form of IS-95 / cdmaOne were the first deployments of CDMA technology, the technology that is being used for all 3G cell phone systems. This formed the basis for a unique evolutionary system as CDMA2000. CDMA2000 is a well established 3G technology, and it is particularly successful in the USA, and Asia Pacific regions as well as having a significant presence in many other parts of the world. It was able to offer 3G services well before W-CDMA became established, and it is now continuing to build on this success.

Integrated Services Digital Network (ISDN) digital telecommunications system

2006-10-09T12:35:00.000+08:00

ISDN or Integrated Services Digital Network is an international standard for end to end digital transmission of voice, data and signaling. It can operate over copper based systems and allows the transmission of digital data over the telecommunications networks, typically ordinary copper based systems and providing higher data speeds and better quality than analogue transmission. The ISDN specifications provide a set of protocols that enable the set up, maintenance and completion of calls.

ISDN, Integrated Services Digital Network, provides a number of significant advantages over analogue systems.
In is basic form it enables two simultaneous telephone calls to be made over the same line simultaneously Faster call connection. It typically takes a second to make connections rather than the much longer delays experienced using purely analogue based systems. Data can be sent more reliably and faster than with the analogue systems Noise, distortion, echoes and crosstalk are virtually eliminated The digital stream can carry any form of data from voice to faxes and internet web pages to data files - this gives the name 'integrated services'

Usage
ISDN is in use around the world, but with the introduction of ADSL it is facing strong competition. The technology never gained much market share in the USA, although it used in other countries. In Japan it became reasonably popular in the late 1990s although it is now in decline with the advent of ADSL. The system was also introduced in Europe where providers such as BT, France Telecom and Deutsche Telekom introduced services.

Configurations
There are two types of channel that are found within ISDN. These are the 'B' and 'D' channels. The B or 'bearer' channels are used to carry the payload data which may be voice and / or data, and the d or 'Delta' channel is intended for signalling and control, although it may also be used for data under some circumstances.

Additionally there are two levels of ISDN access that may be provided. These are known as BRI and PRI.

BRI (Basic Rate Interface) - This consists of two B channels, eac of which provides a bandwidth of 64 kbps under most circumstances. One D channel with a bandwidth of 16 kbps is also provided. Together this configuration is often referred to as 2B+D.

The basic rate lines connect to the network using a standard twisted pair of copper wires. The data can then be transmitted simultaneously in both directions to provide full duplex operation. The data stream is carried as two B channesl as mentioned above, each of which carry 64 kbps (8 k bytes per second). This data is interleaved with the D channel data and this is used for call management: setting up, clearing down of calls, and some additional data to maintain synchronisation and monitoring of the line.

The network end of the line is referred to as the 'Line Termination' (LT) while the user end acts as a termination for the network and is referred to as the 'Network Termination' (NT). Within Europe and Australia, the NT physically exists as a small connection box usually attached to a wall etc, and it converts the two wire line (U interface) coming in from the network to four wires (S/T interface or S bus). The S/T interface allows up to eight items or 'terminal equipments' to be connected, although only two may be used at any time. The terminal equipments may be telephones, computers, etc, and they are connected in what is termed a point to point configuration. In Europe the ISDN line provides up to about 1 watt of power that enables the NT to be run, and also enables a basic ISDN phone to be used for emergency calls. In North America a slightly different approach may be adopted in that the terminal equipment may be directly connected to the network in a point to point configuration as this saves the cost of a network termination unit, but it restricts the flexibility. Additionally power is not normally provided.

PRI (Primary Rate Interface) - This configuration carries a greater number of channels than the Basic Rate Interface and has a D channel with a bandwidth of 64 kbps. The number of B channels varies according to the location. Within Europe and Australia a configuration of 30B+D has been adopted providing an aggregate data rate of 2.048 Mbps (E1). For North America and Japan, a configuration of 23B+1D has been adopted. This provides an aggregate data rate of 1.544 Mbps (T1).

The primary rate connections utilise four wires - a pair for each direction. They are normally 120 ohm balanced lines using twisted pair cable. Primary rate connections always use a point to point configuration.

Primary rate lines are widely used to conenct to Private Branch eXchanges (PBX) in an office etc. Typically this may be used to provide a number of POTS (Plain Old Telephone System) or basic rate ISDN lines to the users.

Calling a WiMAX winner

2006-10-07T09:38:00.000+08:00

The Mobile WiMAX race is on. There's still a question as to how big the potential market for the technology will be, but a lot of big vendors seem to be betting that it will be huge. Ever since Sprint announced its commitment to the technology, it looks like those predictions may pan out.

Over the next two weeks, leading into the WiMAX World conference in Boston the next month, Telephony will be examining the vendors that hope to make an impact on this new promising technology sector. With the help of Current Analysis analyst Peter Jarich, we've selected six vendors to profile -- each vendor for different reasons. There are some obvious choices like Motorola and Samsung, which have unquestionable momentum after being named primary vendors for Sprint's multimillion-dollar deployment, but there may be one or two surprises in the mix. Telephony isn't being so bold as to say that every vendor on the list will be a dominant force in WiMAX, and we're certainly not predicting which vendors of the six will wind up on top. But each of the vendors either stands a good chance of making its mark on the WiMAX industry--through momentum, technology or scale--or represents a sector of the WiMAX community that presents a challenge to the established order of telecom infrastructure players.

The series will kick off Wednesday with Motorola, which has an unquestionable lead in terms of mind share if not market share (remember, not a single vendor has yet recorded a dime of Mobile WiMAX revenues).
info by http://telephonyonline.com

Overview of EDGE

2006-10-04T12:54:00.000+08:00

EDGE is an enhancement to the GSM mobile cellular phone system. The name EDGE stands for Enhanced Data for GSM Evolution and it enables data to be sent over a GSM TDMA system at speeds up to 384 kbps. In some instances EDGE systems may also be known as EGPRS, or Enhanced General Packet Radio Service systems. Although strictly speaking a "2.5G" system, it is anticipated that it will be used to provide data services by operators who have not been able to secure the full 3G licences.

EDGE is intended to build on the enhancements provided by the addition of GPRS (General Packet Radio Service) where packet switching is applied to a network. It then enables a three fold increase in the speed at which data can be transferred by adopting a new form of modulation. GSM uses a form of modulation known as Gaussian Minimum Shift Keying (GMSK), but EDGE changes the modulation to 8PSK and thereby enabling a significant increase in data rate to be achieved.

Technical Overview
It is generally expected that EDGE will be applied to networks where the enhancements provided by GPRS have already been added. Under the original GSM system, a circuit would be allocated to a given user whether data was being transmitted or not. This was fine for voice communications because there would normally be some data present for most of the time. However this is not the case for data transmissions where high levels of data are transmitted in short bursts. TO make more efficient use of the available capability, packet switching is used. Here individual packets of data are routed to the user, enabling the channel or channels to be shared by several users.

To achieve this requires the addition of two additional nodes to the network, namely the Gateway GPRS Service Node (GGSN) and the Serving GPRS Service Node (SGSN). Here the GGSN connects to packet-switched networks such as the Internet and other GPRS networks. The SGSN provides the packet-switched link to mobile stations.

In terms of implementation EDGE systems require an EDGE transceiver unit to be added to each cell along with software upgrades to allow its use. This software upgrades may be implemented remotely. This change means that the inclusion of EDGE onto a network requires a significant investment in the infrastructure and as a result it is these upgrades will normally be implemented over a period of time. However GSM, GPRS and EDGE can all co-exist on the same network.

As both GPRS and EDGE represent significant upgrades to handsets and they are not just software upgrades, new mobile handsets are required.

Modulation
One of the key elements of EDGE is the form of modulation that is used. GPRS, being essentially a packet switched version of GSM uses GMSK, along with GSM itself. This form of modulation limits the data rate that can be transmitted over the air interface. EDGE uses a form of modulation known as 8 PSK. This is a form of phase shift keying where 8 phase states are used. The advantage is that it can transmit high data rates, although it is not as immune to interference and noise. The network therefore switches to 8PSK to allow the high data transfer rates when signal strengths are sufficient to permit the data transfer with a sufficiently low Bit Error Rate. By using 8PSK it is possible to transfer data at 48 kbps per channel rather than 9.6 kbps that is possible using GMSK. By allowing the use of multiple channels the technology allows the transfer of data at rates up to 384 kbps. However it should be remembered that these data transfer rates are only possible when the network is not highly loaded as access to all the channels would not be allowed.

Technical details CDMA

2006-10-02T10:18:00.000+08:00

Code Division Multiplexing (Synchronous CDMA)

Synchronous CDMA, also known as Code Division Multiplexing (CDM), exploits at its core mathematical properties of orthogonality. Suppose we represent data signals as vectors. For example, the binary string "1011" would be represented by the vector (1, 0, 1, 1). We may wish to give a vector a name, we may do so by using boldface letters, e.g. a. We also use an operation on vectors, known as the dot product, to "multiply" vectors, by summing the product of the components. The operation is denoted with a dot between the vectors. For example, the dot product of \mathbf a=(1, 0, 1, 1) and \mathbf b=(1, -1, -1, 0), written as \mathbf a\cdot \mathbf b, would be (1)\times(1)+(0)\times(-1)+(1)\times(-1)+(1)\times(0)=1+(-1)=0. For the special case when the dot product of two vectors is identically 0, the two vectors are said to be orthogonal to each other.

Example
An example of 4 orthogonal digital signals.
Enlarge
An example of 4 orthogonal digital signals.

Suppose now we have a set of vectors that are mutually orthogonal to each other. Usually these vectors are specially constructed for ease of decoding—they are columns or rows from Walsh matrices that are constructed from Walsh functions—but strictly mathematically the only restriction on these vectors is that they are orthogonal. An example of orthogonal functions is shown in the picture on the right. Now, associate with one sender a vector from this set, say v, which is called the chip code. Associate a zero digit with the vector -v, and a one digit with the vector v. For example, if v=(1,-1), then the binary vector (1, 0, 1, 1) would correspond to (1,-1,-1,1,1,-1,1,-1). For the purposes of this article, we call this constructed vector the transmitted vector.

Each sender has a different, unique vector chosen from that set, but the construction of the transmitted vector is identical.

Now, the physical properties of interference say that if two signals at a point are in phase, they will "add up" to give twice the amplitude of each signal, but if they are out of phase, they will "subtract" and give a signal that is the difference of the amplitudes. Digitally, this behaviour can be modelled simply by the addition of the transmission vectors, component by component. So, if we have two senders, both sending simultaneously, one with the chip code (1, -1) and data vector (1, 0, 1, 1), and another with the chip code (1, 1), and data vector (0,0,1,1), the raw signal received would be the sum of the transmission vectors (1,-1,-1,1,1,-1,1,-1)+(-1,-1,-1,-1,1,1,1,1)=(0,-2,-2,0,2,0,2,0).

Suppose a receiver gets such a signal, and wants to detect what the transmitter with chip code (1, -1) is sending. The receiver will make use of the property described in the above foundation section, and take the dot product to the received vector in parts. Take the first two components of the received vector, that is, (0, -2). Now, (0, -2).(1, -1) = (0)(1)+(-2)(-1) = 2. Since this is positive, we can deduce that a one digit was sent. Taking the next two components, (-2, 0), (-2, 0).(1,-1)=(-2)(1)+(0)(-1)=-2. Since this is negative, we can deduce that a zero digit was sent. Continuing in this fashion, we can successfully decode what the transmitter with chip code (1, -1) was sending: (1, 0, 1, 1).

Likewise, applying the same process with chip code (1, 1): (1, 1).(0,-2) = -2 gives digit 0, (1, 1).(-2,0)=(1)(-2)+(1)(0)=-2 gives digit 0, and so on, to give us the data vector sent by the transmitter with chip code (1, 1): (0, 0, 1, 1).

Asynchronous CDMA

The previous example of orthogonal Walsh sequences describes how 2 users can be multiplexed together in a synchronous system, a technique that is commonly referred to as Code Division Multiplexing (CDM). The set of 4 Walsh sequences shown in the figure will afford up to 4 users, and in general, an NxN Walsh matrix can be used to multiplex N users. Multiplexing requires all of the users to be coordinated so that each transmits their assigned sequence v (or the complement, -v) starting at exactly the same time. Thus, this technique finds use in base-to-mobile links, where all of the transmissions originate from the same transmitter and can be perfectly coordinated.

On the other hand, the mobile-to-base links cannot be precisely coordinated, particularly due to the mobility of the handsets, and require a somewhat different approach. Since it is not mathematically possible to create signature sequences that are orthogonal for arbitrarily random starting points, unique "pseudo-random" or "pseudo-noise" (PN) sequences are used in Asynchronous CDMA systems. These PN sequences are statistically uncorrelated, and the sum of a large number of PN sequences results in Multiple Access Interference (MAI) that is approximated by a Gaussian noise process (via the theorem of the "law of large numbers" in statistics). If all of the users are received with the same power level, then the variance (e.g., the noise power) of the MAI increases in direct proportion to the number of users.

All forms of CDMA use spread spectrum process gain to allow receivers to partially discriminate against unwanted signals. Signals with the desired chip code and timing are received, while signals with different chip codes (or the same spreading code but a different timing offset) appear as wideband noise reduced by the process gain.

Since each user generates MAI, controlling the signal strength is an important issue with CDMA transmitters. A CDM (Synchronous CDMA), TDMA or FDMA receiver can in theory completely reject arbitrarily strong signals using different codes, time slots or frequency channels due to the orthogonality of these systems. This is not true for Asynchronous CDMA; rejection of unwanted signals is only partial. If any or all of the unwanted signals are much stronger than the desired signal, they will overwhelm it. This leads to a general requirement in any Asynchronous CDMA system to approximately match the various signal power levels as seen at the receiver. In CDMA cellular, the base station uses a fast closed-loop power control scheme to tightly control each mobile's transmit power.

Advantages of Asynchronous CDMA over other techniques

Asynchronous CDMA's main advantage over CDM (Synchronous CDMA), TDMA and FDMA is that it can use the spectrum more efficiently in mobile telephony applications. TDMA systems must carefully synchronize the transmission times of all the users to ensure that they are received in the correct timeslot and do not cause interference. Since this cannot be perfectly controlled in a mobile environment, each timeslot must have a guard-time, which reduces the probability that users will interfere, but decreases the spectral efficiency. Similarly, FDMA systems must use a guard-band between adjacent channels, due to the random doppler shift of the signal spectrum which occurs due to the user's mobility. The guard-bands will reduce the probability that adjacent channels will interfere, but decrease the utilization of the spectrum.

Most importantly, Asynchronous CDMA offers a key advantage in the flexible allocation of resources. There are a fixed number of orthogonal codes, timeslots or frequency bands that can be allocated for CDM, TDMA and FDMA systems, which remain underutilized due to the bursty nature of telephony and packetized data transmissions. There is no strict limit to the number of users that can be supported in an Asynchronous CDMA system, only a practical limit governed by the desired bit error probability, since the SIR (Signal to Interference Ratio) varies inversely with the number of users. In a bursty traffic environment like mobile telephony, the advantage afforded by Asynchronous CDMA is that the performance (bit error rate) is allowed to fluctuate randomly, with an average value determined by the number of users times the percentage of utilization. Suppose there are 2N users that only talk half of the time, then 2N users can be accommodated with the same average bit error probability as N users that talk all of the time. The key difference here is that the bit error probability for N users talking all of the time is constant, whereas it is a random quantity (with the same mean) for 2N users talking half of the time.

In other words, Asynchronous CDMA is ideally suited to a mobile network where large numbers of transmitters each generate a relatively small amount of traffic at irregular intervals. CDM (Synchronous CDMA), TDMA and FDMA systems cannot recover the underutilized resources inherent to bursty traffic due to the fixed number of orthogonal codes, time slots or frequency channels that can be assigned to individual transmitters. For instance, if there are N time slots in a TDMA system and 2N users that talk half of the time, then half of the time there will be more than N users needing to use more than N timeslots. Furthermore, it would require significant overhead to continually allocate and deallocate the orthogonal code, time-slot or frequency channel resources. By comparison, Asynchronous CDMA transmitters simply send when they have something to say, and go off the air when they don't, keeping the same PN signature sequence as long as they are connected to the system.

Soft handoff

Soft handoff (or soft handover) is an innovation in mobility. It refers to the technique of adding additional base stations (in IS-95 as many as 5) to a connection to be certain that the next base is ready as you move through the terrain. However, it can also be used to move a call from one base station that is approaching congestion to another with better capacity. As a result, signal quality and handoff robustness is improved compared to TDMA systems.

In TDMA and analog systems, each cell transmits on its own frequency, different from those of its neighbouring cells. If a mobile device reaches the edge of the cell currently serving its call, it is told to break its radio link and quickly tune to the frequency of one of the neighbouring cells where the call has been moved by the network due to the mobile's movement. If the mobile is unable to tune to the new frequency in time the call is dropped.

In CDMA, a set of neighbouring cells all use the same frequency for transmission and distinguish cells (or base stations) by means of a number called the "PN offset", a time offset from the beginning of the well-known pseudo-random noise sequence that is used to spread the signal from the base station. Because all of the cells are on the same frequency, listening to different base stations is now an exercise in digital signal processing based on offsets from the PN sequence, not RF transmission and reception based on separate frequencies.

As the CDMA phone roams through the network, it detects the PN offsets of the neighbouring cells and reports the strength of each signal back to the reference cell of the call (usually the strongest cell). If the signal from a neighbouring cell is strong enough, the mobile will be directed to "add a leg" to its call and start transmitting and receiving to and from the new cell in addition to the cell (or cells) already hosting the call. Likewise, if a cell's signal becomes too weak the mobile is directed to drop that leg. In this way, the mobile can move from cell to cell and add and drop legs as necessary in order to keep the call up without ever dropping the link.

It should be noted that this "soft handoff" does not happen via CDMA from cell tower to cell tower. A group of cell sites are linked up with wire and the call is synced over wire, over TDM, ATM, or even IP.

When there are frequency boundaries between different carriers or sub-networks, a CDMA phone behaves in the same way as TDMA or analog and performs a hard handoff in which it breaks the existing connection and tries to pick up on the new frequency where it left off.

Code division multiple access

2006-10-02T10:08:00.000+08:00

Code division multiple access (CDMA) is a form of multiplexing (not a modulation scheme) and a method of multiple access that does not divide up the channel by time (as in TDMA), or frequency (as in FDMA), but instead encodes data with a special code associated with each channel and uses the constructive interference properties of the special codes to perform the multiplexing. CDMA also refers to digital cellular telephony systems that make use of this multiple access scheme, such as those pioneered by Qualcomm, and W-CDMA by the International Telecommunication Union or ITU.

CDMA has since been used in many communications systems, including the Global Positioning System (GPS) and in the OmniTRACS satellite system for transportation logistics.

Usage in mobile telephony

A number of different terms are used to refer to CDMA implementations. The original U.S. standard defined by QUALCOMM was known as IS-95, the IS referring to an Interim Standard of the Telecommunications Industry Association (TIA). IS-95 is often referred to as 2G or second generation cellular. The QUALCOMM brand name cdmaOne may also be used to refer to the 2G CDMA standard. The CDMA has been submitted for approval as a mobile air interface standard to the ITU International Telecommunication Union.

Whereas the Global System for Mobile Communications (GSM) standard is a specification of an entire network infrastructure, the CDMA interface relates only to the air interface—the radio part of the technology. For example GSM specifies an infrastructure based on internationally approved standard while CDMA allows each operator to provide the network features as it finds suited. On the air interface, the signalling suite (GSM: ISDN SS7) work has been progressing to harmonise these.

After a couple of revisions, IS-95 was superseded by the IS-2000 standard. This standard was introduced to meet some of the criteria laid out in the IMT-2000 specification for 3G, or third generation, cellular. It is also referred to as 1xRTT which simply means "1 times Radio Transmission Technology" and indicates that IS-2000 uses the same 1.25 MHz shared channel as the original IS-95 standard. A related scheme called 3xRTT uses three 1.25 MHz carriers for a 3.75 MHz bandwidth that would allow higher data burst rates for an individual user, but the 3xRTT scheme has not been commercially deployed. More recently, QUALCOMM has led the creation of a new CDMA-based technology called 1xEV-DO, or IS-856, which provides the higher packet data transmission rates required by IMT-2000 and desired by wireless network operators.

The QUALCOMM CDMA system includes highly accurate time signals (usually referenced to a GPS receiver in the cell base station), so cell phone CDMA-based clocks are an increasingly popular type of radio clock for use in computer networks. The main advantage of using CDMA cell phone signals for reference clock purposes is that they work better inside buildings, thus often eliminating the need to mount a GPS antenna on the outside of a building.

The US CDMA system is frequently confused with a similar but incompatible technology called Wideband Code Division Multiple Access (W-CDMA) which forms the basis of the W-CDMA air interface. The W-CDMA air interface is used in the global 3G standard UMTS and the Japanese 3G standard FOMA, by NTT DoCoMo and Vodafone; however, the CDMA family of US national standards (including cdmaOne and CDMA2000) are not compatible with the W-CDMA family of International Telecommunication Union (ITU) standards.

Another important application of CDMA — predating and entirely distinct from CDMA cellular — is the Global Positioning System, GPS.

Coverage and Applications

The size of a given cell depends on the power of the signal transmitted by the handset, the terrain, and the radio frequency being used. Various algorithms can reduce the noise introduced by variations in terrain, but require extra information be sent to validate the transfer. Hence, the radio frequency and power of the handset effectively determine the cell size. Long wavelengths need less energy to travel a given distance vs. short wavelengths, so lower frequencies generally result in greater coverage while higher frequencies result in shorter coverage. These characteristics are used by mobile network planners in determining the size and placement of the cells in the network. In cities, many small cells are needed; the use of high frequencies allows sites to be placed more-closely together, with more subscribers provided service. In rural areas with a lower density of subscribers, use of lower frequencies allows each site to provide broader coverage. (See also the Market situation section of GSM.)

Various companies use different variants of CDMA to provide fixed-line networks using Wireless local loop (WLL) technology. Since they can plan with a specific number of subscribers per cell in mind, and these are all stationary, this application of CDMA can be found in most parts of the world.

CDMA is suited for data transfer with bursty behaviour and where delays can be accepted. It is therefore used in Wireless LAN applications; the cell size here is 500 feet because of the high frequency (2.4 GHz) and low power. The suitability for data transfer is the reason for why W-CDMA seems to be "winning technology" for the data portion of third-generation (3G) mobile cellular networks.

Overview of Cellular Phone Carriers

2006-09-30T09:04:00.000+08:00

How many times have you heard of people spending hundreds of dollars on the latest and greatest cell phone only to be disappointed by the bad signal? Dropping calls is another very annoying occurrence with cell phones. You need to look very carefully into the cell phone carrier that you wish to sign up with. You will be signing a contract usually for one year so make sure it’s money well spent.

Who are the main cell phone carrier?

* AT&T
* T-Mobile
* Verizon Wireless
* Cingular
* Nextel
* ALLTEL
* Sprint PCS

The above carriers are only a hand full in and every expanding mobile world. All will approach with special offers and incentives with camera cell phones etc to sign you up. The positives are obvious. You get a free cell phone and maybe some extra minutes talk time but they get a customer for a year. Most carriers have good coverage but it is worth your while looking at your options.

If you rely on your cell phone for work such as sales reps or drivers etc you need to look into the roaming charges. Some people think the charges may only vary slightly from one carrier to another so why bother. This is a lazy approach and untrue. You could save yourself hundreds of dollars per year simply by looking around. You can check the rates out online on most of the carrier’s websites. Roaming rates can be expensive so look long and hard before you decide.

I don’t need to travel so roaming charges are not a worry to me:

If you are happy enough using your phone mainly from the house or just plodding around you are not going to have any concern of high charges for roaming, but there are other ways to save money and lots of it. Many people never think too much about the SMS Text messages they send. Yes it saves money rather than calling and it is fast and generally reliable, however, different carriers have different text rates. You might not think that one-cent saving in not much and rightly so but if you are a regular Text user you need look at the overall yearly saving. Most cell phone carrier companies will offer special saving incentives on SMS Text so look into it.

Where else can I save money?

The latest and the greatest, the camera phone is as popular as a DVD. Everywhere you look people seem to have them. Great fun and very handy for that special moment for when you only wished you had a camera but very I repeat very expensive you decide to send many pictures to friends and family. Here by looking at your different options you can save plenty of you hard earned dollars. All it takes is a quick look around the web or a phone call; most of the carriers have free toll numbers. Monthly service rental will also vary from one company to another.

Another Tip

With so many cellular phone stores around you will be spoiled for choice. Remember stores make commission so if you are in a large shopping mall the chances are that there are a number of different cell stores. Check out the different rates and you will see a difference. Money is not everything, going back to the start of this article you need to make sure that you have an exceptional signal. If you are going to be a loyal customer for a year or so you should expect nothing but the best back in service.

What if I already have my own cell phone?

This is not a problem. If you are out of contract with one of the cell phone carriers you are free to look around just like from the beginning. You can either use your own cell or take them up on their offers, as most will offer you a free cell phone as a new user to the network.

What if I want to terminate my contract before it has officially ended?

Look long and hard at your contract before you sign, especially the smaller print. All carriers have different clauses in their contract but if you want to terminate early there usually is a penalty charge of some sort. One way out of this is to get a prepaid cell phone where you have no contract. You are free to swap from one carrier to another as you please. Be aware prepaid cell phones are more expensive pre minute talk time and Text than if you where on a monthly fee.

Describing GPRS (General Packet Radio Service)

2006-09-30T08:46:00.000+08:00

GSM was the most successful second generation cellular telecommunications system, but the need for higher data rates spawned new developments to enable data to be transferred at much higher rates. The first system to make an impact on the market was GPRS. The letters GPRS stand for General Packet Radio System, and the system enables much higher data rates to be achieved.

GPRS became the first stepping-stone on the path between the second-generation GSM cell phone system and the W-CDMA / UMTS system. With GPRS offering data services with data rates up to 115 kbps, facilities such as web browsing and other services requiring data transfer became possible. Although some data could be transferred using GSM, the rate was too slow for real data applications.

Packet switching
The key element of GPRS is that it uses packet switched data rather than circuit switched data, and this technique makes much more efficient use of the available capacity. This is because most data transfer occurs in what is often termed a "bursty" fashion. The transfer occurs in short peaks, followed by breaks when there is little or no activity.

Using a traditional approach a circuit is switched permanently to a particular user. This is known as a circuit switched mode. In view of the bursty nature of data transfer it means that there are periods when it will not be carrying data.

To improve the situation the overall capacity can be shared between several users. To achieve this the data is split into packets and tags inserted into the packet to provide the destination address. Packets from several sources can then be transmitted over the link. As it is unlikely that the data burst for different users will occur all at the same time, by sharing the overall resource in this fashion, the channel, or combined channels can be used far more efficiently. This approach is known as packet switching, and it is at the core of many cellular data systems, and in this case GPRS.

Network
GPRS and GSM are able to operate alongside one another on the same network, and using the same base stations. However upgrades are needed. The network upgrades reflect many of those needed for 3G, and in this way the investment in converting a network for GPRS prepares the core infrastructure for later evolution to a 3G W-CDMA / UMTS.

The upgraded network, as described in later pages of this tutorial, has both the elements used for GSM as well as new entities that are used for the GPRS packet data service.

Mobiles
Not only does the network need to be upgraded for GPRS, but new GPRS mobiles are also required. It is not possible to upgrade an existing GSM mobile for use as a GPRS mobile, although GSM mobiles can be used for GSM speech on a network that also carries GPRS. To utilise GPRS new modes are required to enable it to transmit the data in the required format.

Network
Although designed to run alongside the GSM system, the core network structure updated for GPRS has several new elements added to enable it to carry the packet data. The network between the BSC and BTS is similar, but behind this there is a new infrastructure to support the packet data.

For GPRS, the data from the BSC is routed through what is termed a Serving GPRS Support Node (SGSN). This forms the gateway to the services within the network, and then a Gateway GPRS Support Node (GGSN) which forms the gateway to the outside world.

SGSN
The SGSN serves a number of functions for GPRS mobiles. It enables authentication to occur, and it then tracks the location of the mobile within the network, and ensures that the quality of service is to the required level.

For the network protocols there are two layers that are used and supported by GPRS, namely X25 and IP. In operation the protocols assign addresses (Packet Data Protocol or PDP addresses) to the devices in the network for the purpose of routing the data through the system. Thus the GGSN appears as a data gateway to the public packet network, and thus the fact that the users are mobiles cannot be seen.

In operation the mobile must attach itself to the SGSN and activate its PDP address. This address is supplied by the GGSN which is associated with the SGSN. As a result a mobile can only attach to one SGSN, although once assigned its address it can receive data from multiple GGSNs using multiple PDP addresses.

GPRS mobiles
Not all GPRS mobiles are designed to offer the same levels of service. As a result they are split into three basic categories according to their capabilities in terms of the ability to connect to GSM and GPRS facilities:

Class A: - This class describes mobile phones that can be connected to both GPRS and GSM services at the same time.
Class B: - These mobiles can be attached to both GPRS and GSM services but they can be used on only one service at a time. A Class B mobile can make or receive a voice call, or send and or receive a SMS message during a GPRS connection. During voice calls or texting the GPRS service is suspended but it is re-established when the voice call or SMS session is complete.
Class C: - This classification covers phones that can be attached to either GPRS or GSM services but user needs to switch manually between the two different types.

GPRS mobiles are also categorized by the data rates they can support. Within GSM there are eight time slots that can be used to provide TDMA, allowing multiple mobiles onto a single RF signal carrier. Within GPRS it is possible to use more than one slot to enable much higher data rates to be achieved when these are available. The different speed classes of the mobiles are dependent upon the number of slots that can be used in either direction. There are a total of 29 speed classes. Class one mobiles are able to send and receive in one slot in either direction, i.e. uplink and downlink, and class 29 mobiles are able to send and receive in all eight slots. The classes within these two limits are able to support sending and receiving in different combinations of uplink and downlink slots.

In order to accommodate the packet data within GPRS it has been necessary to develop the coding schemes. Additionally the layers based on the OSI system has become more important than it was for some of the previous systems and descriptions what are contained within these layers are found below.

GPRS coding
GPRS offers a number of coding schemes with different levels of error detection and correction. These are used dependent upon the radio frequency signal conditions and the requirements for the data being sent. These are given labels CS-1 to CS-4:

CS-1: This applies the highest level of error detection and correction. It is used in scenarios when interference levels are high or signal levels are low. By applying high levels of detection and correction, this prevents the data having to be re-sent too often. Although it is acceptable for many types of data to be delayed, for others there is a more critical time element. This level of detection and coding results in a half code rate, i.e. for every 12 bits that enter the coder, 24 bits result. It results in a throughput of 9.05 kbps actual throughput data rate.
CS-2: This error detection and coding scheme is for better channels. It effectively uses a 2/3 encoder and results in a real data throughput of 13.4 kbps which includes the RLC/MAC header etc.
CS-3: This effectively uses a 3/4 coder and results in a data throughput of 15.6 kbps.
CS-4: This scheme is used when the signal is high and interference levels are low. No correction is applied to the signal allowing for a maximum throughput of 21.4 kbps. If all eight slots were used then this would enable a data throughput of 171.2 kbps to be achieved.

In addition to the error detection and coding schemes, GPRS also employs interleaving techniques to ensure the effects of interference and spurious noise are reduced to a minimum. It allows the error correction techniques to be more effective as interleaving helps reduce the total corruption if a section of data is lost.

As blocks of 20 ms data are carried over four bursts, with a total of 456 bits of information, a total of either 181, 268, 312, or 428 bits of payload data are carried dependent upon the error detection and coding scheme chosen, i.e. from CS-1 to CS-4, respectively.

Layers
Software plays a very large part in the current cellular communications systems. To enable it to be sectioned into areas that can be addressed separately, the concept of layers has been developed. It is used in GSM and other cellular systems but as they become more data-centric, the idea takes a greater prominence. Often these are referred to as layers, 1, 2, and 3.

Layer 1 concerns the physical link between the mobile and the base station. This is often subdivided into two sub-layers, namely the Physical RF layer that includes the modulation and demodulation, and the Physical link layer that manages the responses and controls required for the operation of the RF link. These include elements such as error correction, interleaving and the correct assembly of the data, power control, and the like.

Above this are the Radio Link Control (RLC) and the Medium Access Control (MAC) layers. These organise the logical links between the mobile and the base station. They control the radio link access and they organise the logical channels that route the data to and from the mobile.

There is also the Logical Link Layer (LLC) that formats the data frames and is used to link the elements of the core network to the mobile.

GPRS physical channel
GPRS builds on the basic GSM structure. GPRS uses the same modulation and frame structure that is employed by GSM, and in this way it is an evolution of the GSM standard. Slots can be assigned dynamically by the BSC to GPRS calls dependent upon the demand, the remaining ones being used for GSM traffic.

There is a new data channel that is used for GPRS and it is called the Packet Data Channel (PDCH). The overall slot structure for this channel is the same as that used within GSM, having the same power profile, and timing advance attributes to overcome the different signal travel times to the base station dependent upon the distance the mobile is from the base station. This enables the burst to fit in seamlessly with the existing GSM structure.

Each burst of information for GPRS is 0.577 mS in length and is the same as that used in GSM. It also carries two blocks of 57 bits of information, giving a total of 114 bits per burst. It therefore requires four bursts to carry each 20 mS block of data, i.e. 456 bits of encoded data.

The BSC assigns PDCHs to particular time slots, and there will be times when the PDCH is inactive, allowing the mobile to check for other base stations and monitor their signal strengths to enable the network to judge when handover is required. The GPRS slot may also be used by the base station to judge the time delay using a logical channel known as the Packet Timing Advance Control Channel (PTCCT).

Channel allocation
Although GPRS uses only one physical channel (PDCH) for the sending of data, it employs several logical channels that are mapped into this to enable the GPRS data and facilities to be managed. As the data in GPRS is handled as packet data, rather than circuit switched data the way in which this is organised is very different to that on a standard GSM link. Packets of data are assigned a space within the system according to the current needs, and routed accordingly.

The MAC layer is central to this and there are three MAC modes that are used to control the transmissions. These are named fixed allocation, dynamic allocation, and extended dynamic allocation.

The fixed allocation mode is required when a mobile requires a data to be sent at a consistent data rate. To achieve this, a set of PDCHs are allocated for a given amount of time. When this mode is used there is no requirement to monitor for availability, and the mobile can send and receive data freely. This mode is used for applications such as video conferencing.

When using the dynamic allocation mode, the network allocates time slots as they are required. A mobile is allowed to transmit in the uplink when it sees an identifier flag known as the Uplink Status Flag (USF) that matches its own. The mobile then transmits its data in the allocated slot. This is required because up to eight mobiles can have potential access to a slot, but obviously only one can transmit at any given time.

A further form of allocation known as extended dynamic allocation is also available. Use of this mode allows much higher data rates to be achieved because it enables mobiles to transmit in more than one slot. When the USF indicates that a mobile can use this mode, it can transmit in the number allowed, thereby increasing the rate at which it can send data.

Logical channels
There is a variety of channels used within GPRS, and they can be set into groups dependent upon whether they are for common or dedicated use. Naturally the system does use the GSM control and broadcast channels for initial set up, but all the GPRS actions are carried out within the GPRS logical channels carried within the PDCH.

Broadcast channels:
Packet Broadcast Central Channel (PBCCH): This is a downlink only channel that is used to broadcast information to mobiles and informs them of incoming calls etc. It is very similar in operation to the BCCH used for GSM. In fact the BCCH is still required in the initial to provide a time slot number for the PBCCH. In operation the PBCCH broadcasts general information such as power control parameters, access methods and operational modes, network parameters, etc, required to set up calls.

Common control channels:
Packet Paging Channel (PPCH): This is a downlink only channel and is used to alert the mobile to an incoming call and to alert it to be ready to receive data. It is used for control signalling prior to the call set up. Once the call is in progress a dedicated channel referred to as the PACCH takes over.
Packet Access Grant Channel (PAGCH): This is also a downlink channel and it sends information telling the mobile which traffic channel has been assigned to it. It occurs after the PPCH has informed the mobile that there is an incoming call.
Packet Notification Channel (PNCH): This is another downlink only channel that is used to alert mobiles that there is broadcast traffic intended for a large number of mobiles. It is typically used in what is termed point-to-point multicasting.
Packet Random Access Channel (PRACH): This is an uplink channel that enables the mobile to initiate a burst of data in the uplink. There are two types of PRACH burst, one is an 8 bit standard burst, and a second one using an 11 bit burst has added data to allow for priority setting. Both types of burst allow for timing advance setting.

Dedicated control channels:
Packet Associated Control Channel (PACCH): This channel is present in both uplink and downlink directions and it is sued for control signalling while a call is in progress. It takes over from the PPCH once the call is set up and it carries information such as channel assignments, power control messages and acknowledgements of received data.
Packet Timing Advance Common Control Channel (PTCCH): This channel, which is present in both the uplink and downlink directions is used to adjust the timing advance. This is required to ensure that messages arrive at the correct time at the base station regardless of the distance of the mobile from the base station. As timing is critical in a TDMA system and signals take a small but finite time to travel this aspect is very important if long guard bands are not to be left.

Dedicated traffic channel:
Packet Data Traffic Channel (PDTCH): This channel is used to send the traffic and it is present in both the uplink and downlink directions. Up to eight PDTCHs can be allocated to a mobile to provide high speed data.

When looking at the way in which GPRS operates, it can be seen that there are three basic modes in which it operates. These are: initialisation / idle, standby, and ready.

Initialisation / idle
When the mobile is turned on it must register with the network and update the location register. This is very similar to that performed with a GSM mobile, but it is referred to as a location update. It first locates a suitable cell and transmits a radio burst on the RACH using a shortened burst because it does not know what timing advance is required. The data contained within this burst temporarily identifies the mobile, and indicates that the reason for the update is to perform a location update.

When the mobile performs its location update the network also performs an authentication to ensure that it is allowed to access the network. As for GSM it accesses the HLR and VLR as necessary for the location update and the AuC for authentication. It is at registration that the network detects that the mobile has a GPRS capability. The SGSN also maintains a record of the location of the mobile so that data can be sent there is required.

Standby
The mobile then enters a standby mode, periodically updating its position as required. It monitors the MNC of the base station to ensure that it has not changed base stations and also looks for stronger base station control channels.

The mobile will also monitor the PPCH in case of an incoming alert indicating that data is ready to be sent. As for GSM, most base stations set up a schedule for paging alerts based on the last figures of the mobile number. In this way it does not have to monitor all the available alert slots and can instead only monitor a reduced number where it knows alerts can be sent for it. In this way the receiver can be turned off for longer and battery life can be extended.

Ready
In the ready mode the mobile is attached to the system and a virtual connection is made with the SGSN and GGSN. By making this connection the network knows where to route the packets when they are sent and received. In addition to this the mobile is likely to use the PTCCH to ensure that its timing is correctly set so that it is ready for a data transfer should one be needed.

With the mobile attached to the network, it is prepared for a call or data transfer. To transmit data the mobile attempts a Packet Channel Request using the PRACH uplink channel. As this may be busy the mobile monitors the PCCCH which contains a status bit indicating the status of the base station receiver, whether it is busy or idle and capable of receiving data. When the mobile sees this status bit indicates the receiver is idle, it sends its packet channel request message. If accepted the base station will respond by sending an assignment message on the PAGCH on the downlink. This will indicate which channel the mobile is to use for its packet data transfer as well as other details required for the data transfer.

This only sets up the packet data transfers for the uplink. If data needs to be transferred in the downlink direction then a separate assignment is performed for the downlink channel.

When data is transferred this is controlled by the action of the MAC layer. In most instances it will operate in an acknowledge mode whereby the base station acknowledges each block of data. The acknowledgement may be contained within the data packets being sent in the downlink, or the base station may send data packets down purely to acknowledge the data.

When disconnecting the mobile will send a packet temporary block flow message, and this is acknowledged. Once this has taken place the USF assigned to the mobile becomes redundant and can be assigned to another mobile wanting access. With this the mobile effectively becomes disconnected and although still attached to the network no more data transfer takes place unless it is re-initiated. Separate messages are needed to detach the mobile from the network.

The migration headed the Data Network

2006-09-28T12:33:00.000+08:00

Most united communication could be carried out through various networks. With softswitch, the network more andal and economical without sacrificing the other network available.

Convergence between the network of the circuit (circuit networks) and the network of the package Cellular (packet network) was the evolution of multifunctional network technology in the future.
Eventually, the user of the telephone Cellular will communicate with many lines starting from when telephoning the house, the internet telephone in PC, telephoned the office, or conversely, with not only involved the voice, but also the data.

His process need not replace all the network of the available circuit, the migration cost of the cheap network, and in stages did upgraded headed the network of the package.
One of his methods was with technology softswitch.
This implement could connect between the network of the circuit and the network of the package, including inside was the network of the telephone continue to (PSTN), the based internet IP, the TV cable but also the network Cellular available uptil now.

Technical overview The GSM (Global System for Mobile Communications)

2006-09-26T14:09:00.000+08:00

The GSM system is the most widely used mobile telecommunications system in use in the world today. The letters GSM originally stood for the words Groupe Speciale Mobile, but as it became clear this standard was to be used world wide the meaning of GSM was changed to Global System for Mobile Communications. Since it was first deployed in 1991, the use of GSM has grown steadily, and it is now the most widely cell phone system in the world. GSM reached the 1 billion subscriber point in February 2004, and continued to grown in popularity.

System idea
The GSM system was designed as a second generation (2G) cellular communication system. One of the basic aims was to provide a system that would enable greater capacity to be achieved than the previous first generation analogue systems. GSM achieved this by using a digital TDMA (time division multiple access approach). By adopting this technique more users could be accommodated within the available bandwidth. In addition to this, ciphering of the digitally encoded speech was adopted to retain privacy. Using the earlier analogue systems it was possible for anyone with a scanner receiver to listen to calls and a number of famous personalities had been "eavesdropped" with embarrassing consequences.

Services provided
Speech or voice calls are obviously the primary function for the GSM system. To achieve this the speech is digitally encoded and later decoded using a vocoder. A variety of vocoders are available for use, being aimed at different scenarios.

In addition to the voice services, GSM supports a variety of other data services. Although their performance is nowhere near the level of those provided by 3G, they are nevertheless still important and useful. A variety of data services are supported with user data rates up to 9.6 kbps. Services including Group 3 facsimile, videotext and teletex can be supported.

One service that has grown enormously is the short message service. Developed as part of the GSM specification, it has also been incorporated into other cellular systems. It can be thought of as being similar to the paging service but is far more comprehensive allowing bi-directional messaging, store and forward delivery, and it also allows alphanumeric messages of a reasonable length. This service has become particularly popular, initially with the young as it provided a simple, low fixed cost.

Basic concept
The GSM system had a number of design aims when the development started. It should offer good subjective speech quality, have a low phone or terminal cost, terminals should be able to be handheld, the system should support international roaming, it should offer good spectral efficiency, and the system should offer ISDN compatibility.

The system that developed provided for all of these. The overall system definition for GSM describes not only the air interface but also the network. By adopting this approach it is possible to define the operation of the whole network to enable international roaming as well as enabling network elements from different manufacturers to operate alongside each other, although this last feature is not completely true, especially with older items.

GSM uses 200 kHz RF channels. These are time division multiplexed to enable up to eight users to access each carrier. In this way it is a TDMA / FDMA system.

The base transceiver stations (BTS) are organised into small groups, controlled by a base station controller (BSC) which is typically co-located with one of the BTSs. The BSC with its associated BTSs is termed the base station subsystem (BSS).

Further into the core network is the main switching area. This is known as the mobile switching centre (MSC). Associated with it is the location registers, namely the home location register (HLR) and the visitor location register (VLR) which track the location of mobiles and enable calls to be routed to them. Additionally there is the Authentication Centre (AuC), and the Equipment Identify Register (EIR) that are used in authenticating the mobile before it is allowed onto the network and for billing. The operation of these are explained in the following pages.

Last but not least is the mobile itself. Often termed the ME or mobile equipment, this is the item that the end user sees. One important feature that was first implemented on GSM was the use of a Subscriber Identity Module. This card carried with it the users identity and other information to allow the user to upgrade a phone very easily, while retaining the same identity on the network. It was also used to store other information such as "phone book" and other items. This item alone has allowed people to change phones very easily, and this has fuelled the phone manufacturing industry and enabled new phones with additional features to be launched. This has allowed mobile operators to increase their average revenue per user (ARPU) by ensuring that users are able to access any new features that may be launched on the network requiring more sophisticated phones.

Specification Summary of GSM Cell Phone System

Multiple Access Technology FDMA / TDMA

Duplex Technique FDD

Uplink frequency band 933 - 960 MHz
(basic 900 MHz band only)

Downlink frequency band 890 - 915 MHz
(basic 900 MHz band only)

Channel spacing 200 kHz

Modulation GMSK

Speech coding Various - Original was RPE-LTP/13

Speech channels per RF channel 8

Channel data rate 270.833 kbps

Frame duration 4.615 mS

The architecture of the GSM system with its hardware can broadly be grouped into three main areas: the mobile station, the base station subsystem, and the network subsystem. Each area performs its own functions and when used together they enable the full operational capability of the system to be realised.

Mobile station
Mobile stations (MS), mobile equipment (ME) or as they are most widely known, cell or mobile phones are the section of a GSM cellular network that the user sees and operates. In recent years their size has fallen dramatically while the level of functionality has greatly increased. A further advantage is that the time between charges has significantly increased.

There are a number of elements to the cell phone, although the two main elements are the main hardware and the SIM.

The hardware itself contains the main elements of the mobile phone including the display, case, battery, and the electronics used to generate the signal, and process the data receiver and to be transmitted. It also contains a number known as the International Mobile Equipment Identity (IMEI). This is installed in the phone at manufacture and "cannot" be changed. It is accessed by the network during registration to check whether the equipment has been reported as stolen.

The SIM or Subscriber Identity Module contains the information that provides the identity of the user to the network. It contains are variety of information including a number known as the International Mobile Subscriber Identity (IMSI).

Base station subsystem
The Base Station Subsystem (BSS) section of the GSM network is fundamentally associated with communicating with the mobiles on the network. It consists of two elements, namely the Base Transceiver Station (BTS) and the Base Station Controller (BSC).

The BTS used in a GSM network comprises the radio transmitter receivers, and their associated antennas that transmit and receive to directly communicate with the mobiles. The BTS is the defining element for each cell. The BTS communicates with the mobiles and the interface between the two is known as the Um interface with its associated protocols.

The BSC forms the next stage back into the GSM network. It controls a group of BTSs, and is often co-located with one of the BTSs in its group. It manages the radio resources and controls items such as handover within the group of BTSs, allocates channels and the like. It communicates with the BTSs over what is termed the Abis interface.

Network subsystem
The network subsystem contains a variety of different elements, and is often termed the core network. It provides the main control and interfacing for the whole mobile network. It includes elements including the MSC, HLR, VLR, Auc and more as described below:

The main element within the core network is the Mobile switching Services Centre (MSC). The MSC acts like a normal switching node within a PSTN or ISDN, but also provides additional functionality to enable the requirements of a mobile user to be supported. These include registration, authentication, call location, inter-MSC handovers and call routing to a mobile subscriber. It also provides an interface to the PSTN so that calls can be routed from the mobile network to a phone connected to a landline. Interfaces to other MSCs are provided to enable calls to be made to mobiles on different networks.

To enable the MSC to perform its functions it requires data from a number of databases. One is known as the Home Location Register (HLR). It contains all the administrative information about each subscriber along with their last known location.

When a user switches on their phone, the phone registers with the network and from this it is possible to determine which BTS it communicates with so that incoming calls can be routed appropriately. Even when the phone is not active (but switched on) it re-registers periodically to ensure that the network (HLR) is aware of its latest position.

There is one HLR per network, although it may be distributed across various sub-centres to for operational reasons.

Another of the databases is known as the Visitor Location Register (VLR). This contains selected information from the HLR that enables the selected services for the individual subscriber to be provided.

The VLR can be implemented as a separate entity, but it is commonly realised as an integral part of the MSC, rather than a separate entity. In this way access is made faster and more convenient.

The third register is the Equipment Identity Register (EIR). The EIR is the entity that decides whether a given mobile equipment may be allowed onto the network. Each mobile equipment has a number known as the International Mobile Equipment Identity. This number, as mentioned above, is installed in the equipment and is checked by the network during registration. Dependent upon the information held in the EIR, the mobile may be allocated one of three states - allowed onto the network, barred access, or monitored in case its problems.

The final register is the Authentication Centre (AuC). The AuC is a protected database that contains the secret key also contained in the user's SIM card. It is used for authentication and for ciphering on the radio channel.

Another element in the network is the Gateway Mobile Switching Centre (GMSC). The GMSC is the point to which a ME terminating call is initially routed, without any knowledge of the MS's location. The GMSC is thus in charge of obtaining the MSRN (Mobile Station Roaming Number) from the HLR based on the MSISDN (Mobile Station ISDN number, the "directory number" of a MS) and routing the call to the correct visited MSC. The "MSC" part of the term GMSC is misleading, since the gateway operation does not require any linking to an MSC.

The SMS-G or SMS gateway is the term that is used to collectively describe the two Short Message Services Gateways defined in the GSM standards. The two gateways handle messages directed in different directions. The SMS-GMSC (Short Message Service Gateway Mobile Switching Centre) is for short messages being sent to an ME. The SMS-IWMSC (Short Message Service Inter-Working Mobile Switching Centre) is used for short messages originated with a mobile on that network. The SMS-GMSC role is similar to that of the GMSC, whereas the SMS-IWMSC provides a fixed access point to the Short Message Service Centre.

There are a number of elements to the GSM radio or air interface. There are the aspects of the physical power levels, channels and the like. Additionally there are the different data channels that are employed to carry the data and exchange the protocol messages that enable the radio subsystem to operate correctly.

Basic signal characteristics
The GSM system uses digital TDMA technology combined with a channel bandwidth of 200 kHz. Accordingly the system is able to offer a higher level of spectrum efficiency that that which was achieved with the previous generation of analogue systems. As there are many carrier frequencies that are available, one or more can be allocated to each base station. The system also operates using Frequency Division Duplex and as a result, paired bands are needed for the up and downlink transmissions. The frequency separation is dependent upon the band in use.

The carrier is modulated using Gaussian Minimum Shift Keying (GMSK). GMSK was used for the GSM system because it is able to provide features required for GSM. It is resilient to noise when compared to some other forms of modulation, it occupies a relatively narrow bandwidth, and it has a constant power level.

The data transported by the carrier serves up to eight different users under the basic system. Even though the full data rate on the carrier is approximately 270 kbps, some of this supports the management overhead, and therefore the data rate allotted to each time slot is only 24.8 kbps. In addition to this error correction is required to overcome the problems of interference, fading and the like. This means that the available data rate for transporting the digitally encoded speech is 13 kbps for the basic vocoders.

Power levels
A variety of power levels are allowed by the GSM standard, the lowest being only 800 mW (29 dBm). As mobiles may only transmit for one eighth of the time, i.e. for their allocated slot which is one of eight, the average power is an eighth of the maximum.

Additionally, to reduce the levels of transmitted power and hence the levels of interference, mobiles are able to step the power down in increments of 2 dB from the maximum to a minimum 13 dBm (20 milliwatts). The mobile station measures the signal strength or signal quality (based on the Bit Error Rate), and passes the information to the BTS and hence to the BSC, which ultimately decides if and when the power level should be changed.

A further power saving and interference reducing facility is the discontinuous transmission (DTx) capability that is incorporated within the specification. It is particularly useful because there are long pauses in speech, for example when the person using the mobile is listening, and during these periods there is no need to transmit a signal. In fact it is found that a person speaks for less than 40% of the time during normal telephone conversations. The most important element of DTx is the Voice Activity Detector. It must correctly distinguish between voice and noise inputs, a task that is not trivial. If a voice signal is misinterpreted as noise, the transmitter is turned off an effect known as clipping results and this is particularly annoying to the person listening to the speech. However if noise is misinterpreted as a voice signal too often, the efficiency of DTX is dramatically decreased.

It is also necessary for the system to add background or comfort noise when the transmitter is turned off because complete silence can be very disconcerting for the listener. Accordingly this is added as appropriate. The noise is controlled by the SID (silence indication descriptor).

Multiple access and channel structure
GSM uses a combination of both TDMA and FDMA techniques. The FDMA element involves the division by frequency of the (maximum) 25 MHz bandwidth into 124 carrier frequencies spaced 200 kHz apart as already described.

The carriers are then divided in time, using a TDMA scheme. The fundamental unit of time is called a burst period and it lasts for approximately 0.577 mS (15/26 mS). Eight of these burst periods are grouped into what is known as a TDMA frame. This lasts for approximately 4.615 ms (i.e.120/26 ms) and it forms the basic unit for the definition of logical channels. One physical channel is one burst period allocated in each TDMA frame.

There are different types of frame that are transmitted to carry different data, and also the frames are organised into what are termed multiframes and superframes to provide overall synchronisation.

GSM uses a variety of channels in which the data is carried. In GSM, these channels are separated into physical channels and logical channels. The Physical channels are determined by the timeslot, whereas the logical channels are determined by the information carried within the physical channel. It can be further summarised by saying that several recurring timeslots on a carrier constitute a physical channel. These are then used by different logical channels to transfer information. These channels may either be used for user data (payload) or signalling to enable the system to operate correctly.

Common and dedicated channels
The channels may also be divided into common and dedicated channels. The forward common channels are used for paging to inform a mobile of an incoming call, responding to channel requests, and broadcasting bulletin board information. The return common channel is a random access channel used by the mobile to request channel resources before timing information is conveyed by the BSS.

The dedicated channels are of two main types: those used for signalling, and those used for traffic. The signalling channels are used for maintenance of the call and for enabling call set up, providing facilities such as handover when the call is in progress, and finally terminating the call. The traffic channels handle the actual payload.

The following logical channels are defined in GSM:

TCHf - Full rate traffic channel.

TCH h - Half rate traffic channel.

BCCH - Broadcast Network information, e.g. for describing the current control channel structure. The BCCH is a point-to-multipoint channel (BSS-to-MS).

SCH - Synchronisation of the MSs.

FCHMS - frequency correction.

AGCH - Acknowledge channel requests from MS and allocate a SDCCH.

PCHMS - terminating call announcement.

RACHMS - access requests, response to call announcement, location update, etc.

FACCHt - For time critical signalling over the TCH (e.g. for handover signalling). Traffic burst is stolen for a full signalling burst.

SACCHt - TCH in-band signalling, e.g. for link monitoring.

SDCCH - For signalling exchanges, e.g. during call setup, registration / location updates.

FACCHs - FACCH for the SDCCH. The SDCCH burst is stolen for a full signalling burst. Function not clear in the present version of GSM (could be used for e.g. handover of an eight-rate channel, i.e. using a "SDCCH-like" channel for other purposes than signalling).

SACCHs - SDCCH in-band signalling, e.g. for link monitoring.

If digitised in a linear fashion, the speech would occupy a far greater bandwidth than any cellular system and in this case the GSM system would be able to accommodate. To overcome this, a variety of voice coding systems or vocoders are used. These systems involve analysing the incoming data that represents the speech and then performing a variety of actions upon it to reduce the data rate. At the receiving end the reverse process is undertaken to re-constitute the speech data so that it can be understood. In GSM a variety of vocoders are used, including LPC-RPE, EFR, etc as described in the following paragraphs.

The vocoder that was originally used in the GSM system was the LPC-RPE (Linear Prediction Coding with Regular Pulse Excitation) vocoder. This vocoder took each 20 mS block of speech and then represented it using just 260 bits. This actually equates to a data rate of 13 kbps.

In GSM it is recognised that some bits are more important than others. If some bits are missed or corrupted, it is more important to the voice quality than others. Accordingly the different bits are classified:

Class Ia 50 bits - most important and sensitive to bit errors
Class Ib 132 bits - moderately sensitive to bit errors
Class II 78 bits - least sensitive to bit errors

The 50 Class 1a bits are given a 3 bit Cyclic Redundancy Code (CRC) so that errors can be detected. This makes a total length of 53 bits. If there are any errors, the frame is not used, and it is discarded. In its place a version of the previously correctly received frame is used. These 53 bits, together with the 132 Class Ib bits with a 4 bit tail sequence, are entered into a 1/2 rate convolutional encoder. The total length is 189 bits. The encoder encodes each of the bits that enter as two bits, the output also being dependent upon a combination of the previous 4 input bits. As a result the output from the convolutional encoder consists of 378 bits. The remaining 78 Class II bits are considered the least sensitive to errors and they are not protected and simply added to the data. In this way every 20 ms speech sample generates a total of 456 bits. Accordingly the overall bit rate is 22.8 kbps. Once in this format the data is interleaved to add further protection against interference and noise.

The 456 bits output by the convolutional encoder are divided into 8 blocks of 57 bits, and these blocks are transmitted in eight consecutive time-slots, i.e. a total of four bursts as each burst takes two sets of data.

Later another vocoder called the Enhanced Full Rate (EFR) vocoder was added in response to the poor quality perceived by the users. This new vocoder gave much better sound quality and was adopted by GSM. Using the ACELP (Algebraic Code Excitation Linear Prediction) compression technology it gave a significant improvement in quality over the original LPC-RPE encoder. It became possible as the processing power that was available increased in mobile phones as a result of higher levels of processing power combined with their lower current consumption.

There is also a half rate vocoder. Although this gives much inferior voice quality, it does allow for an increase in network capacity. It is used in some instances when network loading is very high to accommodate all the calls.

System GSM Cell Phone

2006-09-23T09:30:00.000+08:00

The GSM cell or mobile phone system is the most popular in the world. GSM handsets are widely available at good prices and the networks are robust and reliable. The GSM system is also feature-rich with applications such as SMS text messaging, international roaming, SIM cards and the like. It is also being enhanced with technologies including GPRS and EDGE. To achieve this level of success has taken many years and is the result of both technical development and international cooperation.

The first cell phone systems that were developed were analogue systems. Typically they used frequency-modulated carriers for the voice channels and data was carried on a separate shared control channel. When compared to the systems employed today these systems were comparatively straightforward and as a result a vast number of systems appeared. Two of the major systems that were in existence were the AMPS (Advanced Mobile Phone System) that was used in the USA and many other countries and TACS (Total Access Communications System) that was used in the UK as well as many other countries around the world.

Another system that was employed, and was in fact the first system to be commercially deployed was the Nordic Mobile Telephone system (NMT). This was developed by a consortium of companies in Scandinavia and proved that international cooperation was possible.
The success of these systems proved to be their downfall. The use of all the systems installed around the globe increased dramatically and the effects of the limited frequency allocations were soon noticed. To overcome these a number of actions were taken. A system known as E-TACS or Extended-TACS was introduced giving the TACS system further channels. In the USA another system known as Narrowband AMPS (NAMPS) was developed.

New approaches
Neither of these approaches proved to be the long-term solution as more efficient systems were required. With the experience gained from the NMT system, showing that it was possible to develop a system across national boundaries, and with the political situation in Europe lending itself to international cooperation it was decided to develop a new Pan-European System. Furthermore it was realized that economies of scale would bring significant benefits. This was the beginnings of the GSM system.

To achieve the basic definition of a new system a meeting was held in 1982 under the auspices of the Conference of European Posts and Telegraphs (CEPT). They formed a study group called the Groupe Special Mobile ( GSM ) to study and develop a pan-European public land mobile system. Several basic criteria that the new system would have to meet were set down for the new GSM system to meet. These included: good subjective speech quality, low terminal and service cost, support for international roaming, ability to support handheld terminals, support for range of new services and facilities, spectral efficiency, and finally ISDN compatibility.

With the levels of under-capacity being projected for the analogue systems, this gave a real sense of urgency to the GSM development. Although decisions about the system were not taken at an early stage, all had been working toward a digital system. This decision was finally made in February 1987. This gave a variety of advantages. Greater levels of spectral efficiency could be gained, and in addition to this the use of digital circuitry would allow for higher levels of integration in the circuitry. This in turn would result in cheaper handsets with more features. Nevertheless significant hurdles still needed to be overcome. For example, many of the methods for encoding the speech within a sufficiently narrow bandwidth needed to be developed, and this posed a significant risk to the project. Nevertheless the GSM system had been started.

Launch dates
Work continued and a launch date for the new GSM system of 1991 was set for an initial launch of a service with limited coverage and capability to be followed by a complete roll out of the service in major European cities by 1993 and linking of the areas by 1995.

Meanwhile technical development was taking place. Initial trials had shown that time division multiple access techniques offered the best performance with the technology that would be available. This approach had the support of the major manufacturing companies which would ensure that with them on board sufficient equipment both in terms of handsets, base stations and the network infrastructure for GSM would be available.

Further impetus was given to the GSM project when in 1989 the responsibility was passed to the newly formed European Telecommunications Standards Institute (ETSI). Under the auspices of ETSI the specification took place. It provided functional and interface descriptions for each of the functional entities defined in the system. The aim was to provide sufficient guidance for manufacturers that equipment from different manufacturers would be interoperable, while not stopping innovation. The result of the specification work was a set of documents extending to more than 6000 pages. Nevertheless the resultant phone system provided a robust, feature-rich system. The first roaming agreement was signed between Telecom Finland and Vodafone in the UK. Thus the vision of a pan-European network was fast becoming a reality. However this took place before any networks went live.

The aim to launch GSM by 1991 proved to be a target that was too tough to meet. Terminals started to become available in mid 1992 and the real launch took place in the latter part of that year. With such a new service many were sceptical as the analogue systems were still in widespread use. Nevertheless by the end of 1993 GSM had attracted over a million subscribers and there were 25 roaming agreements in place. The growth continued and the next million subscribers were soon attracted.

Global usage
Originally GSM had been planned as a European system. However the first indication that the success of GSM was spreading further a field occurred when the Australian network provider, Telstra signed the GSM Memorandum of Understanding.

Frequencies
Originally it had been intended that GSM would operate on frequencies in the 900 MHz cellular band. In September 1993, the British operator Mercury One-to-One launched a network. Termed DCS 1800 it operated at frequencies in a new 1800 MHz band. By adopting new frequencies new operators and further competition was introduced into the market apart from allowing additional spectrum to be used and further increasing the overall capacity. This trend was followed in many countries, and soon the term DCS 1800 was dropped in favour of calling it GSM as it was purely the same system but operating on a different frequency band. In view of the higher frequency used the distances the signals travelled was slightly shorter but this was compensated for by additional base stations.

In the USA as well a portion of spectrum at 1900 MHz was allocated for cellular usage in 1994. The licensing body, the FCC, did not legislate which technology should be used, and accordingly this enabled GSM to gain a foothold in the US market. This system was known as PCS 1900 (Personal Communication System).

A great success
With GSM being used in many countries outside Europe this reflected the true nature of the name which had been changed from Groupe Special Mobile to Global System for Mobile communications. The number of subscribers grew rapidly and by the beginning of 2004 the total number of GSM subscribers reached 1 billion. Attaining this figure was celebrated at the Cannes 3GSM conference held that year.

Modulation Concept "Orthogonal Frequency Division Multiplex"

2006-09-22T11:36:00.000+08:00

Orthogonal Frequency Division Multiplex, the modulation concept being used for many radio and wireless applications from DAB, DVB, Wi-Fi and Mobile Video.
Orthogonal Frequency Division Multiplex or OFDM is a modulation format that is finding increasing levels of use in today's communications scene. OFDM has been adopted in the Wi-Fi arena where the 802.11a standard uses it to provide data rates up to 54 Mbps in the 5 GHz ISM (Industrial, Scientific and Medical) band. In addition to this the recently ratified 802.11g standard has it in the 2.4 GHz ISM band. If this was not enough it is also being used for digital terrestrial television transmissions as well as DAB digital radio. A new form of broadcasting called Digital Radio Mondiale for the long medium and short wave bands is being launched and this has also adopted COFDM. Then for the future it is being proposed as the modulation technique for fourth generation cell phone systems that are in their early stages of development and OFDM is also being used for many of the proposed mobile phone video systems.

OFDM concept
An OFDM signal consists of a number of closely spaced modulated carriers. When modulation of any form - voice, data, etc. is applied to a carrier, then sidebands spread out either side. It is necessary for a receiver to be able to receive the whole signal to be able to successfully demodulate the data. As a result when signals are transmitted close to one another they must be spaced so that the receiver can separate them using a filter and there must be a guard band between them. This is not the case with OFDM. Although the sidebands from each carrier overlap, they can still be received without the interference that might be expected because they are orthogonal to each another. This is achieved by having the carrier spacing equal to the reciprocal of the symbol period.

Traditional view of receiving signals carrying modulation

To see how OFDM works, it is necessary to look at the receiver. This acts as a bank of demodulators, translating each carrier down to DC. The resulting signal is integrated over the symbol period to regenerate the data from that carrier. The same demodulator also demodulates the other carriers. As the carrier spacing equal to the reciprocal of the symbol period means that they will have a whole number of cycles in the symbol period and their contribution will sum to zero - in other words there is no interference contribution.

OFDM Spectrum

One requirement of the OFDM transmitting and receiving systems is that they must be linear. Any non-linearity will cause interference between the carriers as a result of inter-modulation distortion. This will introduce unwanted signals that would cause interference and impair the orthogonality of the transmission.

In terms of the equipment to be used the high peak to average ratio of multi-carrier systems such as OFDM requires the RF final amplifier on the output of the transmitter to be able to handle the peaks whilst the average power is much lower and this leads to inefficiency. In some systems the peaks are limited. Although this introduces distortion that results in a higher level of data errors, the system can rely on the error correction to remove them.

Data
The data to be transmitted on an OFDM signal is spread across the carriers of the signal, each carrier taking part of the payload. This reduces the data rate taken by each carrier. The lower data rate has the advantage that interference from reflections is much less critical. This is achieved by adding a guard band time or guard interval into the system. This ensures that the data is only sampled when the signal is stable and no new delayed signals arrive that would alter the timing and phase of the signal.

Guard Interval

The distribution of the data across a large number of carriers in the OFDM signal has some further advantages. Nulls caused by multi-path effects or interference on a given frequency only affect a small number of the carriers, the remaining ones being received correctly. By using error-coding techniques, which does mean adding further data to the transmitted signal, it enables many or all of the corrupted data to be reconstructed within the receiver. This can be done because the error correction code is transmitted in a different part of the signal. It is this error coding which is referred to in the "Coded" word in the title of COFDM which is often seen.

Other variants
Flash OFDM - This is a variant that was developed by Flarion and it is a fast hopped form of OFDM. It uses multiple tones and fast hopping to spread signals over a given spectrum band.

VOFDM - Vector OFDM. This form of OFDM uses the concept of MIMO technology. It is being developed by CISCO Systems. MIMO stands for Multiple Input Multiple output and it uses multiple antennas to transmit and receive the signals so that multi-path effects can be utilised to enhance the signal reception and improve the transmission speeds that can be supported.

WOFDM - Wideband OFDM. The concept of this form of OFDM is that it uses a degree of spacing between the channels that is large enough that any frequency errors between transmitter and receiver do not affect the performance. It is particularly applicable to Wi-Fi systems.

OFDM and COFDM have gained a significant presence in the wireless market place. The combination of high data capacity, high spectral efficiency, and its resilience to interference as a result of multi-path effects means that it is ideal for the high data applications that are becoming a common factor in today's communications scene.

CDMA Basics

2006-09-21T14:13:00.000+08:00

CDMA or Code Division Multiple Access is now in widespread use for mobile or cell phone (cellular telecommunications) systems around the world. It was first used for the IS-95 mobile phone system also known by the trade name cdmaOne, and in its later 3G developments as CDMA2000. CDMA is also being used in the other major 3G cell phone system, Wideband-CDMA system originally called UMTS.

CDMA technology is based on a form of transmission known as Direct Sequence Spread Spectrum (DSSS). This form of transmission originally used for military and police communications because the transmissions were difficult to detect in many instances, and even if they were received they were very difficult to decipher without the correct codes. However the possibilities of using this technology to provide a multiple access scheme for mobile telecommunications and have now been exploited in a major way.

Previous cellular telecommunications technologies used either frequency division multiple access (FDMA) where different users were allocated different frequencies, or time division multiple access (TDMA) where they were allotted different time slots on a channel. CDMA is different. Using the CDMA system, different users are allocated different codes to provide access to the system. It can be likened to many different people standing in a room talking to others in many different languages. Although the ambient noise level is very high, it is nevertheless still possible to pick out someone speaking in the same language as yourself.

DSSS basics
The key element of code division multiple access CDMA is its use of DSSS. In essence the required data signal is multiplied with what is known as a spreading or chip code data stream. This has a higher data rate than the data itself and it enables the overall signal to be spread over a much wider bandwidth. Signals with high data rates occupy wider signal bandwidths than those with low data rates.

To decode the signal and receive the original data, the CDMA signal is multiplied with the spreading code to regenerate the original data. When this is done, then only the data with that was generated with the same spreading code is regenerated, all the other data that is generated from different spreading code streams is ignored

This is a powerful principle and using code division multiple access technique, it is possible to transmit several sets of data independently on the same carrier and then reconstitute them at the receiver without mutual interference. In this way a base station can communicate with several mobiles on a single channel. Similarly several mobiles can communicate with a single base station, provided that in each case an independent spreading code is used.

The CDMA spreading codes can either be a random number (or pseudo random), or more usually orthogonal codes are used. Two codes are said to be orthogonal if when they are multiplied together and then the result is added over a period of time they sum to zero. For example a codes 1 -1 -1 1 and 1 -1 1 -1 when multiplied together give 1 1 -1 -1 which gives the sum zero. Although pseudo random number codes can be used there is possibility of data errors being introduced into the system.

Advantages
There are several advantages to using code division multiple access CDMA. The main reason for its acceptance is that it enables more users to use a given amount of spectrum. Its use also enables adjacent base stations to operate on the same channel, allowing more efficient use of the spectrum and it provides for an easier handover.
In view of these advantages CDMA has been adopted for all the 3G technologies and will be around for many years to come.

Basics " Handover and Handoff "

2006-09-20T08:58:00.000+08:00

The concept of a cellular phone system is that it has a large number base stations covering a small area (cells), and as a result frequencies are able to be re-used. Cell phone systems also provide mobility. As a result it is a very basic requirement of the system that as the mobile handset moves out of one cell to the next, it must be possible to hand the call over from the base station of the first cell, to that of the next with no discernable disruption to the call. There are two terms for this process: handover is used within Europe, whereas handoff is the term used in North America.

The handover or handoff process is of major importance within any cellular telecommunications network. It is necessary to ensure it can be performed reliably and without disruption to any calls. Failure for it to perform reliably can result in dropped calls, and this is one of the key factors that can lead to customer dissatisfaction, which in turn may lead to them changing to another cellular network provider. Accordingly handover or handoff is one of the key performance indicators monitored so that a robust handover / handoff regime is maintained on the cellular network.

Handover basics
Although the concept of handover or handoff is relatively straightforward, it is not an easy process to implement in reality. The cellular network needs to decide when handover or handoff is necessary, and to which cell. Also when the handover occurs it is necessary to re-route the call to the relevant base station along with changing the communication between the mobile and the base station to a new channel. All of this needs to be undertaken without any noticeable interruption to the call. The process is quite complicated, and in early systems calls were often lost if the process did not work correctly.

Different cellular standards handle hand over / handoff in slightly different ways. Therefore for the sake of an explanation the example of the way that GSM handles handover is given.

There are a number of parameters that need to be known to determine whether a handover is required. The signal strength of the base station with which communication is being made, along with the signal strengths of the surrounding stations. Additionally the availability of channels also needs to be known. The mobile is obviously best suited to monitor the strength of the base stations, but only the cellular network knows the status of channel availability and the network makes the decision about when the handover is to take place and to which channel of which cell.

Accordingly the mobile continually monitors the signal strengths of the base stations it can hear, including the one it is currently using, and it feeds this information back. When the strength of the signal from the base station that the mobile is using starts to fall to a level where action needs to be taken the cellular network looks at the reported strength of the signals from other cells reported by the mobile. It then checks for channel availability, and if one is available it informs this new cell to reserve a channel for the incoming mobile. When ready, the current base station passes the information for the new channel to the mobile, which then makes the change. Once there the mobile sends a message on the new channel to inform the network it has arrived. If this message is successfully sent and received then the network shuts down communication with the mobile on the old channel, freeing it up for other users, and all communication takes place on the new channel.

Under some circumstances such as when one base transceiver station is nearing its capacity, the network may decide to hand some mobiles over to another base transceiver station they are receiving that has more capacity, and in this way reduce the load on the base transceiver station that is nearly running to capacity. In this way access can be opened to the maximum number of users. In fact channel usage and capacity are very important factors in the design of a cellular network.

Types of handover / handoff
With the advent of CDMA systems where the same channels can be used by several mobiles, and where it is possible to adjacent cells or cell sectors to use the same frequency channel there are a number of different types of handover that can be performed:

* Hard handover

* Soft handover

* Softer handover

Although all of these forms of handover or handoff enable the cellular phone to be connected to a different cell or different cell sector, they are performed in slightly different ways and are available under different conditions.

Hard handover
The definition of a hard handover or handoff is one where an existing connection must be broken before the new one is established. One example of hard handover is when frequencies are changed. As the mobile will normally only be able to transmit on one frequency at a time, the connection must be broken before it can move to the new channel where the connection is re-established. This is often termed and inter-frequency hard handover. While this is the most common form of hard handoff, it is not the only one. It is also possible to have intra-frequency hard handovers where the frequency channel remains the same.

Although there is generally a short break in transmission, this is normally short enough not to be noticed by the user.

Soft hand over
The new 3G technologies use CDMA where it is possible to have neighbouring cells on the same frequency and this opens the possibility of having a form of handover or handoff where it is not necessary to break the connection. This is called soft handover or soft handoff, and it is defined as a handover where a new connection is established before the old one is released. In UMTS most of the handovers that are performed are intra-frequency soft handovers.

Softer handover
The third type of hand over is termed a softer handover, or handoff. In this instance a new signal is either added to or deleted from the active set of signals. It may also occur when a signal is replaced by a stronger signal from a different sector under the same base station. This type of handover or handoff is available within UMTS as well as CDMA2000.

Handover and handoff are performed by all cellular telecommunications networks, and they are a core element of the whole concept of cellular telecommunications. There are a number of requirements for the process. The first is that it occurs reliably and if it does not, users soon become dissatisfied and choose another network provider in a process known as "churn". However it needs to be accomplished in the most efficient manner. Although softer handoff is the most reliable, it also uses more network capacity. The reason for this is that it is communicating with more than one sector or base station at any given instance. Soft handover is also less efficient than hard handover, but again more reliable as the connection is never lost.
It is therefore necessary for the cellular telecommunications network provider to arrange the network to operate in the most efficient manner, while still providing the most reliable service.

Wireless LAN Standards

2006-09-18T08:42:00.000+08:00

(1)Wireless LAN 802.11b
he IEEE 802.11b specification builds on the original IEEE 802.11 standard by providing up to 11Mbps data rates in a direct sequence spread spectrum air interface using Barker codes for spreading 1 and 2 Mbps data rates and CCK modulation for 5.5Mbps and 11Mbps data rates. Nuntius has expertise in WLAN technologies and can provide solutions on multiple platforms. The diagram below shows an 802.11b solution that includes RF, analog baseband, digital baseband (with the PHY), and the MAC. For more information, contact Nuntius Systems, Inc.

Features
2.4GHz ISM Frequency Band
AGC
Antenna Diversity
Rake Receiver Functionality

Data Rates
1 Mbps
2 Mbps
5.5 Mbps
11 Mbps

Modulation
DBPSK
DQPSK
CCK
PBCC (Optional)

(2)Wireless LAN 802.11a
The IEEE 802.11a specification provides higher data rates (up to 54Mbps) in the 5GHz frequency band. It also employs a spreading technique known as Orthogonal Frequency Division Multiplexing (OFDM). Nuntius has expertise in these WLAN technologies and can provide solutions on multiple platforms. The diagram below shows an 802.11a solution that includes RF, analog baseband, digital baseband (with the PHY), and the MAC. For more information, contact Nuntius Systems, Inc.

Features
5 GHz Frequency Band
OFDM
AGC
Channel Coding Supported (K=7)
Carrier Recovery
Data Puncturing

Data Rates
6 Mbps
9 Mbps
12 Mbps
24 Mbps
36 Mbps
48 Mbps
54 Mbps

Modulation
BPSK
QPSK
16-QAM
64-QAM

Wireless LAN Technology

2006-09-18T08:33:00.000+08:00

The earliest wireless networking products came to market about a decade ago, operated in the 900 MHz band. Because these were proprietary designs, an effort soon ensued to pursue a vendor-independent standard, to promote interoperability. This resulted in the formation of the IEEE 802.11 committee, which quickly began to focus on the 2.4-GHz band WLAN. The approval of the 2.4-GHz 802.11 standard was finally achieved in June 1997. The WLAN market finally gained acceptance as a legitimate enterprise technology in 2000 and is now gaining momentum. With the release of Wi-Fi, IEEE 802.11b standard products from several prominent network equipment suppliers have driven WLAN gear into wider acceptance. The market continues to thrive as suppliers unveil new products with higher speeds, increased interoperability and lower prices.

The original 802.11 specification identified 1Mbps and 2Mbps data speeds in a variety of physical medium access methods for the 2.4GHz ISM frequency band. These physical layer modulation methods included frequency hopping spread spectrum (FHSS) and direct sequence spread spectrum (DSSS). This was followed by the 802.11b specification that added 5.5Mbps and 11Mbps data rates using CCK modulation. IEEE also released a physical layer implementation that uses the 5 GHz band supporting data rates up to 54Mbps called 802.11a. The spreading scheme used in this specification is Orthogonal Frequency Division Multiplexing (OFDM). Other standards are being proposed that will augment these standards well into the future.

Building on their established expertise in spread spectrum communications systems, Nuntius has the technology to build Wireless LAN solutions for today's markets as well as the future markets.

Wireline Communications Technology

2006-09-16T08:27:00.000+08:00

Internet telecom began as a revolution in technology. The concept of sending voice across a data network, in packet format, has been nurtured by a hope of eventually creating new markets, but for a long time that hope remained latent.

Today customers are demanding a different kind of communications infrastructure to line up more closely with the different kind of business environment (shaped by the Internet) in which they find themselves operating. Likewise, service providers are beginning to come forth with new ideas about how to deliver features and applications to their customers.

The local exchange carrier market has become ground-zero for much of the work being done in Internet telecom today. Particularly at the local level, competitive and incumbent carriers alike are realizing that just providing dial tone (for Internet or telephony) won't be enough to succeed long term. Carriers are starting to have to do the one thing they've always avoided: respond to the needs of individual business customers. As a result more intelligence will be pushed from the customer premises into the local loop and network EDGE.

The technology to make it all happen, of course, is being put in place today. Just look at the development of the gateways, softswitch, and the broadband voice-over-DSL.and DOCSIS cable deployments to see how the new generation of voice/data infrastructure is as much about facilitating service creation and delivery as it is about creating a more economical transport. Today, service providers are finding alternative and innovative methods for the delivery of voice services on top of DSL lines and coaxial cable lines.

At the same time as the carrier world is being shaken up by convergence, the enterprise communications market is undergoing changes of equal if not greater proportions. As e-business practices become a more integral aspect of enterprises across the board, reliance on communications will only increase. The latest trends in IP PBXs have already begun to reflect this concept. In general, hardware will be de-emphasized at the enterprise level, and displaced by distributed, network-based software platforms, managed remotely.

On the public network side, mobile wireless has become the most visible intersection of telephony and the Internet, a crossing that will be galvanized by third-generation technology. Within the enterprise, the convergence promise of "one-wire" infrastructure is quickly giving way to a "no-wire" model.

Nuntius technology enables the transmission of voice, fax and modem traffic over an IP or ATM backbone network. The software products accomplish this in three functional areas. These functions are designed to execute in a distributed fashion on programmable Digital Signal Processors (DSPs) and RISC and CISC Microprocessors. The three functional modules are Signal Processing Software, Telephony Processing Software and Protocol Processing Software.

Signal Processing Software
The Signal Processing software provides the interface between the analog world we live in and the digital binary world of embedded processors. This software prepares analog or PCM voice samples for transmission over the packet network. Its components perform tone detection and generation, echo cancellation, voice compression, voice activity detection, jitter removal, re-sampling and voice packetization. It also performs equalization, modulation and demodulation in support of fax relay applications.

Telephony Processing Software
The Telephony Processing software addresses the complexities of translating between traditional telephony signals and modern data networks. It interacts with telephony equipment, translating signaling into state changes used by the Protocol Processing software to set up connections.

Protocol Processing Software
The Protocol Processing software receives signaling information that has been interpreted by the Telephony Processing software and converts it from the telephony signaling protocols to the specific packet signaling protocol that is used to set up connections over the data network. It also adds appropriate protocol headers to both voice and signaling packets before transmission. Standard formats for IP, Frame Relay and ATM networks are common.

Network registration Mobile phone

2006-09-16T08:23:00.000+08:00

On any cellular telecommunications system the way in which registration and call set-up occur needs to be carefully managed. Not only does the cellular telecommunications network need to provide quick and efficient service for its rightful customers, but it also needs to be able to offer high levels of security for the user and the network.

There are many different cellular telecommunications systems in use around the globe. Older ones are being phased out, and newer cellular systems are being introduced. Accordingly there is no single way in which registration and call set up are managed. However there are some general principles that are used, and these are illustrated here.

Basic requirements
When the mobile phone is turned on it needs to be able to communicate with the cellular telecommunications network. However the phone does not have an allocated channel, time slot or chip code (dependent upon the type of access method used). It is therefore necessary for there to be some methods or allocated means within the cellular telecommunications network, whereby a newly switched on mobile can communicate with the network and set up the standard communication.

Even if a call is not to be made instantly, the network needs to be able to communicate with the mobile to know where it is. In this way the network can route any calls through the relevant base station as the network would be soon overloaded if the notification of an incoming call had to be sent via several base stations.

Registration
There are a variety of tasks that need to be undertaken when a phone is turned on. This can eb seen by the fact that it takes a few seconds from switching the phone on before it is ready for use. Part of this process is the software start-up for the phone, but the majority comes from the registration process with the cellular network. There are several aspects to the regristration. The first is to make contact with the base station, and next the mobile has to register to allow it to have access to and use the network.

In order to make contact with the base station the mobile uses a paging or control channel. The name of this channel, and the exact way in which it works will vary from one cellular standard to the next, but it is a channel that is used that the mobile can access to indicate its presence. The message sent is often called the "attach" message. Once this has been achieved it is necessary for the mobile to register with the cellular network, and to be accepted onto it.

Network elements
It is necessary to have a register or database of users allowed to register with a given network. With mobiles often being able to access the all the channels available in a country, methods of ensuring the mobile registers with the correct network, and to ensure the account is valid are required. Additionally it is required for billing purposes. To achieve this, an entity on the network often known as the Authentication Centre (AuC) is used. The network and the mobile communicate and numbers giving the identity of the subscriber. Here the user information is checked to provide authentication and encryption parameters that verify the user's identity and ensure the confidentiality of each call protecting users and network operators from fraud.

Once accepted onto the network two further registers are normally required. These are the Home Location Register (HLR) and the Visitors Location Register (VLR). These two registers are required to keep track of the mobile so that the network knows where it is at any time so that calls can be routed to the correct base station or general area of the network. These registers are used to store the last known location of the mobile. Thus at registration the register is updated and then periodically the mobile updates its position. Even when the mobile is in what is termed its idle mode it will periodically communicate with the network to update its position and status.

When the mobile is switched off it sends a detach message. This informs the network that it is switching off, and enables the network to update the last known position for the mobile.

Home and abroad
The two registers are required, one for mobiles for which the network is the home network, i.e. the one with whom the contract exists, and the other for visitors. If there was only one register then every time the mobile sent any message to the foreign network, this would need to be relayed back to the home network and this would require international signalling. The approach which is adopted is to send a message back to the HLR when the mobile first enters the new country saying that the mobile is in a different network and that any calls for that mobile should be forwarded to the foreign visited network.

Overview "Cellular network "

2006-09-15T08:51:00.000+08:00

The network forms the heart of any cellular telephone system. The cellular network fulfils many requirements. Not only does the network enable calls to be routed to and from the mobile phones as well as enabling calls to be maintained as the cell phone moves from one cell to another, but it also enables other essential operations such as access to the network, billing, security and much more. To fulfil all these requirements the cellular network comprises many elements, each having its own function to complete.
The most obvious part of the cellular network is the base station. The antennas and the associated equipment often located in a container below are seen dotted around the country, and especially at the side of highways and motorways. However there is more to the network behind this, as the system needs to have elements of central control and it also needs to link in with the PSTN landline system to enable calls to be made to and from the wire based phones, or between networks.

Different cellular standards often take slightly different approaches for the cellular network required. Despite the differences between the different cellular systems, the basic concepts are very similar. Additionally cellular systems such as GSM have a well defined structure, and this means that manufacturers products can be standardised.

Basic cellular network structure
An overall cellular network contains a number of different elements from the base transceiver station (BTS) itself with its antenna back through a base station controller (BSC), and a mobile switching centre (MSC) to the location registers (HLR and VLR) and the link to the public switched telephone network (PSTN).

Of the units within the cellular network, the BTS provides the direct communication with the mobile phones. There may be a small number of base stations then linked to a base station controller. This unit acts as a small centre to route calls to the required base station, and it also makes some decisions about which of the base station is best suited to a particular call. The links between the BTS and the BSC may use either land lines of even microwave links. Often the BTS antenna towers also support a small microwave dish antenna used for the link to the BSC. The BSC is often co-located with a BTS.

The BSC interfaces with the mobile switching centre. This makes more widespread choices about the routing of calls and interfaces to the land line based PSTN as well as the HLR and VLR.

Base transceiver station, BTS
The base transceiver station or system, BTS consists of a number of different elements. The first is the electronics section normally located in a container at the base of the antenna tower. This contains the electronics for communicating with the mobile handsets and includes radio frequency amplifiers, radio transceivers, radio frequency combiners, control, communication links to the BSC, and power supplies with back up.

The second part of the BTS is the antenna and the feeder to connect the antenna to the base transceiver station itself. These antennas are visible on top of masts and tall buildings enabling them to cover the required area. Finally there is the interface between the base station and its controller further up the network. This consists of control logic and software as well as the cable link to the controller.

BTSs are set up in a variety of places. In towns and cities the characteristic antennas are often seen on the top of buildings, whereas in the country separate masts are used. It is important that the location, height, and orientation are all correct to ensure the required coverage is achieved. If the antenna is too low or in a poor location, there will be insufficient coverage and there will be a coverage "hole". Conversely if the antenna is too high and directed incorrectly, then the signal will be heard well beyond the boundaries of the cell. This may result in interference with another cell using the same frequencies.

The antennas systems used with base stations often have two sets of receive antennas. These provide what is often termed diversity reception, enabling the best signal to be chosen to minimise the effects of multipath propagation. The receiver antennas are connected to low loss cable that routes the signals down to a multicoupler in the base station container. Here a multicoupler splits the signals out to feed the various receivers required for all the RF channels. Similarly the transmitted signal from the combiner is routed up to the transmitting antenna using low loss cable to ensure the optimum transmitted signal.

Mobile switching centre (MSC)
The MSC is the control centre for the cellular system, coordinating the actions of the BSCs, providing overall control, and acting as the switch and connection into the public telephone network. As such it has a variety of communication links into it which will include fibre optic links as well as some microwave links and some copper wire cables. These enable it to communicate with the BSCs, routing calls to them and controlling them as required. It also contains the Home and Visitor Location Registers, the databases detailing the last known locations of the mobiles. It also contains the facilities for the Authentication Centre, allowing mobiles onto the network. In addition to this it will also contain the facilities to generate the billing information for the individual accounts.

In view of the importance of the MSC, it contains many backup and duplicate circuits to ensure that it does not fail. Obviously backup power systems are an essential element of this to guard against the possibility of a major power failure, because if the MSC became inoperative then the whole network would collapse.

Mobile phone electronics

2006-09-14T11:37:00.000+08:00

The mobile phone or cell phone as it is often called is equally important to the network in the operation of the complete cellular telecommunications network. Despite the huge numbers that are made, they still cost a significant amount to manufacture, discounts being offered to users as incentives to use a particular network. Their cost is a reflection of the complexity of the mobile phone electronics. They comprise several different areas of electronics, from radio frequency (RF) to signal processing, and general processing.

The design of a cell phone is particularly challenging. They need to offer high levels of performance, while being able to fit into a very small space, and in addition tot his the electronics circuitry needs to consume very little power so that the life between charges can be maintained.

Mobile phone contents
Mobile phones contain a large amount of circuitry, each of which is carefully designed to optimise its performance. The cell phone comprises analogue electronics as well as digital circuits ranging from processors to display and keypad electronics. A mobile phone typically consists of a single board, but within this there are a number of distinct functional areas, but designed to integrate to become a complete mobile phone:

* Radio frequency - receiver and transmitter

* Digital signal processing

* Analogue / digital conversion

* Control processor

* SIM or USIM card

* Power control and battery

Radio frequency elements
The radio frequency section of the mobile phone is one of the crucial areas of the cell phone design. This area of the mobile phone contains all the transmitter and receiver circuits. Normally direct conversion techniques are generally used in the design for the mobile phone receiver.

The signal output from the receiver is applied to what is termed an IQ demodulator. Here the data in the form of "In-phase" and "Quadrature" components is applied to the IQ demodulator and the raw data extracted for further processing by the phone.

On the transmit side one of the key elements of the circuit design is to keep the battery consumption to a minimum. For GSM this is not too much of a problem. The modulation used is Gaussian Minimum Shift Keying. This form of signal does not incorporate amplitude variations and accordingly it does not need linear amplifiers. This is a distinct advantage because non linear RF amplifiers are more efficient than linear RF amplifiers.

Unfortunately EDGE uses eight point phase shift keying (8PSK) and this requires a linear RF amplifier. As linear amplifiers consume considerably more current this is a distinct disadvantage. To overcome this problem the design for the mobile phone is organised so that phase information is added to the signal at an early stage of the transmitter chain, and the amplitude information is added at the final amplifier.

Analogue to Digital Conversion
Another crucial area of any mobile phone design is the circuitry that converts the signals between analogue and digital formats that are used in different areas. The radio frequency sections of the design use analogue techniques, whereas the processing is all digital.

The digital / analogue conversion circuitry enables the voice to be converted either from analogue or to digital a digital format for the send path, but also between digital and analogue for the receive path. It also provides functions such as providing analogue voltages to steer the VCO in the synthesizer as well as monitoring of the battery voltage, especially during charging. It also provides the conversion for the audio signals to and from the microphone and earpiece so that they can interface with the digital signal processing functions.

Another function that may sometimes be included in this area of the mobile phone design or within the DSP is that of the voice codecs. As the voice data needs to be compressed to enable it to be contained within the maximum allowable data rate, the signal needs to be digitally compressed. This is undertaken using what is termed a codec.

There are a number of codec schemes that can be used, all of which are generally supported by the base stations. The first one to be used in GSM was known as LPC-RPE (Linear Prediction Coding - Regular Pulse Excitation). However another scheme known as AMR (Adaptive Multi-Rate) is now widely used as it enables the data rate to be further reduced when conditions permit without impairing the speech quality too much. By reducing the speech data rate, further capacity is freed up on the network.

Digital Signal Processing
The DSP components of the mobile phone design undertake all the signal processing. Processes such as the radio frequency filtering and signal conditioning at the lower frequencies are undertaken by this circuitry. In addition to this, equalisation and correction for multipath effects is undertaken in this area of the design.

Although these processors are traditionally current hungry, the current processors enable the signal processing to be undertaken in a far more power effective manner than if analogue circuits are used.

Control processor
The control processor is at the heart of the design of the phone. It controls all the processes occurring in the phone from the MMI (Man machine interface) which monitors the keypad presses and arranging for the information to be displayed on the screen. It also looks after all the other elements of the MMI including all the menus that can be found on the phone.

Another function of the control processor is to manage the interface with the mobile network base station. The software required for this is known as the protocol stack and it enables the phone to register, make and receive calls, terminate them and also handle the handovers that are needed when the phone moves from one cell to the next. Additionally the software formats the data to be transmitted into the correct format with error correction codes included. Accordingly the load on this processor can be quite high, especially when there are interactions with the network.

The protocols used to interact with the network are becoming increasingly complicated with the progression from 2G to 3G. Along with the increasing number of handset applications the load on the processor is increasing. To combat this, the design for this area of the phone circuitry often uses ARM processors. This enables high levels of processing to be achieved for relatively low levels of current drain.

A further application handled by this area of the design of the mobile phone is the monitoring the state pf the battery and control of the charging. In view of the sophisticated monitoring and control required to ensure that the battery is properly charged and the user can be informed about the level of charge left, this is an important area of the design.

Battery
Battery design and technology has moved on considerably in the last few years. This has enabled mobile phones to operate for much longer. Initially nickel cadmium cells were used, but these migrated to nickel-metal-hydride cells and then to lithium ion cells. With phones becoming smaller and requiring to operate for longer from a single charge, the capacity of the battery is very important, and all the time the performance of these cells is being improved.

Duplex transmission for cellular telecommunications systems

2006-09-13T14:35:00.000+08:00

Basic radio communications systems use a single channel and what is known as a press to talk system, where the user presses a button or "pressel" on the microphone to talk, and then releases the pressel to listen on the same frequency. This system is known as simplex as it uses a single channel. For a phone system a full duplex system is required where it is possible to speak in both directions at the same time. There are two main ways in which this can be achieved. The first is to transmit in one direction on one frequency, and simultaneously transmit in the other direction on another. To achieve this there must be sufficient frequency separation and filtering to ensure that the transmitter does not interfere with the receiver. A scheme that uses one frequency for transmitting traffic in one direction and another frequency for traffic in the other is known as Frequency Division Duplex (FDD).

The other system uses only a single frequency and can be employed where digital or data systems are used. This requires the analogue audio signal to be digitised. A single frequency is used for the radio frequency signal and short packets of data are sent first in one direction, and then the other. As these data bursts are relatively short the user does not notice the short delay introduced by the fact that the digitised speech signal is not sent immediately. This type of system is known as Time Division Duplex (TDD).

It is often necessary to distinguish between the link from the mobile to the base station, and the link from the base station to the mobile. The first, i.e. the link from the mobile to the base station is often called the uplink or the reverse link as the signal is being transmitted up to the base station. The second, i.e. the link from the base station to the mobile is known as the downlink or the forward link.

CELLULAR MULTIPLE ACCESS Schemes by moblie phone

2006-09-12T11:53:00.000+08:00

In any cellular telecommunications or mobile phone system, it is necessary to have a scheme that enables several multiple users to gain access to it and use it simultaneously. These methods are known as multiple access schemes.

There are three main multiple access schemes that are in use at the moment:

* Frequency Division Multiple Access - FDMA

* Time Division Multiple Access - TDMA

* Code Division Multiple Access - CDMA

FDMA
FDMA is the most straightforward of the multiple access schemes that have been used. As a subscriber comes onto the system, or swaps from one cell to the next, the network allocates a channel or frequency to each one. In this way the different subscribers are allocated a different slot and access to the network. As different frequencies are used, the system is naturally termed Frequency Division Multiple Access. This scheme was used by all analogue systems.

TDMA
The second system came about with the transition to digital schemes. Here digital data could be split up in time and sent as bursts when required. As speech was digitised it could be sent in short data bursts, any small delay caused by sending the data in bursts would be short and not noticed. In this way it became possible to organise the system so that a given number of slots were available on a give transmission. Each subscriber would then be allocated a different time slot in which they could transmit or receive data. As different time slots are used for each subscriber to gain access to the system, it is known as time division multiple access. Obviously this only allows a certain number of users access to the system. Beyond this another channel may be used, so systems that use TDMA may also have elements of FDMA operation as well.

CDMA
The final scheme uses one of the aspects associated with the use of direct sequence spread spectrum. It can be seen from the article in the cellular telecoms area of this site that when extracting the required data from a DSSS signal it was necessary to have the correct spreading or chip code, and all other data from sources using different orthogonal chip codes would be rejected. It is therefore possible to allocate different users different codes, and use this as the means by which different users are given access to the system.

The scheme has been likened to being in a room filled with people all speaking different languages. Even though the noise level is very high, it is still possible to understand someone speaking in your own language. With CDMA different spreading or chip codes are used. When generating a direct sequence spread spectrum, the data to be transmitted is multiplied with spreading or chip code. This widens the spectrum of the signal, but it can only be decided in the receiver if it is again multiplied with the same spreading code. All signals that use different spreading codes are not seen, and are discarded in the process. Thus in the presence of a variety of signals it is possible to receive only the required one.

In this way the base station allocates different codes to different users and when it receives the signal it will use one code to receive the signal from one mobile, and another spreading code to receive the signal from a second mobile. In this way the same frequency channel can be used to serve a number of different mobiles.

Situation today
Although the latest cellular telecommunications systems use CDMA as their basis, elements of TDMA and FDMA are also used. Both the major schemes, UMTS and CDMA2000 have a limit on the number of users who are able to use a single channel. In some instances two or more channels may be allocated to a particular cell. This means that the system still uses an element of FDMA.

Additionally UMTS incorporates some timeslots, and this means that the scheme uses elements of TDMA.
While CDMA is currently the dominant technology, both the other forms of access scheme are still in evidence, not just in legacy technologies, but utilised as part of the main access scheme in the latest 3G systems.

Digital Wireless Communications Technology

2006-09-11T12:49:00.000+08:00

The primitive state of mobile data transmission will be transformed over the next three years as operators construct third-generation networks. For all the hype about the wireless Internet, today's mobile data networks are rather primitive. Usually bolted onto cell phone systems built for voice, they offer low bit rates and poor interoperability. All this will change over the next three years, as operators construct 2.5 and 3G mobile networks. The aim is to provide packet-switched data to a handheld terminal with throughput measured in hundreds of Kbps.

3G has been in gestation since 1992, when the International Telecommunication Union (ITU) began work on a standard called IMT-2000 with 16 different proposals. The ITU envisaged IMT-2000 as a single global standard, but the world's regulators, vendors, and carriers were unable to reach a unanimous agreement. By 1998, the ITU was faced with 13 different 3G-radio interfaces based on CDMA. Qualcomm and the Interim Standard 95 (IS-95) CDMA industry wanted one harmonized CDMA standard based on their chosen technology, cdma2000, because it would be backward compatible. Ericsson and the GSM camp wanted their own technology, WCDMA, which is incompatible with today's IS-95 systems. In October 1999, representatives from different countries finally agreed to disagree. The result is a "federal standard," or, more accurately, fudge. IMT-2000 will have at least three optional modes of operation: W-CDMA, cdma2000 and time division duplex, an optional component of W-CDMA. So the two technologies, WCDMA and cdma2000, remains pitted against each other. WCDMA is gaining most of the industry's attention, threatening to sweep the world's footprint with its emphasis on global roaming and promises of economies of scale. Timing and cost of the technology remain as drawbacks. Below, in the table, is a summary of the family of five sets of 3G IMT-2000 wireless standards approved and published by ITU on May 2000.

The path to 3G will be gradual, and expensive to deploy, everyone wants to ensure compatibility with their existing systems.

Three IMT-2000 modes are based on Code Division Multiple Access, a system that enables many users to share the same frequency band at the same time. CDMA codes are chosen so that they cancel each other out. For exact cancellation, signals must be perfectly timed; base stations need to make very precise measurements of their time and location. They do this by using signals from Global Positioning System (GPS) satellites, which can pinpoint anywhere on Earth to within four meters and measure time more accurately than the Earth's own rotation. The only CDMA system in use so far is cdmaOne, developed by Qualcomm but now supervised by an independent organization called the CDMA Development Group (CDG). It has been standardized by the Telecommunica-tions Industry Association (TIA) as IS-95A, and is popular among cellular operators in America and Asia. Because it already uses CDMA, it is easier to upgrade to 3G compared to rival systems based on Time Division Multiple Access (TDMA).

cdmaOne spreads every signal over a 1.25MHz channel, transmitting on the entire bandwidth at once. It uses a set of 64 codes, known as Walsh sequences, so in theory up to 64 phones could use the channel at once. In practice, that number depends on the data throughput. The basic system offers voice and 14.4Kbps data rates, which facilitates between 15 and 20 users. An upgrade called IS-95b offers data rates of up to 115Kbits/sec, which would mean only two users per channel.

To reach the IMT-2000 target of 2Mbits/sec, CDMA systems need to use more codes, a different modulation scheme, and wider bandwidths. The official upgrade, developed by Qualcomm and ratified by the ITU, is known as cdma2000 3X. The 3 in 3X comes from its 3.75MHz bandwidth, the result of three cdmaOne 1.25MHz channels joined together. As an intermediary step, some cdmaOne operators are deploying a technology called cdma2000 1X, which uses the same 1.25MHz channels and doubles the number of codes to 128, thus doubling either the throughput per user or the number of users in a cell.

Qualcomm and Motorola are also pushing rival schemes that enhance 1XRT, known respectively as High Data Rate (HDR) and 1Xtreme. Both 1Xtreme and HDR work by altering the modulation scheme, or the way data is actually represented in radio waves. Most cell phones use a system called Phase Shift Keying (PSK), which interrupts a wave and moves it to a different point in its cycle. The bit rate depends on the frequency of these interruptions, known as symbols, and on the number of shapes that each symbol can take. The symbols in quadrature PSK, the system used by cdmaOne, can take four different shapes. This means that each shape can represent two bits, since two bits can take four combinations. The 8-PSK variation could represent three bits per symbol, increasing the data rate by half. HDR and 1Xtreme automatically increase the number of shapes to the highest number supported, depending on their connection quality. By late this year, most CDMA operators plan to migrate to 1X, which will give them data enhancements of up to 144 kbps with primarily a software upgrade. Shortly after, in early 2002, CDMA operators are likely to add what is known as 1X EV (evolution), which will allow them to dedicate a 1.25 Mhz channel to data services that will offer speeds up to 2.4 Mbps.

Europe, South Korea, Japan and are rolling out Wideband CDMA (WCDMA), or Universal Mobile Telecommunications System (UMTS). This requires the new spectrum assigned by the ITU, and thus won't be used in the United States without establishing new spectrum. It is technically very similar to cdma2000 3XRT but uses a slightly wider bandwidth, hence the name. The wider bands are necessary so that the system can interoperate with Global System for Mobile Communications (GSM), the most ubiquitous second-generation (2G) wireless standard. Japan's NTT DoCoMo is expected to be the world's fiirst operator to launch a WCDMA system in fall of 2001. However, DoCoMo's WCDMA technology is slightly different from the standards accepted by the ITU. The standard, dubbed Japanese WCDMA, or JW-CDMA, has certain subtleties, including proprietary protocol stacks.

In the US, TDMA operator AT&T wireless announced plans to deploy GSM services and 2.5 G general packet radio service (GPRS, and then onto WCDMA. Its alliances and affiliates in Canada: include Rogers Cantel and TelCorp PCS will follow. They are not waiting on enhanced data rates for GSM evolution (EDGE). AT& T Wireless says it still plans to migrate to EDGE, but industry dynamics are suggesting that EDGE might fail --- eroding the economies of scales that TDMA operators have sought. EDGE technology was originally envisioned as a spectrally efficient option for European operators that were not successful at winning new spectrum. All but one major European operator has obtained new spectrum licenses. Other US operators, such as Cingular, testing GPRS, are likely to follow the AT&T Wireless movement. Carriers in Europe have not announced intentions to deploy EDGE technology, but are heavily committed to GSM with GPRS. The CDMA community is just beginning to embrace the idea of subscriber identity modules that store subscriber information and are commonly used in GSM phones today.

Because the WCDMA standard is a moving target, most vendors are working on deploying Release 99 of the standard and adding in corrections, which standards bodies will complete in June of 2001. Release 2000 will be broken into two parts, Release 4 and Release 5. WCDMA Release 99 has two branches. One controls voice traffic, while the other supports data traffic. This means the core network is circuit switched with packet data running on top. Together, Release 4 and Release 5 will create an all IP network.

The sidebar icons summarize the different embedded software technologies essential to the operation of Wireless Wide Area Network (WWAN) wireless infrastructure.

"BASIC CELLULAR" Concepts

2006-09-11T08:41:00.000+08:00

Cellular telecommunications systems are widely used today and need to offer very efficient use of the available frequency spectrum. With billions of mobile phones in use around the globe today, it is necessary to re-use the available frequencies many times over without mutual interference of one cell phone to another.

Early schemes for radio telephones schemes used a single central transmitter to cover a wide area. These radio telephone systems suffered from the limited number of channels that were available. Often the waiting lists for connection were many times greater than the number of people that were actually connected.

The need for a spectrum efficient system
To illustrate the need for efficient spectrum usage for a radio telecommunications system, take the example where each user is allocated a channel. While more effective systems are now in use, the example will take the case of an analogue system. Each channel needs to have a bandwidth of around 25 kHz to enable sufficient audio quality to be carried as well as enabling there to be a guard band between adjacent signals to ensure there are no undue levels of interference. Using this concept it is only possible to accommodate 40 users in a frequency band 1 MHz wide. Even of 100 MHz were allocated to the system this would only enable 4000 users to have access to the system. Today cellular systems have millions of subscribers and therefore a far more efficient method of using the available spectrum is needed.

Cell system for frequency re-use
The method that is employed is to enable the frequencies to be re-used. Any transmitter will only have a certain coverage area. Beyond this the signal level will fall to a limited below which it cannot be used and will not cause significant interference to users associated with a different transmitter. This means that it is possible to re-use a channel once outside the range of the transmitter. The same is also true in the reverse direction for the receiver, where it will only be able to receive signals over a given range. In this way it is possible to arrange split up an area into several smaller regions, each covered by a different transmitter / receiver station.

These regions are conveniently known as cells, and give rise to the name of a cellular telecommunications system. Diagrammatically these cells are often shown as hexagonal shapes that conveniently fit together. In reality this is not the case. They have an irregular boundary because of the terrain over which they travel. Hills, buildings and other objects all cause the signal to be attenuated and diminish differently in each direction.

It is also very difficult to define the exact edge of a cell. The signal strength gradually reduces and towards the edge of the cell performance will fall. As the mobiles themselves will have different levels of sensitivity, this adds a further greying of the edge of the cell. Therefore it is never possible to have a sharp cut-off between cells. In some areas they may overlap, whereas in others there will be a "hole" in coverage.

Cell clusters
To overcome this problem, in a basic cellular system, adjacent cells are allocated different frequency bands so that they can overlap without causing interference. In this way cells can be grouped together in what is termed a cluster.

Often these clusters contain seven cells, but other configurations are also possible. Seven is a convenient number, but there are a number of conflicting requirements that need to be balanced when choosing the number of cells in a cluster:

* Limiting interference levels

* Number of channels that can be allocated to each cell site

It is necessary to limit the interference between cells having the same frequency. The topology of the cell configuration has a large impact on this. The larger the number of cells in the cluster, the greater the distance between cells sharing the same frequencies.

In the ideal world it might be good to choose a large number of cells to be in each cluster. Unfortunately there is only a limited number of channels available. This means that the larger the number of cells in a cluster, the smaller the number available to each cell, and this reduces the capacity.

This means that there is a balance that needs to be made between the number of cells in a cluster, and the interference levels, and the capacity that is required.

Cell size
Even though the number of cells in a cluster can help govern the number of users that can be accommodated, by making all the cells smaller it is possible to increase the overall capacity of the network. However a greater number of transmitter receiver or base stations are required if cells are made smaller and this increases the cost to the operator. Accordingly in areas where there are more users, small low power base stations are installed.

The different types of cells are given different names according to their size and function:

* Macro cells

* Micro cells

* Pico cells

* Selective cells

* Umbrella cells

Macro cells are large cells that are usually used for remote or sparsely populated areas. These may be 10 km or possibly more in diameter. Micro cells are those that are normally found in densely populated areas which may have a diameter of around 1 km. Picocells may also be used for covering very small areas such as particular areas of buildings, or possibly tunnels where coverage from a larger cell is not possible. Obviously for the small cells, the power levels used by the base stations are much lower and the antennas are not position to cover wide areas. In this way the coverage is minimised and the interference to adjacent cells is reduced.

Other types of cell may be used for some specialist applications. Sometimes cells termed selective cells may be used where full 360 degree coverage is not required. They may be used to fill in a hole in the coverage, or to address a problem such as the entrance to a tunnel etc. Another type of cells known as an umbrella cell is sometimes used in instances such as those where a heavily used road crosses an area where there are microcells. Under normal circumstances this would result in a large number of handovers as people driving along the road would quickly cross the microcells. An umbrella cell would take in the coverage of the microcells (but use different channels to those allocated to the microcells). However it would enable those people moving along the road to be handled by the umbrella cell and experience fewer handovers than if they had to pass from one microcell to the next.

Cell Phone Systems

2006-09-10T10:20:00.000+08:00

The development and history of the mobile phone has seen a tremendous number of changes since the first cell phones were introduced. It was only at the beginning of the 1980s when mobile phone technology started to be deployed commercially. Since then there have been many new cell phone or mobile phone systems introduced, and many improvements have been made in the technology. The mobile phones themselves as well as the associated equipment including base stations and the other network equipment has become much cheaper and far smaller.

One of the major changes is the level of market penetration that has been achieved. Around one in six of the world's population now has a mobile phone. When the first systems were introduced the operating and ownership costs were such that they were only used by businesses with a real need for their employees to be able to keep in touch all the time. Now they are an almost essential personal belonging for most people. In many countries market penetration has exceeded 70%, and it is not uncommon for many people to have one phone for business and another for personal use. Accordingly in some segments of the population the market penetration is very much higher than 70%.

Development overview
The phones themselves have undergone many changes during their history. The technologies that have been used have improved dramatically. The first systems to be launched were based on analogue technology. The early phones were very large and could certainly not be placed in a pocket like the phones of today.

The first generation (1G) phone systems as they are now known were overtaken in the early 1990s by the first digital systems.

The high levels of use and limited frequency allocation meant that greater spectrum use efficiency was needed. Accordingly the next or second-generation (2G) phone systems were introduced to meet this need.

As the usage of phones increased and people became more mobile, new possibilities emerged for using the phones for data transfer. They could be used to download information from the Internet, or to send video. The first stage in this migration was to provide a medium speed data transfer capability. These systems were accordingly known as 2.5G.

However the ultimate aim was to provide a relatively high-speed data transfer capability. These full third generation (3G) systems took longer to develop and roll-out than had been originally anticipated as a result of higher development costs and a downturn in the global economy. However they are able to provide a significant improvement in capability over the 2.5 G systems

Analogue Systems
There was an enormous variety of first generation systems that were introduced. Much of the early development of cellular systems had been undertaken in the USA, but the first fully commercial system to be launched was the Nordic Mobile Telephone (NMT) system. Shortly after this a system known as the Advanced Mobile Phone System (AMPS) was launched commercially. This was developed primarily by Bell and was introduced in the USA although many other countries used this system later. A further system known as Total Access Communication System (TACS) developed by Motorola was introduced in the UK and many other countries.

These were the main systems that were developed, although around the globe many variants were developed to suit the needs of the individual countries.

Although there were differences in the specifications of the systems, they were all very similar in concept. The voice information was carried on a frequency-modulated carrier. A control channel was also used to enable the mobile to be routed to a suitable vacant channel. The channel spacing for each system was different. NMT used a 12.5 kHz channel spacing, AMPS, a 30 kHz spacing and TACS a 25 kHz spacing. A later development of AMPS called NAMPS or narrowband AMPS used a 10 kHz channel spacing to conserve spectrum.

Digital systems
The analogue systems were very successful, but their very success started to show some of their shortcomings. The main one was the inefficient way in which they sued the spectrum. With the growth rates that were being seen, there was insufficient spectrum to support the quality of service that was required. By converting to a digital system, considerable savings could be made. A number of systems arose from this initiative. These second-generation systems as they were termed, started to be deployed in the early 1990s and their history is just as remarkable.

The system that was developed in Europe was the result of 26 telecommunications companies working together. Work actually started in 1982, and the roll-out commenced in 1991. The system known by the letters GSM was originally called Groupe Speciale Mobile but this was later changed to Global System for Mobile communications in view of the wide involvement in its development. It used time division multiple access (TDMA) to allow up to eight users to use each of the channels that are spaced 200 kHz apart. The basic system used frequencies in the 900 MHz band, but other bands in the 1800 and 1900 MHz (USA) bands were added. New bands in the 850 MHz region were also added.

In the USA a system specially designed to operate alongside their AMPS system was devised. The system was known under a variety of names including Digital AMPS or DAMPS, and US Digital Cellular (USDC), although it is normally known just as TDMA today as it relies on TDMA technology. The system was originally defined under standard number IS-54, although this was later updated to IS-136 and it uses a 30 kHz channel spacing to make it compatible with the existing AMPS systems in operation.

Another development in the USA from Qualcomm took a major leap in technology. It introduced a totally new concept for multiple access. Based on direct sequence spread spectrum (DSSS) that had previously been used for military transmissions, it used a multiple access system known as code division multiple access (CDMA). The new system offered far greater levels of spectrum efficiency although it required more complicated circuitry in the handsets. The system was defined under standard IS-95 and each carrier had a bandwidth of 1.25 MHz, although many users could use the same channel. The specification was updated from IS-95A to IS-95B. It was this later standard that went under the trade name cdmaOne.

2.5G
Once the second-generation systems became established it soon became apparent that the limited data capabilities of some of the 2G systems were a significant disadvantage. Many applications for data transfer with the increased use of the Internet and laptop computers were seen. Even though the third generation systems were on the horizon, developments were needed to provide a service before they entered the market. One of the first was the General Packet Radio Service (GPRS) development for the GSM system. Its approach centred on the use of packet data. Up until this time all circuits had been dedicated to a given user in an approach known as circuit switched, i.e. where a complete circuit is switched for a given user. This was inefficient when a channel was only carrying data for a small percentage of the time. The new packet switched approach routed individual packets of data from the transmitter to the receiver allowing the same circuit to be used by different users. This enabled circuits to be used more efficiently and charges to be metered according to the data transferred.

Further data rate improvements were made using a system known as EDGE (Enhanced data Rates for GSM Evolution). This basically took the GPRS system and added a new modulation scheme, 8PSK, to enable a much higher data rate to be achieved. Whilst the symbol rate remained the same at 270.833 samples per second, each symbol carried three bits instead of one.

Whilst GPRS and EDGE were applied to GSM networks, enhancements were also applied to the CDMA system that originated in the USA. Here an evolutionary path from 2G through 2.5G to 3G was created. The intermediate stage was development of cdmaOne was CDMA2000 1X. This scheme retained the 1.25 MHz bandwidth of IS95 / cdmaOne, but by adding further channels enabled data transfer rates of 307 kbps to be achieved, thereby doubling the capacity of IS95B.

Third Generation
Although technologies such as GPRS, EDGE and CDMA2000 1X were able to deliver significantly higher data rates than their predecessors, the final migration was to the full 3G service. There were three main technologies.

From Europe there was the UMTS (Universal Mobile Telecommunications System) using wideband CDMA (W-CDMA). This system used a 5 MHz channel spacing and provided data rates of up to 2 Mbps.

Then there were the CDMA2000 evolutions. The first to be launched was CDMA2000 1xEV-DO. Here the letters EV-DO stood for Evolution Data Only. The idea for this system was that many of the applications would only need a data connection, as in the case of a data card for use in a PC to provide a wireless Internet capability over a mobile phone system. For any applications needing both data and voice a standard 1X channel would be required in addition. Although using CDMA technology, the EV-DO system also used TDMA technology as well to provide the throughput whilst still maintaining backward compatibility with IS95 (cdmaOne) and CDMA2000 1X.

The next evolution of the CDMA2000 family was CDMA2000 1xEV-DV. This was an evolution of the 1X system, and totally distinct from 1xEV-DO and it provided a full data and voice capability. Again this system was able to provide backward compatibility with IS95 (cdmaOne) and CDMA2000 1X whilst still being able to provide a data capability of 3.1 Mbps in the forward direction.

These major two players in the 3G scene both used what is called frequency division duplex (FDD) where the forward and reverse links used different frequencies. Within UMTS there is a specification covering a time division duplex (TDD) system where the forward and reverse links used the same frequency but use different timeslots. However the TDD version is not being deployed for some time.
A third 3G system that originated in China uses TDD. Known as time division synchronous CDMA (TD-SCDMA) this system used a 1.6 MHz channel spacing and was thought to be likely to take a significant portion of the Chinese market along with those in neighbouring countries

Summary
It took just over 20 years to migrate from the first analogue systems to the 3G systems capable of high data rate transfers. Now people are working on the 4G standards and it remains to be seen what new services and capabilities this new technology will offer.