Convergence Technologies for 3G Networks: IP, UMTS, EGPRS and ATM / Edition 1

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UMTS (Universal Mobile Telecommunications Service) is a so-called "third-generation (3G)," broadband, packet-based transmission of text, digitized voice, video, and multimedia at data rates up to 2 megabits per second (Mbps) that will offer a consistent set of services to mobile computer and phone users no matter where they are located in the world. More and more frequently traditional data communications technologies such as IP (Internet Protocol) and ATM (Asynchronous Transfer Mode) are deployed in wireless network infrastructures (UMTS). In this context numerous new protocols have been standardised recently, but little information on these changes or the application of the technology to the telecommunications industry has been published. All ATM and most IP books cover very general deployment of the technologies in wireline networks. The same applies for books on UMTS where core network issues are often not addressed in much detail. There is little information of either the application of IP and ATM to the telecomms industry or the impact of this on circuit-switched networks. This is primary reading for professional engineers in the wireless telecoms and data comms industries (equipment/network design and development). It is also ideal for professionals from an IP/ATM networking background who now need to become familiar with data applications in wireless communications networks. It is also useful for people from a telecoms background, not so acquainted with the world of data communications.

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Editorial Reviews

From the Publisher
"…this is an excellent volume, a must-have for systems architects…also to be commended for its cohesive and comprehensive assembly of many complex standards." (Computing, October 4, 2005)

"This both is very detailed, yet readable. It would be an excellent read for both students and telecommunications professionals…" (Computing, June 8, 2005)

"…well-structured…it provides detailed, and carefully selected and prepared, material." (Computing, October 21, 2004)

"...very detailed yet excellent read for both students and professionals..." (The IEE Communications Engineer, June/July 2004)

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Product Details

  • ISBN-13: 9780470860915
  • Publisher: Wiley
  • Publication date: 3/1/2004
  • Edition number: 1
  • Pages: 670
  • Product dimensions: 7.11 (w) x 9.84 (h) x 1.72 (d)

Meet the Author

Dr. Jeffrey Bannister is a co-founder and Telecommunications Specialist at Orbitage. A native of Ireland, he received his Ph.D. in Telecommunications/High-Speed electronics from Trinity College in Dublin. He has over 15 years of experience, and holds an internationally recognized teaching qualifications. Jeffrey has also been a lecturer, research fellow and course developer with the Dublin Institute of Technology, Temasek Polytechnic, Singapore, and Trinity College in Dublin, as well as providing consultation to a number of companies in Europe and Asia. He has been living in Malaysia for the past 5 years.

Mr. Paul Mather is a co-founder of Orbitage and has been located in the ASEAN region for the last seven year, during which time he has been involved in course development, training and consultancy for a number of companies. Prior to his relocation from Blackpool, UK, he worked for a British college, where he was engaged as both a lecturer in Information Engineering and as the computer network manager. As a certified internal verifier of vocational qualifications, he has comprehensive experience in delivery, assessment and development of a variety of IT and Communication programs. He is credited with establishing the first Novel Educational Academic Partnership in the ASEAN region. In an industrial context, he has worked in the IT and Communications fields for over 18 years, this work has taken him to many countries as well as various oil and gas platforms in the North Sea.

Mr. Sebastian Coope is an IP/Software Specialist at Orbitage. From a small village called Bollington near Manchester originally, he received his Masters in Data Communications and Networking from Leeds Metropolitan University. He has worked in a wide range of roles as software engineer development and project manager, as well as consultant in the fields of network security and management. He has also worked as lecturer and consultant at both Temasek Polytechnic Singapore and the University of Staffordshire. At Orbitage he has led the team responsible for the development of mobile application products. He is also co-author of Computer Systems (Coope, Cowley and Willis), a university text on computer architecture.

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Read an Excerpt

Convergence Technologies for 3G Networks

By Jeffrey Bannister Paul Mather Sebastian Coope

John Wiley & Sons

ISBN: 0-470-86091-X

Chapter One

Principles of Communications


Many practical communication systems use a network which allows for full connectivity between devices without requiring a permanent physical link to exist between two devices. The dominant technology for voice communications is circuit switching. As the name implies, it creates a series of links between network nodes, with a channel on each physical link being allocated to the specific connection. In this manner a dedicated link is established between the two devices.

Circuit switching is generally considered inefficient since a channel is dedicated to the link even if no data is being transmitted. If the example of voice communications is considered, this does not come close to 100% channel efficiency. In fact, research has shown that it is somewhat less that 40%. For data which is particularly bursty this system is even more inefficient. Generally before a connection is established, there is a delay; however, once connected, the link is transparent to the user, allowing for seamless transmission at a fixed data rate. In essence, it appears like a direct connection between the two stations. Some permanent type circuits such as leased lines do not have a connection delay since the link is configured when it isinitially set up. Circuit switching is used principally in the public switched telephone network (PSTN), and private networks such as a PBX or a private wide area network (WAN). Its fundamental driving force has been to handle voice traffic, i.e. minimize delay, but more significantly permit no variation in delay. The PSTN is not well suited to data transmission due to its inefficiencies; however, the disadvantages are somewhat overcome due to link transparency and worldwide availability.

The concept of packet switching evolved in the early 1970s to overcome the limitations of the circuit-switched telecommunications network by implementing a system better suited to handling digital traffic. The data to be transferred is split into small packets, which have an upper size limit that is dependent on the particular type of network. For example, with asynchronous transfer mode (ATM) the cell size is fixed at 53 bytes whereas an Ethernet network carries frames that can vary in size from 64 bytes up to 1500 bytes. A packet contains a section of the data plus some additional control information referred to as a header. This data, which has been segmented at the transmitter into packet sizes that the network can handle, will be rebuilt into the original data at the receiver. The additional header information is similar in concept to the address on an envelope and provides information on how to route the packet, and possibly where the correct final destination is. It may also include some error checking to ensure that the data has not been corrupted on the way. On a more complex network consisting of internetworking devices, packets that arrive at a network node are briefly stored before being passed on, once the next leg of the journey is available, until they arrive at their destination. This mechanism actually consists of two processes, which are referred to as buffering and forwarding. It allows for much greater line efficiency since a link between nodes can be shared by many packets from different users. It also allows for variable rates of transmission since each node retransmits the information at the correct rate for that link. In addition, priorities can be introduced where packets with a higher priority are transmitted first. The packet-switched system is analogous to the postal system. There are two general approaches for transmission of packets on the network: datagrams and virtual circuits.

2.1.1 Datagram approach

With the datagram approach, each packet is treated independently, i.e. once on the network, a packet has no relation to any others. A network node makes a routing decision and picks the best path on which to send the packet, so different packets for the same destination do not necessarily follow the same route and may therefore arrive out of sequence, as illustrated in Figure 2.1. The headers in the figure for each of the packets will have some common information, such as the address of the receiver, and some information which is different, such as a sequence number. Reasons for packets arriving out of sequence may be that a route has become congested or has failed. Because packets can arrive out of order the destination node needs to reorder the packets before reassembly. Another possibility with datagrams is that a packet may be lost if there is a problem at a node; depending on the mechanism used the packet may be resent or just discarded. The Internet is an example of a datagram network; however, when a user dials in to an ISP via the PSTN (or ISDN), that link will be a serial link, most probably using the PPP protocol (see Chapter 5). This access link is a circuit-switched connection in that the bandwidth is dedicated to the user.

2.1.2 Virtual circuits

Since the packets are treated independently across the network, datagram networks tend to have a high amount of overhead because the packet needs to carry the full address of the final destination. This overhead on an IP network, for example, will be a minimum of 20 bytes. This may not be of significance when transferring large data files of 1500 bytes or so but if voice over IP (VoIP) is transferred the data may be 32 bytes or less and here it is apparent that the overhead is significant. This approach establishes a virtual circuit through the nodes prior to sending packets and the same route is used for each packet. The system may not guarantee delivery but if packets are delivered they will be in the correct order. The information on the established virtual circuit is contained in the header of each packet, and the nodes are not required to make any routing decisions but forward the packets according to the information when the virtual circuit was established. This scheme differs from a circuit-switched system as packets are still queued and retransmitted at each node and they do have a header which includes addressing information to identify the next leg of the journey. The header here may be much reduced since only localized addressing is required, such as 'send me out on virtual circuit 5' rather than a 4-byte address for the IP datagram system. There are two types of virtual circuit, permanent and switched:

A permanent virtual circuit is comparable to a leased line and is set up once and then may last for years.

A switched virtual circuit is set up as and when required in a similar fashion to a telephone call. This type of circuit introduces a setup phase each and every time prior to data transfer.

Figure 2.2 shows a network containing a virtual circuit. Packets traverse the virtual circuit in order and a single physical link, e.g. an STM-1 line, can have a number of virtual circuits associated with it.

The term connectionless data transfer is used on a packet-switched network to describe communication where each packet header has sufficient information for it to reach its destination independently, such as a destination address. On the other hand, the term connection-oriented is used to denote that there is a logical connection established between two communicating hosts. These terms, connection-oriented and connectionless, are often incorrectly used as meaning the same as virtual circuit and datagram. Connection-oriented and connectionless are services offered by a network, whereas virtual circuits and datagrams are part of the underlying structure, thus a connection-oriented service may be offered on a datagram network, for example, TCP over IP.


In an analogue phone system, the original voice signal is directly transmitted on the physical medium. Any interference to this signal results in distortion of the original signal, which is particularly difficult to remove since it is awkward to distinguish between the signal and noise as the signal can be any value within the prescribed range. When the signals travel long distances and have to be amplified the amplifiers introduce yet further noise. Also, it is extremely easy to intercept and listen in to the transmitted signal. With digital transmission, the original analogue signal is now represented by a binary signal. Since the value of this signal can only be a 0 or a 1, it is much less susceptible to noise interference and when the signal travels long distances repeaters can be used to regenerate and thus clean the signal. A noise margin can be set in the centre of the signal, and any value above this is considered to be of value 1, and below of value 0, as illustrated in Figure 2.3. The carrier does not generally transport as much information in a given time when compared to an analogue system, but this disadvantage is far outweighed by its performance in the face of noise as well as the capability of compressing the data. Furthermore, an encryption scheme can be added on top of the data to prevent easy interception. For this reason, all modern cellular communication systems use digital encoding.

2.2.1 Representing analogue signals in digital format

Since the telephone exchange now works on a digital system in many countries, this necessitates the transmission of analogue signals in digital format. For example, consider transmitting voice across the mobile telephone network. Figure 2.4 shows such a system. The analogue voice is filtered, digitized into a binary stream and coded for transmission. It will travel across the mobile network(s) in digital form until it reaches the destination mobile device. This will convert from digital back to analogue for output to the device's loudspeaker. Converting the analogue signal to digital and then back to analogue does introduce a certain amount of noise but this is minimal compared to leaving the signal in its original analogue state.


Before real-time analogue data can be transmitted on a digital packet-switched network it must undergo a conversion process. The original analogue signal must be sampled (or measured), converted to a digital form (quantized), coded, optionally compressed and encrypted.

2.3.1 Sampling

Sampling is the process whereby the analogue signal is measured at regular intervals and its value recorded at each discrete time interval. It is very important that the signal is sampled at a rate higher than twice the highest frequency component of the original analogue signal otherwise a form of interference called aliasing may be introduced. Consider the problem highlighted in Figure 2.5. Here a 1 kHz signal is being sampled at 4000/second (4 kHz). However, there is a 5 kHz component also present, and the two produce the same result after sampling. For this reason the signal is filtered before sampling to remove any high-frequency components. For the PSTN, the signal is filtered such that the highest frequency is 3.4 kHz and sampling takes place at 8 kHz. Once the signal has been sampled it can then be generally compressed by encoding to reduce the overall amount of data to be sent. This encoded data is then bundled in packets or cells for transmission over the network. The exact amount of data that is carried in each packet is important. Packing a lot of data per packet causes a delay while the packet is being filled. This is referred to as packetization delay, and is described in Section 2.3.6. On the other hand, if the packets are not filled sufficiently this can lead to inefficiency as most of the packet can be taken up by protocol headers.

2.3.2 Coding and CODECs

When converting information from an audio or video stream into digital data, large amounts of information can be generated. Consider, for example, capturing a single frame on a 24-bit true colour graphics screen with a resolution of 1024 × 768 bits. Without compression this will generate 1024 × 768 × 3 (3 bytes = 24 bits of colour) = 2 359 296 or 2.25 megabytes of data. Sending 24 frames per second when capturing a video image will produce 54 megabytes of data every second, yielding a required data rate of 432 Mbps, which is unsustainable on the wireless network.

To reduce the amount of data in the transmission the information is compressed before sending. Many techniques have been employed for both video and audio data but all compression algorithms use one of two basic types of method:

Lossless compression removes redundancy from the information source and on decompression reproduces the original data exactly. This technique is used by graphics compression standards such as GIF and PNG. One technique used for PNG compression is the colour lookup table. Without compression the colour image on a screen requires each colour to be represented by 3 bytes (24 bits), even though there may be 256 or fewer different colours within a particular image. To compress the image each 3-byte code is replaced with a single byte and the actual 3-byte colour data stored in a separate table. This will produce a three-fold saving, less the small space to store the colour table of 768 bytes, and will involve little extra processing of the original image data.

Lossy compression, on the other hand, relies on the fact that there is a lot of information within the image that the eye will not notice if removed. For example, the human eye is less sensitive to changes in colour than changes in intensity when looking at information in a picture. Consequently when images are compressed using the JPEG standard, the colour resolution can be reduced by half when scanning the original image. Lossy compression tends to produce higher compression rates than lossless compression but only really works well on real-world images, for example photographs. Lossless compression techniques such as PNG are more suitable for simple graphics images such as cartoons, figures or line drawings.

A CODEC is a term which refers to a coder/decoder and defines a given compression/ decompression algorithm or technique. For audio compression the technique used for voice data is generally different to that used for music or other audio data. The reason for this is that voice CODECs exploit certain special human voice characteristics to reduce the bandwidth still further. These voice CODECs work well with a voice signal but will not reproduce music well since the CODEC will throw away parts of the original signal not expected to be there. Table 2.1 shows a summary of popular audio CODECs that are currently in use. Some of these are already used in wireless cellular networks such as GSM; others are recommended for use with UMTS and IP.


Excerpted from Convergence Technologies for 3G Networks by Jeffrey Bannister Paul Mather Sebastian Coope Excerpted by permission.
All rights reserved. No part of this excerpt may be reproduced or reprinted without permission in writing from the publisher.
Excerpts are provided by Dial-A-Book Inc. solely for the personal use of visitors to this web site.

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Table of Contents

About the Authors.

1. Introduction.

2. Principles of Communications .

3. GSM Fundamentals.

4. General Packet Radio System.

5. IP Applications for GPRS/UMTS.

6. Universal Mobile Telecommunications System.

7. UMTS Transmission Networks.

8. IP Telephony for UMTS Release 4.

9. Release 5 and Beyond (All-IP).

Glossary of Terms.


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